From 63de8343958b91c8836c5e6ddf1c0106b40e9fe6 Mon Sep 17 00:00:00 2001
From: Joshua Colp <jcolp@digium.com>
Date: Thu, 2 Apr 2009 17:20:52 +0000
Subject: [PATCH] Merge in the RTP engine API.
This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.
Review: http://reviewboard.digium.com/r/209/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
UPGRADE.txt | 6 +-
apps/app_dial.c | 10 +-
channels/chan_agent.c | 1 -
channels/chan_bridge.c | 1 -
channels/chan_gtalk.c | 95 +-
channels/chan_h323.c | 105 +-
channels/chan_jingle.c | 79 +-
channels/chan_local.c | 1 -
channels/chan_mgcp.c | 94 +-
channels/chan_sip.c | 1070 ++++----
channels/chan_skinny.c | 99 +-
channels/chan_unistim.c | 93 +-
configs/sip.conf.sample | 2 +
include/asterisk/_private.h | 1 +
include/asterisk/rtp.h | 416 ---
include/asterisk/rtp_engine.h | 1594 +++++++++++
include/asterisk/stun.h | 71 +
main/Makefile | 4 +-
main/asterisk.c | 2 -
main/loader.c | 2 -
main/rtp.c | 4865 ---------------------------------
main/rtp_engine.c | 1572 +++++++++++
main/stun.c | 475 ++++
res/res_rtp_asterisk.c | 2579 +++++++++++++++++
24 files changed, 7113 insertions(+), 6124 deletions(-)
delete mode 100644 include/asterisk/rtp.h
create mode 100644 include/asterisk/rtp_engine.h
create mode 100644 include/asterisk/stun.h
delete mode 100644 main/rtp.c
create mode 100644 main/rtp_engine.c
create mode 100644 main/stun.c
create mode 100644 res/res_rtp_asterisk.c
diff --git a/UPGRADE.txt b/UPGRADE.txt
index 62551b0f9a..35b0d455aa 100644
--- a/UPGRADE.txt
+++ b/UPGRADE.txt
@@ -20,7 +20,11 @@
From 1.6.2 to 1.6.3:
-* Nothing, yet!
+* The usage of RTP inside of Asterisk has now become modularized. This means
+ the Asterisk RTP stack now exists as a loadable module, res_rtp_asterisk.
+ If you are not using autoload=yes in modules.conf you will need to ensure
+ it is set to load. If not, then any module which uses RTP (such as chan_sip)
+ will not be able to send or receive calls.
From 1.6.1 to 1.6.2:
diff --git a/apps/app_dial.c b/apps/app_dial.c
index 8f6a49ba3c..96bb570813 100644
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -54,7 +54,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/utils.h"
#include "asterisk/app.h"
#include "asterisk/causes.h"
-#include "asterisk/rtp.h"
+#include "asterisk/rtp_engine.h"
#include "asterisk/cdr.h"
#include "asterisk/manager.h"
#include "asterisk/privacy.h"
@@ -745,7 +745,9 @@ static void do_forward(struct chanlist *o,
char *new_cid_num, *new_cid_name;
struct ast_channel *src;
- ast_rtp_make_compatible(c, in, single);
+ if (single) {
+ ast_rtp_instance_early_bridge_make_compatible(c, in);
+ }
if (ast_test_flag64(o, OPT_FORCECLID)) {
new_cid_num = ast_strdup(S_OR(in->macroexten, in->exten));
new_cid_name = NULL; /* XXX no name ? */
@@ -1745,7 +1747,9 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", numsubst);
/* Setup outgoing SDP to match incoming one */
- ast_rtp_make_compatible(tc, chan, !outgoing && !rest);
+ if (!outgoing && !rest) {
+ ast_rtp_instance_early_bridge_make_compatible(tc, chan);
+ }
/* Inherit specially named variables from parent channel */
ast_channel_inherit_variables(chan, tc);
diff --git a/channels/chan_agent.c b/channels/chan_agent.c
index 4e1c282401..b15f7a04e7 100644
--- a/channels/chan_agent.c
+++ b/channels/chan_agent.c
@@ -52,7 +52,6 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/pbx.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
-#include "asterisk/rtp.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
#include "asterisk/file.h"
diff --git a/channels/chan_bridge.c b/channels/chan_bridge.c
index 84909e795f..bd1d0fbeed 100644
--- a/channels/chan_bridge.c
+++ b/channels/chan_bridge.c
@@ -39,7 +39,6 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/pbx.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
-#include "asterisk/rtp.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
#include "asterisk/file.h"
diff --git a/channels/chan_gtalk.c b/channels/chan_gtalk.c
index d608cc05c5..f63cc20270 100644
--- a/channels/chan_gtalk.c
+++ b/channels/chan_gtalk.c
@@ -52,7 +52,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/pbx.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
-#include "asterisk/rtp.h"
+#include "asterisk/rtp_engine.h"
+#include "asterisk/stun.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
#include "asterisk/file.h"
@@ -112,8 +113,8 @@ struct gtalk_pvt {
char cid_name[80]; /*!< Caller ID name */
char exten[80]; /*!< Called extension */
struct ast_channel *owner; /*!< Master Channel */
- struct ast_rtp *rtp; /*!< RTP audio session */
- struct ast_rtp *vrtp; /*!< RTP video session */
+ struct ast_rtp_instance *rtp; /*!< RTP audio session */
+ struct ast_rtp_instance *vrtp; /*!< RTP video session */
int jointcapability; /*!< Supported capability at both ends (codecs ) */
int peercapability;
struct gtalk_pvt *next; /* Next entity */
@@ -183,11 +184,6 @@ static int gtalk_sendhtml(struct ast_channel *ast, int subclass, const char *dat
static struct gtalk_pvt *gtalk_alloc(struct gtalk *client, const char *us, const char *them, const char *sid);
static char *gtalk_do_reload(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *gtalk_show_channels(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-/*----- RTP interface functions */
-static int gtalk_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp,
- struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active);
-static enum ast_rtp_get_result gtalk_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
-static int gtalk_get_codec(struct ast_channel *chan);
/*! \brief PBX interface structure for channel registration */
static const struct ast_channel_tech gtalk_tech = {
@@ -197,7 +193,7 @@ static const struct ast_channel_tech gtalk_tech = {
.requester = gtalk_request,
.send_digit_begin = gtalk_digit_begin,
.send_digit_end = gtalk_digit_end,
- .bridge = ast_rtp_bridge,
+ .bridge = ast_rtp_instance_bridge,
.call = gtalk_call,
.hangup = gtalk_hangup,
.answer = gtalk_answer,
@@ -216,14 +212,6 @@ static struct sched_context *sched; /*!< The scheduling context */
static struct io_context *io; /*!< The IO context */
static struct in_addr __ourip;
-/*! \brief RTP driver interface */
-static struct ast_rtp_protocol gtalk_rtp = {
- type: "Gtalk",
- get_rtp_info: gtalk_get_rtp_peer,
- set_rtp_peer: gtalk_set_rtp_peer,
- get_codec: gtalk_get_codec,
-};
-
static struct ast_cli_entry gtalk_cli[] = {
AST_CLI_DEFINE(gtalk_do_reload, "Reload GoogleTalk configuration"),
AST_CLI_DEFINE(gtalk_show_channels, "Show GoogleTalk channels"),
@@ -371,7 +359,7 @@ static int add_codec_to_answer(const struct gtalk_pvt *p, int codec, iks *dcodec
iks_insert_node(dcodecs, payload_gsm);
res++;
}
- ast_rtp_lookup_code(p->rtp, 1, codec);
+
return res;
}
@@ -523,18 +511,19 @@ static int gtalk_answer(struct ast_channel *ast)
return res;
}
-static enum ast_rtp_get_result gtalk_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
+static enum ast_rtp_glue_result gtalk_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
struct gtalk_pvt *p = chan->tech_pvt;
- enum ast_rtp_get_result res = AST_RTP_GET_FAILED;
+ enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_FORBID;
if (!p)
return res;
ast_mutex_lock(&p->lock);
if (p->rtp){
- *rtp = p->rtp;
- res = AST_RTP_TRY_PARTIAL;
+ ao2_ref(p->rtp, +1);
+ *instance = p->rtp;
+ res = AST_RTP_GLUE_RESULT_LOCAL;
}
ast_mutex_unlock(&p->lock);
@@ -547,7 +536,7 @@ static int gtalk_get_codec(struct ast_channel *chan)
return p->peercapability;
}
-static int gtalk_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active)
+static int gtalk_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *trtp, int codecs, int nat_active)
{
struct gtalk_pvt *p;
@@ -567,6 +556,13 @@ static int gtalk_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, str
return 0;
}
+static struct ast_rtp_glue gtalk_rtp_glue = {
+ .type = "Gtalk",
+ .get_rtp_info = gtalk_get_rtp_peer,
+ .get_codec = gtalk_get_codec,
+ .update_peer = gtalk_set_rtp_peer,
+};
+
static int gtalk_response(struct gtalk *client, char *from, ikspak *pak, const char *reasonstr, const char *reasonstr2)
{
iks *response = NULL, *error = NULL, *reason = NULL;
@@ -617,13 +613,13 @@ static int gtalk_is_answered(struct gtalk *client, ikspak *pak)
/* codec points to the first <payload-type/> tag */
codec = iks_first_tag(iks_first_tag(iks_first_tag(pak->x)));
while (codec) {
- ast_rtp_set_m_type(tmp->rtp, atoi(iks_find_attrib(codec, "id")));
- ast_rtp_set_rtpmap_type(tmp->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
+ ast_rtp_codecs_payloads_set_m_type(ast_rtp_instance_get_codecs(tmp->rtp), tmp->rtp, atoi(iks_find_attrib(codec, "id")));
+ ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(tmp->rtp), tmp->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
codec = iks_next_tag(codec);
}
/* Now gather all of the codecs that we are asked for */
- ast_rtp_get_current_formats(tmp->rtp, &tmp->peercapability, &peernoncodeccapability);
+ ast_rtp_codecs_payload_formats(ast_rtp_instance_get_codecs(tmp->rtp), &tmp->peercapability, &peernoncodeccapability);
/* at this point, we received an awser from the remote Gtalk client,
which allows us to compare capabilities */
@@ -810,7 +806,7 @@ static int gtalk_create_candidates(struct gtalk *client, struct gtalk_pvt *p, ch
goto safeout;
}
- ast_rtp_get_us(p->rtp, &sin);
+ ast_rtp_instance_get_local_address(p->rtp, &sin);
ast_find_ourip(&us, bindaddr);
if (!strcmp(ast_inet_ntoa(us), "127.0.0.1")) {
ast_log(LOG_WARNING, "Found a loopback IP on the system, check your network configuration or set the bindaddr attribute.");
@@ -951,8 +947,9 @@ static struct gtalk_pvt *gtalk_alloc(struct gtalk *client, const char *us, const
tmp->initiator = 1;
}
/* clear codecs */
- tmp->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
- ast_rtp_pt_clear(tmp->rtp);
+ tmp->rtp = ast_rtp_instance_new(NULL, sched, &bindaddr, NULL);
+ ast_rtp_instance_set_prop(tmp->rtp, AST_RTP_PROPERTY_RTCP, 1);
+ ast_rtp_codecs_payloads_clear(ast_rtp_instance_get_codecs(tmp->rtp), tmp->rtp);
/* add user configured codec capabilites */
if (client->capability)
@@ -1014,20 +1011,20 @@ static struct ast_channel *gtalk_new(struct gtalk *client, struct gtalk_pvt *i,
/* Set Frame packetization */
if (i->rtp)
- ast_rtp_codec_setpref(i->rtp, &i->prefs);
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(i->rtp), i->rtp, &i->prefs);
tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | (i->jointcapability & AST_FORMAT_VIDEO_MASK);
fmt = ast_best_codec(tmp->nativeformats);
if (i->rtp) {
- ast_rtp_setstun(i->rtp, 1);
- ast_channel_set_fd(tmp, 0, ast_rtp_fd(i->rtp));
- ast_channel_set_fd(tmp, 1, ast_rtcp_fd(i->rtp));
+ ast_rtp_instance_set_prop(i->rtp, AST_RTP_PROPERTY_STUN, 1);
+ ast_channel_set_fd(tmp, 0, ast_rtp_instance_fd(i->rtp, 0));
+ ast_channel_set_fd(tmp, 1, ast_rtp_instance_fd(i->rtp, 1));
}
if (i->vrtp) {
- ast_rtp_setstun(i->rtp, 1);
- ast_channel_set_fd(tmp, 2, ast_rtp_fd(i->vrtp));
- ast_channel_set_fd(tmp, 3, ast_rtcp_fd(i->vrtp));
+ ast_rtp_instance_set_prop(i->vrtp, AST_RTP_PROPERTY_STUN, 1);
+ ast_channel_set_fd(tmp, 2, ast_rtp_instance_fd(i->vrtp, 0));
+ ast_channel_set_fd(tmp, 3, ast_rtp_instance_fd(i->vrtp, 1));
}
if (state == AST_STATE_RING)
tmp->rings = 1;
@@ -1142,9 +1139,9 @@ static void gtalk_free_pvt(struct gtalk *client, struct gtalk_pvt *p)
if (p->owner)
ast_log(LOG_WARNING, "Uh oh, there's an owner, this is going to be messy.\n");
if (p->rtp)
- ast_rtp_destroy(p->rtp);
+ ast_rtp_instance_destroy(p->rtp);
if (p->vrtp)
- ast_rtp_destroy(p->vrtp);
+ ast_rtp_instance_destroy(p->vrtp);
gtalk_free_candidates(p->theircandidates);
ast_free(p);
}
@@ -1207,13 +1204,13 @@ static int gtalk_newcall(struct gtalk *client, ikspak *pak)
codec = iks_first_tag(iks_first_tag(iks_first_tag(pak->x)));
while (codec) {
- ast_rtp_set_m_type(p->rtp, atoi(iks_find_attrib(codec, "id")));
- ast_rtp_set_rtpmap_type(p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
+ ast_rtp_codecs_payloads_set_m_type(ast_rtp_instance_get_codecs(p->rtp), p->rtp, atoi(iks_find_attrib(codec, "id")));
+ ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(p->rtp), p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
codec = iks_next_tag(codec);
}
/* Now gather all of the codecs that we are asked for */
- ast_rtp_get_current_formats(p->rtp, &p->peercapability, &peernoncodeccapability);
+ ast_rtp_codecs_payload_formats(ast_rtp_instance_get_codecs(p->rtp), &p->peercapability, &peernoncodeccapability);
p->jointcapability = p->capability & p->peercapability;
ast_mutex_unlock(&p->lock);
@@ -1277,16 +1274,16 @@ static int gtalk_update_stun(struct gtalk *client, struct gtalk_pvt *p)
p->ourcandidates->username);
/* Find out the result of the STUN */
- ast_rtp_get_peer(p->rtp, &aux);
+ ast_rtp_instance_get_remote_address(p->rtp, &aux);
/* If the STUN result is different from the IP of the hostname,
lock on the stun IP of the hostname advertised by the
remote client */
if (aux.sin_addr.s_addr &&
aux.sin_addr.s_addr != sin.sin_addr.s_addr)
- ast_rtp_stun_request(p->rtp, &aux, username);
+ ast_rtp_instance_stun_request(p->rtp, &aux, username);
else
- ast_rtp_stun_request(p->rtp, &sin, username);
+ ast_rtp_instance_stun_request(p->rtp, &sin, username);
if (aux.sin_addr.s_addr) {
ast_debug(4, "Receiving RTP traffic from IP %s, matches with remote candidate's IP %s\n", ast_inet_ntoa(aux.sin_addr), tmp->ip);
@@ -1387,7 +1384,7 @@ static struct ast_frame *gtalk_rtp_read(struct ast_channel *ast, struct gtalk_pv
if (!p->rtp)
return &ast_null_frame;
- f = ast_rtp_read(p->rtp);
+ f = ast_rtp_instance_read(p->rtp, 0);
gtalk_update_stun(p->parent, p);
if (p->owner) {
/* We already hold the channel lock */
@@ -1438,7 +1435,7 @@ static int gtalk_write(struct ast_channel *ast, struct ast_frame *frame)
if (p) {
ast_mutex_lock(&p->lock);
if (p->rtp) {
- res = ast_rtp_write(p->rtp, frame);
+ res = ast_rtp_instance_write(p->rtp, frame);
}
ast_mutex_unlock(&p->lock);
}
@@ -1447,7 +1444,7 @@ static int gtalk_write(struct ast_channel *ast, struct ast_frame *frame)
if (p) {
ast_mutex_lock(&p->lock);
if (p->vrtp) {
- res = ast_rtp_write(p->vrtp, frame);
+ res = ast_rtp_instance_write(p->vrtp, frame);
}
ast_mutex_unlock(&p->lock);
}
@@ -2062,7 +2059,7 @@ static int load_module(void)
return 0;
}
- ast_rtp_proto_register(>alk_rtp);
+ ast_rtp_glue_register(>alk_rtp_glue);
ast_cli_register_multiple(gtalk_cli, ARRAY_LEN(gtalk_cli));
/* Make sure we can register our channel type */
@@ -2086,7 +2083,7 @@ static int unload_module(void)
ast_cli_unregister_multiple(gtalk_cli, ARRAY_LEN(gtalk_cli));
/* First, take us out of the channel loop */
ast_channel_unregister(>alk_tech);
- ast_rtp_proto_unregister(>alk_rtp);
+ ast_rtp_glue_unregister(>alk_rtp_glue);
if (!ast_mutex_lock(>alklock)) {
/* Hangup all interfaces if they have an owner */
diff --git a/channels/chan_h323.c b/channels/chan_h323.c
index 2342ecfbba..c3e074d143 100644
--- a/channels/chan_h323.c
+++ b/channels/chan_h323.c
@@ -76,7 +76,7 @@ extern "C" {
#include "asterisk/utils.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
-#include "asterisk/rtp.h"
+#include "asterisk/rtp_engine.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
#include "asterisk/cli.h"
@@ -161,7 +161,7 @@ struct oh323_pvt {
char accountcode[256]; /*!< Account code */
char rdnis[80]; /*!< Referring DNIS, if available */
int amaflags; /*!< AMA Flags */
- struct ast_rtp *rtp; /*!< RTP Session */
+ struct ast_rtp_instance *rtp; /*!< RTP Session */
struct ast_dsp *vad; /*!< Used for in-band DTMF detection */
int nativeformats; /*!< Codec formats supported by a channel */
int needhangup; /*!< Send hangup when Asterisk is ready */
@@ -254,7 +254,7 @@ static const struct ast_channel_tech oh323_tech = {
.write = oh323_write,
.indicate = oh323_indicate,
.fixup = oh323_fixup,
- .bridge = ast_rtp_bridge,
+ .bridge = ast_rtp_instance_bridge,
};
static const char* redirectingreason2str(int redirectingreason)
@@ -381,8 +381,8 @@ static void __oh323_update_info(struct ast_channel *c, struct oh323_pvt *pvt)
if (pvt->update_rtp_info > 0) {
if (pvt->rtp) {
ast_jb_configure(c, &global_jbconf);
- ast_channel_set_fd(c, 0, ast_rtp_fd(pvt->rtp));
- ast_channel_set_fd(c, 1, ast_rtcp_fd(pvt->rtp));
+ ast_channel_set_fd(c, 0, ast_rtp_instance_fd(pvt->rtp, 0));
+ ast_channel_set_fd(c, 1, ast_rtp_instance_fd(pvt->rtp, 1));
ast_queue_frame(pvt->owner, &ast_null_frame); /* Tell Asterisk to apply changes */
}
pvt->update_rtp_info = -1;
@@ -444,7 +444,7 @@ static void __oh323_destroy(struct oh323_pvt *pvt)
AST_SCHED_DEL(sched, pvt->DTMFsched);
if (pvt->rtp) {
- ast_rtp_destroy(pvt->rtp);
+ ast_rtp_instance_destroy(pvt->rtp);
}
/* Free dsp used for in-band DTMF detection */
@@ -510,7 +510,7 @@ static int oh323_digit_begin(struct ast_channel *c, char digit)
if (h323debug) {
ast_log(LOG_DTMF, "Begin sending out-of-band digit %c on %s\n", digit, c->name);
}
- ast_rtp_senddigit_begin(pvt->rtp, digit);
+ ast_rtp_instance_dtmf_begin(pvt->rtp, digit);
ast_mutex_unlock(&pvt->lock);
} else if (pvt->txDtmfDigit != digit) {
/* in-band DTMF */
@@ -549,7 +549,7 @@ static int oh323_digit_end(struct ast_channel *c, char digit, unsigned int durat
if (h323debug) {
ast_log(LOG_DTMF, "End sending out-of-band digit %c on %s, duration %d\n", digit, c->name, duration);
}
- ast_rtp_senddigit_end(pvt->rtp, digit);
+ ast_rtp_instance_dtmf_end(pvt->rtp, digit);
ast_mutex_unlock(&pvt->lock);
} else {
/* in-band DTMF */
@@ -747,11 +747,11 @@ static struct ast_frame *oh323_rtp_read(struct oh323_pvt *pvt)
/* Only apply it for the first packet, we just need the correct ip/port */
if (pvt->options.nat) {
- ast_rtp_setnat(pvt->rtp, pvt->options.nat);
+ ast_rtp_instance_set_prop(pvt->rtp, AST_RTP_PROPERTY_NAT, pvt->options.nat);
pvt->options.nat = 0;
}
- f = ast_rtp_read(pvt->rtp);
+ f = ast_rtp_instance_read(pvt->rtp, 0);
/* Don't send RFC2833 if we're not supposed to */
if (f && (f->frametype == AST_FRAME_DTMF) && !(pvt->options.dtmfmode & (H323_DTMF_RFC2833 | H323_DTMF_CISCO))) {
return &ast_null_frame;
@@ -808,7 +808,7 @@ static struct ast_frame *oh323_read(struct ast_channel *c)
break;
case 1:
if (pvt->rtp)
- fr = ast_rtcp_read(pvt->rtp);
+ fr = ast_rtp_instance_read(pvt->rtp, 1);
else
fr = &ast_null_frame;
break;
@@ -842,7 +842,7 @@ static int oh323_write(struct ast_channel *c, struct ast_frame *frame)
if (pvt) {
ast_mutex_lock(&pvt->lock);
if (pvt->rtp && !pvt->recvonly)
- res = ast_rtp_write(pvt->rtp, frame);
+ res = ast_rtp_instance_write(pvt->rtp, frame);
__oh323_update_info(c, pvt);
ast_mutex_unlock(&pvt->lock);
}
@@ -910,7 +910,7 @@ static int oh323_indicate(struct ast_channel *c, int condition, const void *data
res = 0;
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_new_source(pvt->rtp);
+ ast_rtp_instance_new_source(pvt->rtp);
res = 0;
break;
case AST_CONTROL_PROCEEDING:
@@ -946,17 +946,17 @@ static int oh323_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
static int __oh323_rtp_create(struct oh323_pvt *pvt)
{
- struct in_addr our_addr;
+ struct sockaddr_in our_addr;
if (pvt->rtp)
return 0;
- if (ast_find_ourip(&our_addr, bindaddr)) {
+ if (ast_find_ourip(&our_addr.sin_addr, bindaddr)) {
ast_mutex_unlock(&pvt->lock);
ast_log(LOG_ERROR, "Unable to locate local IP address for RTP stream\n");
return -1;
}
- pvt->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, our_addr);
+ pvt->rtp = ast_rtp_instance_new(NULL, sched, &our_addr, NULL);
if (!pvt->rtp) {
ast_mutex_unlock(&pvt->lock);
ast_log(LOG_WARNING, "Unable to create RTP session: %s\n", strerror(errno));
@@ -965,24 +965,24 @@ static int __oh323_rtp_create(struct oh323_pvt *pvt)
if (h323debug)
ast_debug(1, "Created RTP channel\n");
- ast_rtp_setqos(pvt->rtp, tos, cos, "H323 RTP");
+ ast_rtp_instance_set_qos(pvt->rtp, tos, cos, "H323 RTP");
if (h323debug)
ast_debug(1, "Setting NAT on RTP to %d\n", pvt->options.nat);
- ast_rtp_setnat(pvt->rtp, pvt->options.nat);
+ ast_rtp_instance_set_prop(pvt->rtp, AST_RTP_PROPERTY_NAT, pvt->options.nat);
if (pvt->dtmf_pt[0] > 0)
- ast_rtp_set_rtpmap_type(pvt->rtp, pvt->dtmf_pt[0], "audio", "telephone-event", 0);
+ ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, pvt->dtmf_pt[0], "audio", "telephone-event", 0);
if (pvt->dtmf_pt[1] > 0)
- ast_rtp_set_rtpmap_type(pvt->rtp, pvt->dtmf_pt[1], "audio", "cisco-telephone-event", 0);
+ ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, pvt->dtmf_pt[1], "audio", "cisco-telephone-event", 0);
if (pvt->peercapability)
- ast_rtp_codec_setpref(pvt->rtp, &pvt->peer_prefs);
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, &pvt->peer_prefs);
if (pvt->owner && !ast_channel_trylock(pvt->owner)) {
ast_jb_configure(pvt->owner, &global_jbconf);
- ast_channel_set_fd(pvt->owner, 0, ast_rtp_fd(pvt->rtp));
- ast_channel_set_fd(pvt->owner, 1, ast_rtcp_fd(pvt->rtp));
+ ast_channel_set_fd(pvt->owner, 0, ast_rtp_instance_fd(pvt->rtp, 0));
+ ast_channel_set_fd(pvt->owner, 1, ast_rtp_instance_fd(pvt->rtp, 1));
ast_queue_frame(pvt->owner, &ast_null_frame); /* Tell Asterisk to apply changes */
ast_channel_unlock(pvt->owner);
} else
@@ -1028,13 +1028,13 @@ static struct ast_channel *__oh323_new(struct oh323_pvt *pvt, int state, const c
if (!pvt->rtp)
__oh323_rtp_create(pvt);
#if 0
- ast_channel_set_fd(ch, 0, ast_rtp_fd(pvt->rtp));
- ast_channel_set_fd(ch, 1, ast_rtcp_fd(pvt->rtp));
+ ast_channel_set_fd(ch, 0, ast_rtp_instance_fd(pvt->rtp, 0));
+ ast_channel_set_fd(ch, 1, ast_rtp_instance_fd(pvt->rtp, 1));
#endif
#ifdef VIDEO_SUPPORT
if (pvt->vrtp) {
- ast_channel_set_fd(ch, 2, ast_rtp_fd(pvt->vrtp));
- ast_channel_set_fd(ch, 3, ast_rtcp_fd(pvt->vrtp));
+ ast_channel_set_fd(ch, 2, ast_rtp_instance_fd(pvt->vrtp, 0));
+ ast_channel_set_fd(ch, 3, ast_rtp_instance_fd(pvt->vrtp, 1));
}
#endif
#ifdef T38_SUPPORT
@@ -1112,7 +1112,7 @@ static struct oh323_pvt *oh323_alloc(int callid)
}
if (!pvt->cd.call_token) {
ast_log(LOG_ERROR, "Not enough memory to alocate call token\n");
- ast_rtp_destroy(pvt->rtp);
+ ast_rtp_instance_destroy(pvt->rtp);
ast_free(pvt);
return NULL;
}
@@ -1912,7 +1912,7 @@ static struct rtp_info *external_rtp_create(unsigned call_reference, const char
return NULL;
}
/* figure out our local RTP port and tell the H.323 stack about it */
- ast_rtp_get_us(pvt->rtp, &us);
+ ast_rtp_instance_get_local_address(pvt->rtp, &us);
ast_mutex_unlock(&pvt->lock);
ast_copy_string(info->addr, ast_inet_ntoa(us.sin_addr), sizeof(info->addr));
@@ -1931,7 +1931,6 @@ static void setup_rtp_connection(unsigned call_reference, const char *remoteIp,
{
struct oh323_pvt *pvt;
struct sockaddr_in them;
- struct rtpPayloadType rtptype;
int nativeformats_changed;
enum { NEED_NONE, NEED_HOLD, NEED_UNHOLD } rtp_change = NEED_NONE;
@@ -1953,7 +1952,7 @@ static void setup_rtp_connection(unsigned call_reference, const char *remoteIp,
__oh323_rtp_create(pvt);
if ((pt == 2) && (pvt->jointcapability & AST_FORMAT_G726_AAL2)) {
- ast_rtp_set_rtpmap_type(pvt->rtp, pt, "audio", "G726-32", AST_RTP_OPT_G726_NONSTANDARD);
+ ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, pt, "audio", "G726-32", AST_RTP_OPT_G726_NONSTANDARD);
}
them.sin_family = AF_INET;
@@ -1962,13 +1961,13 @@ static void setup_rtp_connection(unsigned call_reference, const char *remoteIp,
them.sin_port = htons(remotePort);
if (them.sin_addr.s_addr) {
- ast_rtp_set_peer(pvt->rtp, &them);
+ ast_rtp_instance_set_remote_address(pvt->rtp, &them);
if (pvt->recvonly) {
pvt->recvonly = 0;
rtp_change = NEED_UNHOLD;
}
} else {
- ast_rtp_stop(pvt->rtp);
+ ast_rtp_instance_stop(pvt->rtp);
if (!pvt->recvonly) {
pvt->recvonly = 1;
rtp_change = NEED_HOLD;
@@ -1978,7 +1977,7 @@ static void setup_rtp_connection(unsigned call_reference, const char *remoteIp,
/* Change native format to reflect information taken from OLC/OLCAck */
nativeformats_changed = 0;
if (pt != 128 && pvt->rtp) { /* Payload type is invalid, so try to use previously decided */
- rtptype = ast_rtp_lookup_pt(pvt->rtp, pt);
+ struct ast_rtp_payload_type rtptype = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(pvt->rtp), pt);
if (h323debug)
ast_debug(1, "Native format is set to %d from %d by RTP payload type %d\n", rtptype.code, pvt->nativeformats, pt);
if (pvt->nativeformats != rtptype.code) {
@@ -2359,7 +2358,7 @@ static void cleanup_connection(unsigned call_reference, const char *call_token)
}
if (pvt->rtp) {
/* Immediately stop RTP */
- ast_rtp_destroy(pvt->rtp);
+ ast_rtp_instance_destroy(pvt->rtp);
pvt->rtp = NULL;
}
/* Free dsp used for in-band DTMF detection */
@@ -2421,7 +2420,7 @@ static void set_dtmf_payload(unsigned call_reference, const char *token, int pay
return;
}
if (pvt->rtp) {
- ast_rtp_set_rtpmap_type(pvt->rtp, payload, "audio", (is_cisco ? "cisco-telephone-event" : "telephone-event"), 0);
+ ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, payload, "audio", (is_cisco ? "cisco-telephone-event" : "telephone-event"), 0);
}
pvt->dtmf_pt[is_cisco ? 1 : 0] = payload;
ast_mutex_unlock(&pvt->lock);
@@ -2452,7 +2451,7 @@ static void set_peer_capabilities(unsigned call_reference, const char *token, in
}
}
if (pvt->rtp)
- ast_rtp_codec_setpref(pvt->rtp, &pvt->peer_prefs);
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, &pvt->peer_prefs);
}
ast_mutex_unlock(&pvt->lock);
}
@@ -3113,19 +3112,19 @@ static int reload(void)
static struct ast_cli_entry cli_h323_reload =
AST_CLI_DEFINE(handle_cli_h323_reload, "Reload H.323 configuration");
-static enum ast_rtp_get_result oh323_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
+static enum ast_rtp_glue_result oh323_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
struct oh323_pvt *pvt;
- enum ast_rtp_get_result res = AST_RTP_TRY_PARTIAL;
+ enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_LOCAL;
if (!(pvt = (struct oh323_pvt *)chan->tech_pvt))
- return AST_RTP_GET_FAILED;
+ return AST_RTP_GLUE_RESULT_FORBID;
ast_mutex_lock(&pvt->lock);
- *rtp = pvt->rtp;
+ *instance = pvt->rtp ? ao2_ref(pvt->rtp, +1), pvt->rtp : NULL;
#if 0
if (pvt->options.bridge) {
- res = AST_RTP_TRY_NATIVE;
+ res = AST_RTP_GLUE_RESULT_REMOTE;
}
#endif
ast_mutex_unlock(&pvt->lock);
@@ -3133,11 +3132,6 @@ static enum ast_rtp_get_result oh323_get_rtp_peer(struct ast_channel *chan, stru
return res;
}
-static enum ast_rtp_get_result oh323_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
-{
- return AST_RTP_GET_FAILED;
-}
-
static char *convertcap(int cap)
{
switch (cap) {
@@ -3165,7 +3159,7 @@ static char *convertcap(int cap)
}
}
-static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active)
+static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *trtp, int codecs, int nat_active)
{
/* XXX Deal with Video */
struct oh323_pvt *pvt;
@@ -3183,19 +3177,18 @@ static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, str
ast_log(LOG_ERROR, "No Private Structure, this is bad\n");
return -1;
}
- ast_rtp_get_peer(rtp, &them);
- ast_rtp_get_us(rtp, &us);
+ ast_rtp_instance_get_remote_address(rtp, &them);
+ ast_rtp_instance_get_local_address(rtp, &us);
#if 0 /* Native bridge still isn't ready */
h323_native_bridge(pvt->cd.call_token, ast_inet_ntoa(them.sin_addr), mode);
#endif
return 0;
}
-static struct ast_rtp_protocol oh323_rtp = {
+static struct ast_rtp_glue oh323_rtp_glue = {
.type = "H323",
.get_rtp_info = oh323_get_rtp_peer,
- .get_vrtp_info = oh323_get_vrtp_peer,
- .set_rtp_peer = oh323_set_rtp_peer,
+ .update_peer = oh323_set_rtp_peer,
};
static enum ast_module_load_result load_module(void)
@@ -3250,7 +3243,7 @@ static enum ast_module_load_result load_module(void)
}
ast_cli_register_multiple(cli_h323, sizeof(cli_h323) / sizeof(struct ast_cli_entry));
- ast_rtp_proto_register(&oh323_rtp);
+ ast_rtp_glue_register(&oh323_rtp_glue);
/* Register our callback functions */
h323_callback_register(setup_incoming_call,
@@ -3271,7 +3264,7 @@ static enum ast_module_load_result load_module(void)
/* start the h.323 listener */
if (h323_start_listener(h323_signalling_port, bindaddr)) {
ast_log(LOG_ERROR, "Unable to create H323 listener.\n");
- ast_rtp_proto_unregister(&oh323_rtp);
+ ast_rtp_glue_unregister(&oh323_rtp_glue);
ast_cli_unregister_multiple(cli_h323, sizeof(cli_h323) / sizeof(struct ast_cli_entry));
ast_cli_unregister(&cli_h323_reload);
h323_end_process();
@@ -3310,7 +3303,7 @@ static int unload_module(void)
ast_cli_unregister(&cli_h323_reload);
ast_channel_unregister(&oh323_tech);
- ast_rtp_proto_unregister(&oh323_rtp);
+ ast_rtp_glue_unregister(&oh323_rtp_glue);
if (!ast_mutex_lock(&iflock)) {
/* hangup all interfaces if they have an owner */
diff --git a/channels/chan_jingle.c b/channels/chan_jingle.c
index d239fd717f..e1a60ae7e2 100644
--- a/channels/chan_jingle.c
+++ b/channels/chan_jingle.c
@@ -53,7 +53,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/pbx.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
-#include "asterisk/rtp.h"
+#include "asterisk/rtp_engine.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
#include "asterisk/file.h"
@@ -112,9 +112,9 @@ struct jingle_pvt {
char exten[80]; /*!< Called extension */
struct ast_channel *owner; /*!< Master Channel */
char audio_content_name[100]; /*!< name attribute of content tag */
- struct ast_rtp *rtp; /*!< RTP audio session */
+ struct ast_rtp_instance *rtp; /*!< RTP audio session */
char video_content_name[100]; /*!< name attribute of content tag */
- struct ast_rtp *vrtp; /*!< RTP video session */
+ struct ast_rtp_instance *vrtp; /*!< RTP video session */
int jointcapability; /*!< Supported capability at both ends (codecs ) */
int peercapability;
struct jingle_pvt *next; /* Next entity */
@@ -183,11 +183,6 @@ static int jingle_sendhtml(struct ast_channel *ast, int subclass, const char *da
static struct jingle_pvt *jingle_alloc(struct jingle *client, const char *from, const char *sid);
static char *jingle_show_channels(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *jingle_do_reload(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-/*----- RTP interface functions */
-static int jingle_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp,
- struct ast_rtp *vrtp, struct ast_rtp *tpeer, int codecs, int nat_active);
-static enum ast_rtp_get_result jingle_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
-static int jingle_get_codec(struct ast_channel *chan);
/*! \brief PBX interface structure for channel registration */
static const struct ast_channel_tech jingle_tech = {
@@ -197,7 +192,7 @@ static const struct ast_channel_tech jingle_tech = {
.requester = jingle_request,
.send_digit_begin = jingle_digit_begin,
.send_digit_end = jingle_digit_end,
- .bridge = ast_rtp_bridge,
+ .bridge = ast_rtp_instance_bridge,
.call = jingle_call,
.hangup = jingle_hangup,
.answer = jingle_answer,
@@ -216,15 +211,6 @@ static struct sched_context *sched; /*!< The scheduling context */
static struct io_context *io; /*!< The IO context */
static struct in_addr __ourip;
-
-/*! \brief RTP driver interface */
-static struct ast_rtp_protocol jingle_rtp = {
- type: "Jingle",
- get_rtp_info: jingle_get_rtp_peer,
- set_rtp_peer: jingle_set_rtp_peer,
- get_codec: jingle_get_codec,
-};
-
static struct ast_cli_entry jingle_cli[] = {
AST_CLI_DEFINE(jingle_do_reload, "Reload Jingle configuration"),
AST_CLI_DEFINE(jingle_show_channels, "Show Jingle channels"),
@@ -304,7 +290,6 @@ static void add_codec_to_answer(const struct jingle_pvt *p, int codec, iks *dcod
iks_insert_attrib(payload_g723, "name", "G723");
iks_insert_node(dcodecs, payload_g723);
}
- ast_rtp_lookup_code(p->rtp, 1, codec);
}
static int jingle_accept_call(struct jingle *client, struct jingle_pvt *p)
@@ -398,18 +383,19 @@ static int jingle_answer(struct ast_channel *ast)
return res;
}
-static enum ast_rtp_get_result jingle_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
+static enum ast_rtp_glue_result jingle_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
struct jingle_pvt *p = chan->tech_pvt;
- enum ast_rtp_get_result res = AST_RTP_GET_FAILED;
+ enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_FORBID;
if (!p)
return res;
ast_mutex_lock(&p->lock);
if (p->rtp) {
- *rtp = p->rtp;
- res = AST_RTP_TRY_PARTIAL;
+ ao2_ref(p->rtp, +1);
+ *instance = p->rtp;
+ res = AST_RTP_GLUE_RESULT_LOCAL;
}
ast_mutex_unlock(&p->lock);
@@ -422,7 +408,7 @@ static int jingle_get_codec(struct ast_channel *chan)
return p->peercapability;
}
-static int jingle_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, struct ast_rtp *tpeer, int codecs, int nat_active)
+static int jingle_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *tpeer, int codecs, int nat_active)
{
struct jingle_pvt *p;
@@ -442,6 +428,13 @@ static int jingle_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, st
return 0;
}
+static struct ast_rtp_glue jingle_rtp_glue = {
+ .type = "Jingle",
+ .get_rtp_info = jingle_get_rtp_peer,
+ .get_codec = jingle_get_codec,
+ .update_peer = jingle_set_rtp_peer,
+};
+
static int jingle_response(struct jingle *client, ikspak *pak, const char *reasonstr, const char *reasonstr2)
{
iks *response = NULL, *error = NULL, *reason = NULL;
@@ -621,7 +614,7 @@ static int jingle_create_candidates(struct jingle *client, struct jingle_pvt *p,
goto safeout;
}
- ast_rtp_get_us(p->rtp, &sin);
+ ast_rtp_instance_get_local_address(p->rtp, &sin);
ast_find_ourip(&us, bindaddr);
/* Setup our first jingle candidate */
@@ -779,7 +772,7 @@ static struct jingle_pvt *jingle_alloc(struct jingle *client, const char *from,
ast_copy_string(tmp->them, idroster, sizeof(tmp->them));
tmp->initiator = 1;
}
- tmp->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
+ tmp->rtp = ast_rtp_instance_new(NULL, sched, &bindaddr, NULL);
tmp->parent = client;
if (!tmp->rtp) {
ast_log(LOG_WARNING, "Out of RTP sessions?\n");
@@ -825,18 +818,18 @@ static struct ast_channel *jingle_new(struct jingle *client, struct jingle_pvt *
/* Set Frame packetization */
if (i->rtp)
- ast_rtp_codec_setpref(i->rtp, &i->prefs);
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(i->rtp), i->rtp, &i->prefs);
tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | (i->jointcapability & AST_FORMAT_VIDEO_MASK);
fmt = ast_best_codec(tmp->nativeformats);
if (i->rtp) {
- ast_channel_set_fd(tmp, 0, ast_rtp_fd(i->rtp));
- ast_channel_set_fd(tmp, 1, ast_rtcp_fd(i->rtp));
+ ast_channel_set_fd(tmp, 0, ast_rtp_instance_fd(i->rtp, 0));
+ ast_channel_set_fd(tmp, 1, ast_rtp_instance_fd(i->rtp, 1));
}
if (i->vrtp) {
- ast_channel_set_fd(tmp, 2, ast_rtp_fd(i->vrtp));
- ast_channel_set_fd(tmp, 3, ast_rtcp_fd(i->vrtp));
+ ast_channel_set_fd(tmp, 2, ast_rtp_instance_fd(i->vrtp, 0));
+ ast_channel_set_fd(tmp, 3, ast_rtp_instance_fd(i->vrtp, 1));
}
if (state == AST_STATE_RING)
tmp->rings = 1;
@@ -942,9 +935,9 @@ static void jingle_free_pvt(struct jingle *client, struct jingle_pvt *p)
if (p->owner)
ast_log(LOG_WARNING, "Uh oh, there's an owner, this is going to be messy.\n");
if (p->rtp)
- ast_rtp_destroy(p->rtp);
+ ast_rtp_instance_destroy(p->rtp);
if (p->vrtp)
- ast_rtp_destroy(p->vrtp);
+ ast_rtp_instance_destroy(p->vrtp);
jingle_free_candidates(p->theircandidates);
ast_free(p);
}
@@ -1009,8 +1002,8 @@ static int jingle_newcall(struct jingle *client, ikspak *pak)
ast_copy_string(p->audio_content_name, iks_find_attrib(content, "name"), sizeof(p->audio_content_name));
while (codec) {
- ast_rtp_set_m_type(p->rtp, atoi(iks_find_attrib(codec, "id")));
- ast_rtp_set_rtpmap_type(p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
+ ast_rtp_codecs_payloads_set_m_type(ast_rtp_instance_get_codecs(p->rtp), p->rtp, atoi(iks_find_attrib(codec, "id")));
+ ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(p->rtp), p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
codec = iks_next(codec);
}
}
@@ -1025,8 +1018,8 @@ static int jingle_newcall(struct jingle *client, ikspak *pak)
ast_copy_string(p->video_content_name, iks_find_attrib(content, "name"), sizeof(p->video_content_name));
while (codec) {
- ast_rtp_set_m_type(p->rtp, atoi(iks_find_attrib(codec, "id")));
- ast_rtp_set_rtpmap_type(p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
+ ast_rtp_codecs_payloads_set_m_type(ast_rtp_instance_get_codecs(p->rtp), p->rtp, atoi(iks_find_attrib(codec, "id")));
+ ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(p->rtp), p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
codec = iks_next(codec);
}
}
@@ -1079,7 +1072,7 @@ static int jingle_update_stun(struct jingle *client, struct jingle_pvt *p)
sin.sin_port = htons(tmp->port);
snprintf(username, sizeof(username), "%s:%s", tmp->ufrag, p->ourcandidates->ufrag);
- ast_rtp_stun_request(p->rtp, &sin, username);
+ ast_rtp_instance_stun_request(p->rtp, &sin, username);
tmp = tmp->next;
}
return 1;
@@ -1169,7 +1162,7 @@ static struct ast_frame *jingle_rtp_read(struct ast_channel *ast, struct jingle_
if (!p->rtp)
return &ast_null_frame;
- f = ast_rtp_read(p->rtp);
+ f = ast_rtp_instance_read(p->rtp, 0);
jingle_update_stun(p->parent, p);
if (p->owner) {
/* We already hold the channel lock */
@@ -1220,7 +1213,7 @@ static int jingle_write(struct ast_channel *ast, struct ast_frame *frame)
if (p) {
ast_mutex_lock(&p->lock);
if (p->rtp) {
- res = ast_rtp_write(p->rtp, frame);
+ res = ast_rtp_instance_write(p->rtp, frame);
}
ast_mutex_unlock(&p->lock);
}
@@ -1229,7 +1222,7 @@ static int jingle_write(struct ast_channel *ast, struct ast_frame *frame)
if (p) {
ast_mutex_lock(&p->lock);
if (p->vrtp) {
- res = ast_rtp_write(p->vrtp, frame);
+ res = ast_rtp_instance_write(p->vrtp, frame);
}
ast_mutex_unlock(&p->lock);
}
@@ -1879,7 +1872,7 @@ static int load_module(void)
return 0;
}
- ast_rtp_proto_register(&jingle_rtp);
+ ast_rtp_glue_register(&jingle_rtp_glue);
ast_cli_register_multiple(jingle_cli, ARRAY_LEN(jingle_cli));
/* Make sure we can register our channel type */
if (ast_channel_register(&jingle_tech)) {
@@ -1902,7 +1895,7 @@ static int unload_module(void)
ast_cli_unregister_multiple(jingle_cli, ARRAY_LEN(jingle_cli));
/* First, take us out of the channel loop */
ast_channel_unregister(&jingle_tech);
- ast_rtp_proto_unregister(&jingle_rtp);
+ ast_rtp_glue_unregister(&jingle_rtp_glue);
if (!ast_mutex_lock(&jinglelock)) {
/* Hangup all interfaces if they have an owner */
diff --git a/channels/chan_local.c b/channels/chan_local.c
index de161d6afd..e426e10fa3 100644
--- a/channels/chan_local.c
+++ b/channels/chan_local.c
@@ -39,7 +39,6 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/pbx.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
-#include "asterisk/rtp.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
#include "asterisk/file.h"
diff --git a/channels/chan_mgcp.c b/channels/chan_mgcp.c
index 1c14829756..cad9d94971 100644
--- a/channels/chan_mgcp.c
+++ b/channels/chan_mgcp.c
@@ -52,7 +52,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/pbx.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
-#include "asterisk/rtp.h"
+#include "asterisk/rtp_engine.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
#include "asterisk/cli.h"
@@ -282,7 +282,7 @@ struct mgcp_subchannel {
int id;
struct ast_channel *owner;
struct mgcp_endpoint *parent;
- struct ast_rtp *rtp;
+ struct ast_rtp_instance *rtp;
struct sockaddr_in tmpdest;
char txident[80]; /*! \todo FIXME txident is replaced by rqnt_ident in endpoint.
This should be obsoleted */
@@ -408,7 +408,7 @@ static int transmit_response(struct mgcp_subchannel *sub, char *msg, struct mgcp
static int transmit_notify_request(struct mgcp_subchannel *sub, char *tone);
static int transmit_modify_request(struct mgcp_subchannel *sub);
static int transmit_notify_request_with_callerid(struct mgcp_subchannel *sub, char *tone, char *callernum, char *callername);
-static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp *rtp, int codecs);
+static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp_instance *rtp, int codecs);
static int transmit_connection_del(struct mgcp_subchannel *sub);
static int transmit_audit_endpoint(struct mgcp_endpoint *p);
static void start_rtp(struct mgcp_subchannel *sub);
@@ -447,7 +447,7 @@ static const struct ast_channel_tech mgcp_tech = {
.fixup = mgcp_fixup,
.send_digit_begin = mgcp_senddigit_begin,
.send_digit_end = mgcp_senddigit_end,
- .bridge = ast_rtp_bridge,
+ .bridge = ast_rtp_instance_bridge,
};
static void mwi_event_cb(const struct ast_event *event, void *userdata)
@@ -503,7 +503,7 @@ static int unalloc_sub(struct mgcp_subchannel *sub)
sub->alreadygone = 0;
memset(&sub->tmpdest, 0, sizeof(sub->tmpdest));
if (sub->rtp) {
- ast_rtp_destroy(sub->rtp);
+ ast_rtp_instance_destroy(sub->rtp);
sub->rtp = NULL;
}
dump_cmd_queues(NULL, sub); /* SC */
@@ -1003,7 +1003,7 @@ static int mgcp_hangup(struct ast_channel *ast)
/* Reset temporary destination */
memset(&sub->tmpdest, 0, sizeof(sub->tmpdest));
if (sub->rtp) {
- ast_rtp_destroy(sub->rtp);
+ ast_rtp_instance_destroy(sub->rtp);
sub->rtp = NULL;
}
@@ -1203,7 +1203,7 @@ static struct ast_frame *mgcp_rtp_read(struct mgcp_subchannel *sub)
/* Retrieve audio/etc from channel. Assumes sub->lock is already held. */
struct ast_frame *f;
- f = ast_rtp_read(sub->rtp);
+ f = ast_rtp_instance_read(sub->rtp, 0);
/* Don't send RFC2833 if we're not supposed to */
if (f && (f->frametype == AST_FRAME_DTMF) && !(sub->parent->dtmfmode & MGCP_DTMF_RFC2833))
return &ast_null_frame;
@@ -1261,7 +1261,7 @@ static int mgcp_write(struct ast_channel *ast, struct ast_frame *frame)
ast_mutex_lock(&sub->lock);
if ((sub->parent->sub == sub) || !sub->parent->singlepath) {
if (sub->rtp) {
- res = ast_rtp_write(sub->rtp, frame);
+ res = ast_rtp_instance_write(sub->rtp, frame);
}
}
ast_mutex_unlock(&sub->lock);
@@ -1297,7 +1297,7 @@ static int mgcp_senddigit_begin(struct ast_channel *ast, char digit)
res = -1; /* Let asterisk play inband indications */
} else if (p->dtmfmode & MGCP_DTMF_RFC2833) {
ast_log(LOG_DEBUG, "Sending DTMF using RFC2833");
- ast_rtp_senddigit_begin(sub->rtp, digit);
+ ast_rtp_instance_dtmf_begin(sub->rtp, digit);
} else {
ast_log(LOG_ERROR, "Don't know about DTMF_MODE %d\n", p->dtmfmode);
}
@@ -1324,7 +1324,7 @@ static int mgcp_senddigit_end(struct ast_channel *ast, char digit, unsigned int
tmp[2] = digit;
tmp[3] = '\0';
transmit_notify_request(sub, tmp);
- ast_rtp_senddigit_end(sub->rtp, digit);
+ ast_rtp_instance_dtmf_end(sub->rtp, digit);
} else {
ast_log(LOG_ERROR, "Don't know about DTMF_MODE %d\n", p->dtmfmode);
}
@@ -1453,7 +1453,7 @@ static int mgcp_indicate(struct ast_channel *ast, int ind, const void *data, siz
ast_moh_stop(ast);
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_new_source(sub->rtp);
+ ast_rtp_instance_new_source(sub->rtp);
break;
case -1:
transmit_notify_request(sub, "");
@@ -1481,7 +1481,7 @@ static struct ast_channel *mgcp_new(struct mgcp_subchannel *sub, int state)
fmt = ast_best_codec(tmp->nativeformats);
ast_string_field_build(tmp, name, "MGCP/%s@%s-%d", i->name, i->parent->name, sub->id);
if (sub->rtp)
- ast_channel_set_fd(tmp, 0, ast_rtp_fd(sub->rtp));
+ ast_channel_set_fd(tmp, 0, ast_rtp_instance_fd(sub->rtp, 0));
if (i->dtmfmode & (MGCP_DTMF_INBAND | MGCP_DTMF_HYBRID)) {
i->dsp = ast_dsp_new();
ast_dsp_set_features(i->dsp, DSP_FEATURE_DIGIT_DETECT);
@@ -1874,12 +1874,12 @@ static int process_sdp(struct mgcp_subchannel *sub, struct mgcp_request *req)
sin.sin_family = AF_INET;
memcpy(&sin.sin_addr, hp->h_addr, sizeof(sin.sin_addr));
sin.sin_port = htons(portno);
- ast_rtp_set_peer(sub->rtp, &sin);
+ ast_rtp_instance_set_remote_address(sub->rtp, &sin);
#if 0
printf("Peer RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
#endif
/* Scan through the RTP payload types specified in a "m=" line: */
- ast_rtp_pt_clear(sub->rtp);
+ ast_rtp_codecs_payloads_clear(ast_rtp_instance_get_codecs(sub->rtp), sub->rtp);
codecs = ast_strdupa(m + len);
while (!ast_strlen_zero(codecs)) {
if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
@@ -1888,7 +1888,7 @@ static int process_sdp(struct mgcp_subchannel *sub, struct mgcp_request *req)
ast_log(LOG_WARNING, "Error in codec string '%s' at '%s'\n", m, codecs);
return -1;
}
- ast_rtp_set_m_type(sub->rtp, codec);
+ ast_rtp_codecs_payloads_set_m_type(ast_rtp_instance_get_codecs(sub->rtp), sub->rtp, codec);
codec_count++;
codecs += len;
}
@@ -1901,11 +1901,11 @@ static int process_sdp(struct mgcp_subchannel *sub, struct mgcp_request *req)
if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2)
continue;
/* Note: should really look at the 'freq' and '#chans' params too */
- ast_rtp_set_rtpmap_type(sub->rtp, codec, "audio", mimeSubtype, 0);
+ ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(sub->rtp), sub->rtp, codec, "audio", mimeSubtype, 0);
}
/* Now gather all of the codecs that were asked for: */
- ast_rtp_get_current_formats(sub->rtp, &peercapability, &peerNonCodecCapability);
+ ast_rtp_codecs_payload_formats(ast_rtp_instance_get_codecs(sub->rtp), &peercapability, &peerNonCodecCapability);
p->capability = capability & peercapability;
if (mgcpdebug) {
ast_verbose("Capabilities: us - %d, them - %d, combined - %d\n",
@@ -2043,7 +2043,7 @@ static int transmit_response(struct mgcp_subchannel *sub, char *msg, struct mgcp
}
-static int add_sdp(struct mgcp_request *resp, struct mgcp_subchannel *sub, struct ast_rtp *rtp)
+static int add_sdp(struct mgcp_request *resp, struct mgcp_subchannel *sub, struct ast_rtp_instance *rtp)
{
int len;
int codec;
@@ -2066,9 +2066,9 @@ static int add_sdp(struct mgcp_request *resp, struct mgcp_subchannel *sub, struc
ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
return -1;
}
- ast_rtp_get_us(sub->rtp, &sin);
+ ast_rtp_instance_get_local_address(sub->rtp, &sin);
if (rtp) {
- ast_rtp_get_peer(rtp, &dest);
+ ast_rtp_instance_get_remote_address(sub->rtp, &dest);
} else {
if (sub->tmpdest.sin_addr.s_addr) {
dest.sin_addr = sub->tmpdest.sin_addr;
@@ -2094,11 +2094,11 @@ static int add_sdp(struct mgcp_request *resp, struct mgcp_subchannel *sub, struc
if (mgcpdebug) {
ast_verbose("Answering with capability %d\n", x);
}
- codec = ast_rtp_lookup_code(sub->rtp, 1, x);
+ codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(sub->rtp), 1, x);
if (codec > -1) {
snprintf(costr, sizeof(costr), " %d", codec);
strncat(m, costr, sizeof(m) - strlen(m) - 1);
- snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(1, x, 0));
+ snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype2(1, x, 0));
strncat(a, costr, sizeof(a) - strlen(a) - 1);
}
}
@@ -2108,11 +2108,11 @@ static int add_sdp(struct mgcp_request *resp, struct mgcp_subchannel *sub, struc
if (mgcpdebug) {
ast_verbose("Answering with non-codec capability %d\n", x);
}
- codec = ast_rtp_lookup_code(sub->rtp, 0, x);
+ codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(sub->rtp), 0, x);
if (codec > -1) {
snprintf(costr, sizeof(costr), " %d", codec);
strncat(m, costr, sizeof(m) - strlen(m) - 1);
- snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(0, x, 0));
+ snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype2(0, x, 0));
strncat(a, costr, sizeof(a) - strlen(a) - 1);
if (x == AST_RTP_DTMF) {
/* Indicate we support DTMF... Not sure about 16,
@@ -2136,7 +2136,7 @@ static int add_sdp(struct mgcp_request *resp, struct mgcp_subchannel *sub, struc
return 0;
}
-static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp *rtp, int codecs)
+static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp_instance *rtp, int codecs)
{
struct mgcp_request resp;
char local[256];
@@ -2147,13 +2147,13 @@ static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp
if (ast_strlen_zero(sub->cxident) && rtp) {
/* We don't have a CXident yet, store the destination and
wait a bit */
- ast_rtp_get_peer(rtp, &sub->tmpdest);
+ ast_rtp_instance_get_remote_address(rtp, &sub->tmpdest);
return 0;
}
ast_copy_string(local, "p:20", sizeof(local));
for (x = 1; x <= AST_FORMAT_AUDIO_MASK; x <<= 1) {
if (p->capability & x) {
- snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype(1, x, 0));
+ snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype2(1, x, 0));
strncat(local, tmp, sizeof(local) - strlen(local) - 1);
}
}
@@ -2172,7 +2172,7 @@ static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp
return send_request(p, sub, &resp, oseq); /* SC */
}
-static int transmit_connect_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp *rtp)
+static int transmit_connect_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp_instance *rtp)
{
struct mgcp_request resp;
char local[256];
@@ -2183,7 +2183,7 @@ static int transmit_connect_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp
ast_copy_string(local, "p:20", sizeof(local));
for (x = 1; x <= AST_FORMAT_AUDIO_MASK; x <<= 1) {
if (p->capability & x) {
- snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype(1, x, 0));
+ snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype2(1, x, 0));
strncat(local, tmp, sizeof(local) - strlen(local) - 1);
}
}
@@ -2611,21 +2611,17 @@ static void start_rtp(struct mgcp_subchannel *sub)
ast_mutex_lock(&sub->lock);
/* check again to be on the safe side */
if (sub->rtp) {
- ast_rtp_destroy(sub->rtp);
+ ast_rtp_instance_destroy(sub->rtp);
sub->rtp = NULL;
}
/* Allocate the RTP now */
- sub->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
+ sub->rtp = ast_rtp_instance_new(NULL, sched, &bindaddr, NULL);
if (sub->rtp && sub->owner)
- ast_channel_set_fd(sub->owner, 0, ast_rtp_fd(sub->rtp));
+ ast_channel_set_fd(sub->owner, 0, ast_rtp_instance_fd(sub->rtp, 0));
if (sub->rtp) {
- ast_rtp_setqos(sub->rtp, qos.tos_audio, qos.cos_audio, "MGCP RTP");
- ast_rtp_setnat(sub->rtp, sub->nat);
+ ast_rtp_instance_set_qos(sub->rtp, qos.tos_audio, qos.cos_audio, "MGCP RTP");
+ ast_rtp_instance_set_prop(sub->rtp, AST_RTP_PROPERTY_NAT, sub->nat);
}
-#if 0
- ast_rtp_set_callback(p->rtp, rtpready);
- ast_rtp_set_data(p->rtp, p);
-#endif
/* Make a call*ID */
snprintf(sub->callid, sizeof(sub->callid), "%08lx%s", ast_random(), sub->txident);
/* Transmit the connection create */
@@ -3940,22 +3936,22 @@ static struct mgcp_gateway *build_gateway(char *cat, struct ast_variable *v)
return (gw_reload ? NULL : gw);
}
-static enum ast_rtp_get_result mgcp_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
+static enum ast_rtp_glue_result mgcp_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
struct mgcp_subchannel *sub = NULL;
if (!(sub = chan->tech_pvt) || !(sub->rtp))
- return AST_RTP_GET_FAILED;
+ return AST_RTP_GLUE_RESULT_FORBID;
- *rtp = sub->rtp;
+ *instance = sub->rtp ? ao2_ref(sub->rtp, +1), sub->rtp : NULL;
if (sub->parent->canreinvite)
- return AST_RTP_TRY_NATIVE;
+ return AST_RTP_GLUE_RESULT_REMOTE;
else
- return AST_RTP_TRY_PARTIAL;
+ return AST_RTP_GLUE_RESULT_LOCAL;
}
-static int mgcp_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active)
+static int mgcp_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *trtp, int codecs, int nat_active)
{
/* XXX Is there such thing as video support with MGCP? XXX */
struct mgcp_subchannel *sub;
@@ -3967,10 +3963,10 @@ static int mgcp_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, stru
return -1;
}
-static struct ast_rtp_protocol mgcp_rtp = {
+static struct ast_rtp_glue mgcp_rtp_glue = {
.type = "MGCP",
.get_rtp_info = mgcp_get_rtp_peer,
- .set_rtp_peer = mgcp_set_rtp_peer,
+ .update_peer = mgcp_set_rtp_peer,
};
static void destroy_endpoint(struct mgcp_endpoint *e)
@@ -3984,7 +3980,7 @@ static void destroy_endpoint(struct mgcp_endpoint *e)
transmit_connection_del(sub);
}
if (sub->rtp) {
- ast_rtp_destroy(sub->rtp);
+ ast_rtp_instance_destroy(sub->rtp);
sub->rtp = NULL;
}
memset(sub->magic, 0, sizeof(sub->magic));
@@ -4276,7 +4272,7 @@ static int load_module(void)
return AST_MODULE_LOAD_FAILURE;
}
- ast_rtp_proto_register(&mgcp_rtp);
+ ast_rtp_glue_register(&mgcp_rtp_glue);
ast_cli_register_multiple(cli_mgcp, sizeof(cli_mgcp) / sizeof(struct ast_cli_entry));
/* And start the monitor for the first time */
@@ -4379,7 +4375,7 @@ static int unload_module(void)
}
close(mgcpsock);
- ast_rtp_proto_unregister(&mgcp_rtp);
+ ast_rtp_glue_unregister(&mgcp_rtp_glue);
ast_cli_unregister_multiple(cli_mgcp, sizeof(cli_mgcp) / sizeof(struct ast_cli_entry));
sched_context_destroy(sched);
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 8afe7766a2..4d0f06f4ad 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -229,7 +229,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/pbx.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
-#include "asterisk/rtp.h"
+#include "asterisk/rtp_engine.h"
#include "asterisk/udptl.h"
#include "asterisk/acl.h"
#include "asterisk/manager.h"
@@ -271,6 +271,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/ast_version.h"
#include "asterisk/event.h"
#include "asterisk/tcptls.h"
+#include "asterisk/stun.h"
/*** DOCUMENTATION
<application name="SIPDtmfMode" language="en_US">
@@ -691,6 +692,7 @@ enum check_auth_result {
AUTH_PEER_NOT_DYNAMIC = -6,
AUTH_ACL_FAILED = -7,
AUTH_BAD_TRANSPORT = -8,
+ AUTH_RTP_FAILED = 9,
};
/*! \brief States for outbound registrations (with register= lines in sip.conf */
@@ -1011,6 +1013,7 @@ static const struct cfsip_options {
#define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */
#define DEFAULT_SDPSESSION "Asterisk PBX" /*!< Default SDP session name, (s=) header unless re-defined in sip.conf */
#define DEFAULT_SDPOWNER "root" /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
+#define DEFAULT_ENGINE "asterisk" /*!< Default RTP engine to use for sessions */
#endif
/*@}*/
@@ -1029,6 +1032,7 @@ static char default_mohinterpret[MAX_MUSICCLASS]; /*!< Global setting for moh c
static char default_mohsuggest[MAX_MUSICCLASS]; /*!< Global setting for moh class to suggest when putting
* a bridged channel on hold */
static char default_parkinglot[AST_MAX_CONTEXT]; /*!< Parkinglot */
+static char default_engine[256]; /*!< Default RTP engine */
static int default_maxcallbitrate; /*!< Maximum bitrate for call */
static struct ast_codec_pref default_prefs; /*!< Default codec prefs */
static unsigned int default_transports; /*!< Default Transports (enum sip_transport) that are acceptable */
@@ -1611,6 +1615,7 @@ struct sip_pvt {
AST_STRING_FIELD(rpid_from); /*!< Our RPID From header */
AST_STRING_FIELD(url); /*!< URL to be sent with next message to peer */
AST_STRING_FIELD(parkinglot); /*!< Parkinglot */
+ AST_STRING_FIELD(engine); /*!< RTP engine to use */
);
char via[128]; /*!< Via: header */
struct sip_socket socket; /*!< The socket used for this dialog */
@@ -1699,9 +1704,9 @@ struct sip_pvt {
struct sip_peer *relatedpeer; /*!< If this dialog is related to a peer, which one
Used in peerpoke, mwi subscriptions */
struct sip_registry *registry; /*!< If this is a REGISTER dialog, to which registry */
- struct ast_rtp *rtp; /*!< RTP Session */
- struct ast_rtp *vrtp; /*!< Video RTP session */
- struct ast_rtp *trtp; /*!< Text RTP session */
+ struct ast_rtp_instance *rtp; /*!< RTP Session */
+ struct ast_rtp_instance *vrtp; /*!< Video RTP session */
+ struct ast_rtp_instance *trtp; /*!< Text RTP session */
struct sip_pkt *packets; /*!< Packets scheduled for re-transmission */
struct sip_history_head *history; /*!< History of this SIP dialog */
size_t history_entries; /*!< Number of entires in the history */
@@ -1844,6 +1849,7 @@ struct sip_peer {
AST_STRING_FIELD(mohsuggest); /*!< Music on Hold class */
AST_STRING_FIELD(parkinglot); /*!< Parkinglot */
AST_STRING_FIELD(useragent); /*!< User agent in SIP request (saved from registration) */
+ AST_STRING_FIELD(engine); /*!< RTP Engine to use */
);
struct sip_socket socket; /*!< Socket used for this peer */
unsigned int transports:3; /*!< Transports (enum sip_transport) that are acceptable for this peer */
@@ -2564,14 +2570,6 @@ static void handle_response_subscribe(struct sip_pvt *p, int resp, char *rest, s
static int handle_response_register(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
-/*----- RTP interface functions */
-static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active);
-static enum ast_rtp_get_result sip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
-static enum ast_rtp_get_result sip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
-static enum ast_rtp_get_result sip_get_trtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
-static int sip_get_codec(struct ast_channel *chan);
-static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p, int *faxdetect);
-
/*------ T38 Support --------- */
static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan);
@@ -2592,6 +2590,9 @@ static enum st_refresher st_get_refresher(struct sip_pvt *);
static enum st_mode st_get_mode(struct sip_pvt *);
static struct sip_st_dlg* sip_st_alloc(struct sip_pvt *const p);
+/*------- RTP Glue functions -------- */
+static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, int codecs, int nat_active);
+
/*!--- SIP MWI Subscription support */
static int sip_subscribe_mwi(const char *value, int lineno);
static void sip_subscribe_mwi_destroy(struct sip_subscription_mwi *mwi);
@@ -2620,8 +2621,8 @@ static const struct ast_channel_tech sip_tech = {
.fixup = sip_fixup, /* called with chan locked */
.send_digit_begin = sip_senddigit_begin, /* called with chan unlocked */
.send_digit_end = sip_senddigit_end,
- .bridge = ast_rtp_bridge, /* XXX chan unlocked ? */
- .early_bridge = ast_rtp_early_bridge,
+ .bridge = ast_rtp_instance_bridge, /* XXX chan unlocked ? */
+ .early_bridge = ast_rtp_instance_early_bridge,
.send_text = sip_sendtext, /* called with chan locked */
.func_channel_read = acf_channel_read,
.queryoption = sip_queryoption,
@@ -2694,17 +2695,6 @@ static int map_s_x(const struct _map_x_s *table, const char *s, int errorvalue)
return errorvalue;
}
-
-/*! \brief Interface structure with callbacks used to connect to RTP module */
-static struct ast_rtp_protocol sip_rtp = {
- .type = "SIP",
- .get_rtp_info = sip_get_rtp_peer,
- .get_vrtp_info = sip_get_vrtp_peer,
- .get_trtp_info = sip_get_trtp_peer,
- .set_rtp_peer = sip_set_rtp_peer,
- .get_codec = sip_get_codec,
-};
-
/*!
* duplicate a list of channel variables, \return the copy.
*/
@@ -4593,11 +4583,11 @@ static void do_setnat(struct sip_pvt *p, int natflags)
if (p->rtp) {
ast_debug(1, "Setting NAT on RTP to %s\n", mode);
- ast_rtp_setnat(p->rtp, natflags);
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_NAT, natflags);
}
if (p->vrtp) {
ast_debug(1, "Setting NAT on VRTP to %s\n", mode);
- ast_rtp_setnat(p->vrtp, natflags);
+ ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_NAT, natflags);
}
if (p->udptl) {
ast_debug(1, "Setting NAT on UDPTL to %s\n", mode);
@@ -4605,7 +4595,7 @@ static void do_setnat(struct sip_pvt *p, int natflags)
}
if (p->trtp) {
ast_debug(1, "Setting NAT on TRTP to %s\n", mode);
- ast_rtp_setnat(p->trtp, natflags);
+ ast_rtp_instance_set_prop(p->trtp, AST_RTP_PROPERTY_NAT, natflags);
}
}
@@ -4697,6 +4687,51 @@ static void copy_socket_data(struct sip_socket *to_sock, const struct sip_socket
*to_sock = *from_sock;
}
+/*! \brief Initialize RTP portion of a dialog
+ * \returns -1 on failure, 0 on success
+ */
+static int dialog_initialize_rtp(struct sip_pvt *dialog)
+{
+ if (!sip_methods[dialog->method].need_rtp) {
+ return 0;
+ }
+
+ if (!(dialog->rtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr, NULL))) {
+ return -1;
+ }
+
+ if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT) && (dialog->capability & AST_FORMAT_VIDEO_MASK)) {
+ if (!(dialog->vrtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr, NULL))) {
+ return -1;
+ }
+ ast_rtp_instance_set_timeout(dialog->vrtp, global_rtptimeout);
+ ast_rtp_instance_set_hold_timeout(dialog->vrtp, global_rtpholdtimeout);
+
+ ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_RTCP, 1);
+ }
+
+ if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_TEXTSUPPORT)) {
+ if (!(dialog->trtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr, NULL))) {
+ return -1;
+ }
+ ast_rtp_instance_set_timeout(dialog->trtp, global_rtptimeout);
+ ast_rtp_instance_set_hold_timeout(dialog->trtp, global_rtpholdtimeout);
+
+ ast_rtp_instance_set_prop(dialog->trtp, AST_RTP_PROPERTY_RTCP, 1);
+ }
+
+ ast_rtp_instance_set_timeout(dialog->rtp, global_rtptimeout);
+ ast_rtp_instance_set_hold_timeout(dialog->rtp, global_rtpholdtimeout);
+
+ ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_RTCP, 1);
+ ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
+ ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
+
+ ast_rtp_instance_set_qos(dialog->rtp, global_tos_audio, 0, "SIP RTP");
+
+ return 0;
+}
+
/*! \brief Create address structure from peer reference.
* This function copies data from peer to the dialog, so we don't have to look up the peer
* again from memory or database during the life time of the dialog.
@@ -4724,17 +4759,6 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
ast_copy_flags(&dialog->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
ast_copy_flags(&dialog->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
dialog->capability = peer->capability;
- if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS) &&
- (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT) ||
- !(dialog->capability & AST_FORMAT_VIDEO_MASK)) &&
- dialog->vrtp) {
- ast_rtp_destroy(dialog->vrtp);
- dialog->vrtp = NULL;
- }
- if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_TEXTSUPPORT) && dialog->trtp) {
- ast_rtp_destroy(dialog->trtp);
- dialog->trtp = NULL;
- }
dialog->prefs = peer->prefs;
if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_T38SUPPORT)) {
if (!dialog->udptl) {
@@ -4750,29 +4774,28 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
}
do_setnat(dialog, ast_test_flag(&dialog->flags[0], SIP_NAT) & SIP_NAT_ROUTE);
+ ast_string_field_set(dialog, engine, peer->engine);
+
+ if (dialog_initialize_rtp(dialog)) {
+ return -1;
+ }
+
if (dialog->rtp) { /* Audio */
- ast_rtp_setdtmf(dialog->rtp, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
- ast_rtp_setdtmfcompensate(dialog->rtp, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
- ast_rtp_set_rtptimeout(dialog->rtp, peer->rtptimeout);
- ast_rtp_set_rtpholdtimeout(dialog->rtp, peer->rtpholdtimeout);
- ast_rtp_set_rtpkeepalive(dialog->rtp, peer->rtpkeepalive);
+ ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
+ ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
+ ast_rtp_instance_set_timeout(dialog->rtp, peer->rtptimeout);
+ ast_rtp_instance_set_hold_timeout(dialog->rtp, peer->rtpholdtimeout);
/* Set Frame packetization */
- ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(dialog->rtp), dialog->rtp, &dialog->prefs);
dialog->autoframing = peer->autoframing;
}
if (dialog->vrtp) { /* Video */
- ast_rtp_setdtmf(dialog->vrtp, 0);
- ast_rtp_setdtmfcompensate(dialog->vrtp, 0);
- ast_rtp_set_rtptimeout(dialog->vrtp, peer->rtptimeout);
- ast_rtp_set_rtpholdtimeout(dialog->vrtp, peer->rtpholdtimeout);
- ast_rtp_set_rtpkeepalive(dialog->vrtp, peer->rtpkeepalive);
+ ast_rtp_instance_set_timeout(dialog->vrtp, peer->rtptimeout);
+ ast_rtp_instance_set_hold_timeout(dialog->vrtp, peer->rtpholdtimeout);
}
if (dialog->trtp) { /* Realtime text */
- ast_rtp_setdtmf(dialog->trtp, 0);
- ast_rtp_setdtmfcompensate(dialog->trtp, 0);
- ast_rtp_set_rtptimeout(dialog->trtp, peer->rtptimeout);
- ast_rtp_set_rtpholdtimeout(dialog->trtp, peer->rtpholdtimeout);
- ast_rtp_set_rtpkeepalive(dialog->trtp, peer->rtpkeepalive);
+ ast_rtp_instance_set_timeout(dialog->trtp, peer->rtptimeout);
+ ast_rtp_instance_set_hold_timeout(dialog->trtp, peer->rtpholdtimeout);
}
ast_string_field_set(dialog, peername, peer->name);
@@ -4786,6 +4809,7 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
ast_string_field_set(dialog, fullcontact, peer->fullcontact);
ast_string_field_set(dialog, context, peer->context);
ast_string_field_set(dialog, parkinglot, peer->parkinglot);
+ ast_string_field_set(dialog, engine, peer->engine);
ref_proxy(dialog, obproxy_get(dialog, peer));
dialog->callgroup = peer->callgroup;
dialog->pickupgroup = peer->pickupgroup;
@@ -4881,6 +4905,10 @@ static int create_addr(struct sip_pvt *dialog, const char *opeer, struct sockadd
return res;
}
+ if (dialog_initialize_rtp(dialog)) {
+ return -1;
+ }
+
do_setnat(dialog, ast_test_flag(&dialog->flags[0], SIP_NAT) & SIP_NAT_ROUTE);
ast_string_field_set(dialog, tohost, peername);
@@ -5155,15 +5183,13 @@ static void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist)
p->notify_headers = NULL;
}
if (p->rtp) {
- ast_rtp_destroy(p->rtp);
+ ast_rtp_instance_destroy(p->rtp);
}
if (p->vrtp) {
- ast_rtp_destroy(p->vrtp);
+ ast_rtp_instance_destroy(p->vrtp);
}
if (p->trtp) {
- while (ast_rtp_get_bridged(p->trtp))
- usleep(1);
- ast_rtp_destroy(p->trtp);
+ ast_rtp_instance_destroy(p->trtp);
}
if (p->udptl)
ast_udptl_destroy(p->udptl);
@@ -5682,42 +5708,50 @@ static int sip_hangup(struct ast_channel *ast)
if (!p->pendinginvite) {
struct ast_channel *bridge = ast_bridged_channel(oldowner);
- char *audioqos = "";
- char *videoqos = "";
- char *textqos = "";
+ char quality_buf[AST_MAX_USER_FIELD], *quality;
- if (p->rtp)
- ast_rtp_set_vars(oldowner, p->rtp);
+ if (p->rtp) {
+ ast_rtp_instance_set_stats_vars(oldowner, p->rtp);
+ }
if (bridge) {
struct sip_pvt *q = bridge->tech_pvt;
- if (IS_SIP_TECH(bridge->tech) && q)
- ast_rtp_set_vars(bridge, q->rtp);
+ if (IS_SIP_TECH(bridge->tech) && q) {
+ ast_rtp_instance_set_stats_vars(bridge, q->rtp);
+ }
+ }
+
+ if (p->do_history || oldowner) {
+ if (p->rtp && (quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
+ if (p->do_history) {
+ append_history(p, "RTCPaudio", "Quality:%s", quality);
+ }
+ if (oldowner) {
+ pbx_builtin_setvar_helper(oldowner, "RTPAUDIOQOS", quality);
+ }
+ }
+ if (p->vrtp && (quality = ast_rtp_instance_get_quality(p->vrtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
+ if (p->do_history) {
+ append_history(p, "RTCPvideo", "Quality:%s", quality);
+ }
+ if (oldowner) {
+ pbx_builtin_setvar_helper(oldowner, "RTPVIDEOQOS", quality);
+ }
+ }
+ if (p->trtp && (quality = ast_rtp_instance_get_quality(p->trtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
+ if (p->do_history) {
+ append_history(p, "RTCPtext", "Quality:%s", quality);
+ }
+ if (oldowner) {
+ pbx_builtin_setvar_helper(oldowner, "RTPTEXTQOS", quality);
+ }
+ }
}
- if (p->vrtp)
- videoqos = ast_rtp_get_quality(p->vrtp, NULL, RTPQOS_SUMMARY);
- if (p->trtp)
- textqos = ast_rtp_get_quality(p->trtp, NULL, RTPQOS_SUMMARY);
/* Send a hangup */
transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
- /* Get RTCP quality before end of call */
- if (p->do_history) {
- if (p->rtp)
- append_history(p, "RTCPaudio", "Quality:%s", audioqos);
- if (p->vrtp)
- append_history(p, "RTCPvideo", "Quality:%s", videoqos);
- if (p->trtp)
- append_history(p, "RTCPtext", "Quality:%s", textqos);
- }
- if (p->rtp && oldowner)
- pbx_builtin_setvar_helper(oldowner, "RTPAUDIOQOS", audioqos);
- if (p->vrtp && oldowner)
- pbx_builtin_setvar_helper(oldowner, "RTPVIDEOQOS", videoqos);
- if (p->trtp && oldowner)
- pbx_builtin_setvar_helper(oldowner, "RTPTEXTQOS", textqos);
} else {
/* Note we will need a BYE when this all settles out
but we can't send one while we have "INVITE" outstanding. */
@@ -5772,7 +5806,10 @@ static int sip_answer(struct ast_channel *ast)
ast_setstate(ast, AST_STATE_UP);
ast_debug(1, "SIP answering channel: %s\n", ast->name);
- ast_rtp_new_source(p->rtp);
+ if (p->t38.state == T38_PEER_DIRECT) {
+ change_t38_state(p, T38_ENABLED);
+ }
+ ast_rtp_instance_new_source(p->rtp);
res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE);
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
}
@@ -5807,7 +5844,7 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
if ((ast->_state != AST_STATE_UP) &&
!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
- ast_rtp_new_source(p->rtp);
+ ast_rtp_instance_new_source(p->rtp);
p->invitestate = INV_EARLY_MEDIA;
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE, FALSE);
ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
@@ -5816,7 +5853,7 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
transmit_reinvite_with_sdp(p, FALSE, FALSE);
} else {
p->lastrtptx = time(NULL);
- res = ast_rtp_write(p->rtp, frame);
+ res = ast_rtp_instance_write(p->rtp, frame);
}
}
sip_pvt_unlock(p);
@@ -5835,7 +5872,7 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
}
p->lastrtptx = time(NULL);
- res = ast_rtp_write(p->vrtp, frame);
+ res = ast_rtp_instance_write(p->vrtp, frame);
}
sip_pvt_unlock(p);
}
@@ -5844,7 +5881,7 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
if (p) {
sip_pvt_lock(p);
if (p->red) {
- ast_red_buffer_t140(p->trtp, frame);
+ ast_rtp_red_buffer(p->trtp, frame);
} else {
if (p->trtp) {
/* Activate text early media */
@@ -5856,7 +5893,7 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
}
p->lastrtptx = time(NULL);
- res = ast_rtp_write(p->trtp, frame);
+ res = ast_rtp_instance_write(p->trtp, frame);
}
}
sip_pvt_unlock(p);
@@ -5944,11 +5981,15 @@ static int sip_senddigit_begin(struct ast_channel *ast, char digit)
sip_pvt_lock(p);
switch (ast_test_flag(&p->flags[0], SIP_DTMF)) {
case SIP_DTMF_INBAND:
- res = -1; /* Tell Asterisk to generate inband indications */
+ if (p->rtp && ast_rtp_instance_dtmf_mode_get(p->rtp) == AST_RTP_DTMF_MODE_INBAND) {
+ ast_rtp_instance_dtmf_begin(p->rtp, digit);
+ } else {
+ res = -1; /* Tell Asterisk to generate inband indications */
+ }
break;
case SIP_DTMF_RFC2833:
if (p->rtp)
- ast_rtp_senddigit_begin(p->rtp, digit);
+ ast_rtp_instance_dtmf_begin(p->rtp, digit);
break;
default:
break;
@@ -5973,10 +6014,14 @@ static int sip_senddigit_end(struct ast_channel *ast, char digit, unsigned int d
break;
case SIP_DTMF_RFC2833:
if (p->rtp)
- ast_rtp_senddigit_end(p->rtp, digit);
+ ast_rtp_instance_dtmf_end(p->rtp, digit);
break;
case SIP_DTMF_INBAND:
- res = -1; /* Tell Asterisk to stop inband indications */
+ if (p->rtp && ast_rtp_instance_dtmf_mode_get(p->rtp) == AST_RTP_DTMF_MODE_INBAND) {
+ ast_rtp_instance_dtmf_end(p->rtp, digit);
+ } else {
+ res = -1; /* Tell Asterisk to stop inband indications */
+ }
break;
}
sip_pvt_unlock(p);
@@ -6071,11 +6116,11 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
res = -1;
break;
case AST_CONTROL_HOLD:
- ast_rtp_new_source(p->rtp);
+ ast_rtp_instance_new_source(p->rtp);
ast_moh_start(ast, data, p->mohinterpret);
break;
case AST_CONTROL_UNHOLD:
- ast_rtp_new_source(p->rtp);
+ ast_rtp_instance_new_source(p->rtp);
ast_moh_stop(ast);
break;
case AST_CONTROL_VIDUPDATE: /* Request a video frame update */
@@ -6121,7 +6166,7 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
}
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_new_source(p->rtp);
+ ast_rtp_instance_new_source(p->rtp);
break;
case -1:
res = -1;
@@ -6235,23 +6280,29 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
ast_debug(3, "This channel will not be able to handle video.\n");
if ((ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) || (ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) {
- i->vad = ast_dsp_new();
- ast_dsp_set_features(i->vad, DSP_FEATURE_DIGIT_DETECT);
- if (global_relaxdtmf)
- ast_dsp_set_digitmode(i->vad, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_RELAXDTMF);
+ if (!i->rtp || ast_rtp_instance_dtmf_mode_set(i->rtp, AST_RTP_DTMF_MODE_INBAND)) {
+ i->vad = ast_dsp_new();
+ ast_dsp_set_features(i->vad, DSP_FEATURE_DIGIT_DETECT);
+ if (global_relaxdtmf)
+ ast_dsp_set_digitmode(i->vad, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_RELAXDTMF);
+ }
+ } else if (ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) {
+ if (i->rtp) {
+ ast_rtp_instance_dtmf_mode_set(i->rtp, AST_RTP_DTMF_MODE_RFC2833);
+ }
}
/* Set file descriptors for audio, video, realtime text and UDPTL as needed */
if (i->rtp) {
- ast_channel_set_fd(tmp, 0, ast_rtp_fd(i->rtp));
- ast_channel_set_fd(tmp, 1, ast_rtcp_fd(i->rtp));
+ ast_channel_set_fd(tmp, 0, ast_rtp_instance_fd(i->rtp, 0));
+ ast_channel_set_fd(tmp, 1, ast_rtp_instance_fd(i->rtp, 1));
}
if (needvideo && i->vrtp) {
- ast_channel_set_fd(tmp, 2, ast_rtp_fd(i->vrtp));
- ast_channel_set_fd(tmp, 3, ast_rtcp_fd(i->vrtp));
+ ast_channel_set_fd(tmp, 2, ast_rtp_instance_fd(i->vrtp, 0));
+ ast_channel_set_fd(tmp, 3, ast_rtp_instance_fd(i->vrtp, 1));
}
if (needtext && i->trtp)
- ast_channel_set_fd(tmp, 4, ast_rtp_fd(i->trtp));
+ ast_channel_set_fd(tmp, 4, ast_rtp_instance_fd(i->trtp, 0));
if (i->udptl)
ast_channel_set_fd(tmp, 5, ast_udptl_fd(i->udptl));
@@ -6475,19 +6526,19 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p
switch(ast->fdno) {
case 0:
- f = ast_rtp_read(p->rtp); /* RTP Audio */
+ f = ast_rtp_instance_read(p->rtp, 0); /* RTP Audio */
break;
case 1:
- f = ast_rtcp_read(p->rtp); /* RTCP Control Channel */
+ f = ast_rtp_instance_read(p->rtp, 1); /* RTCP Control Channel */
break;
case 2:
- f = ast_rtp_read(p->vrtp); /* RTP Video */
+ f = ast_rtp_instance_read(p->vrtp, 0); /* RTP Video */
break;
case 3:
- f = ast_rtcp_read(p->vrtp); /* RTCP Control Channel for video */
+ f = ast_rtp_instance_read(p->vrtp, 1); /* RTCP Control Channel for video */
break;
case 4:
- f = ast_rtp_read(p->trtp); /* RTP Text */
+ f = ast_rtp_instance_read(p->trtp, 0); /* RTP Text */
if (sipdebug_text) {
int i;
unsigned char* arr = f->data.ptr;
@@ -6694,50 +6745,11 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
p->ocseq = INITIAL_CSEQ;
if (sip_methods[intended_method].need_rtp) {
- p->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
- /* If the global videosupport flag is on, we always create a RTP interface for video */
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT))
- p->vrtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT))
- p->trtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT))
- p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, bindaddr.sin_addr);
- if (!p->rtp|| (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp)
- || (ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT) && !p->trtp)) {
- ast_log(LOG_WARNING, "Unable to create RTP audio %s%ssession: %s\n",
- ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "and video " : "",
- ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT) ? "and text " : "", strerror(errno));
- if (p->chanvars) {
- ast_variables_destroy(p->chanvars);
- p->chanvars = NULL;
- }
- ao2_t_ref(p, -1, "failed to create RTP audio session, drop p");
- return NULL;
- }
- ast_rtp_setqos(p->rtp, global_tos_audio, global_cos_audio, "SIP RTP");
- ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
- ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
- ast_rtp_set_rtptimeout(p->rtp, global_rtptimeout);
- ast_rtp_set_rtpholdtimeout(p->rtp, global_rtpholdtimeout);
- ast_rtp_set_rtpkeepalive(p->rtp, global_rtpkeepalive);
- if (p->vrtp) {
- ast_rtp_setqos(p->vrtp, global_tos_video, global_cos_video, "SIP VRTP");
- ast_rtp_setdtmf(p->vrtp, 0);
- ast_rtp_setdtmfcompensate(p->vrtp, 0);
- ast_rtp_set_rtptimeout(p->vrtp, global_rtptimeout);
- ast_rtp_set_rtpholdtimeout(p->vrtp, global_rtpholdtimeout);
- ast_rtp_set_rtpkeepalive(p->vrtp, global_rtpkeepalive);
- }
- if (p->trtp) {
- ast_rtp_setqos(p->trtp, global_tos_text, global_cos_text, "SIP TRTP");
- ast_rtp_setdtmf(p->trtp, 0);
- ast_rtp_setdtmfcompensate(p->trtp, 0);
- }
- if (p->udptl)
+ if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) && (p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, bindaddr.sin_addr))) {
ast_udptl_setqos(p->udptl, global_tos_audio, global_cos_audio);
+ }
p->maxcallbitrate = default_maxcallbitrate;
p->autoframing = global_autoframing;
- ast_rtp_codec_setpref(p->rtp, &p->prefs);
}
if (useglobal_nat && sin) {
@@ -6769,6 +6781,7 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
}
ast_string_field_set(p, context, sip_cfg.default_context);
ast_string_field_set(p, parkinglot, default_parkinglot);
+ ast_string_field_set(p, engine, default_engine);
AST_LIST_HEAD_INIT_NOLOCK(&p->request_queue);
@@ -7403,7 +7416,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
int iterator;
int sendonly = -1;
int numberofports;
- struct ast_rtp *newaudiortp, *newvideortp, *newtextrtp; /* Buffers for codec handling */
+ struct ast_rtp_codecs newaudiortp, newvideortp, newtextrtp;
int newjointcapability; /* Negotiated capability */
int newpeercapability;
int newnoncodeccapability;
@@ -7428,33 +7441,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
return -1;
}
- /* Initialize the temporary RTP structures we use to evaluate the offer from the peer */
-#ifdef LOW_MEMORY
- newaudiortp = ast_threadstorage_get(&ts_audio_rtp, ast_rtp_alloc_size());
-#else
- newaudiortp = alloca(ast_rtp_alloc_size());
-#endif
- memset(newaudiortp, 0, ast_rtp_alloc_size());
- ast_rtp_new_init(newaudiortp);
- ast_rtp_pt_clear(newaudiortp);
-
-#ifdef LOW_MEMORY
- newvideortp = ast_threadstorage_get(&ts_video_rtp, ast_rtp_alloc_size());
-#else
- newvideortp = alloca(ast_rtp_alloc_size());
-#endif
- memset(newvideortp, 0, ast_rtp_alloc_size());
- ast_rtp_new_init(newvideortp);
- ast_rtp_pt_clear(newvideortp);
-
-#ifdef LOW_MEMORY
- newtextrtp = ast_threadstorage_get(&ts_text_rtp, ast_rtp_alloc_size());
-#else
- newtextrtp = alloca(ast_rtp_alloc_size());
-#endif
- memset(newtextrtp, 0, ast_rtp_alloc_size());
- ast_rtp_new_init(newtextrtp);
- ast_rtp_pt_clear(newtextrtp);
+ /* Make sure that the codec structures are all cleared out */
+ ast_rtp_codecs_payloads_clear(&newaudiortp, NULL);
+ ast_rtp_codecs_payloads_clear(&newvideortp, NULL);
+ ast_rtp_codecs_payloads_clear(&newtextrtp, NULL);
/* Update our last rtprx when we receive an SDP, too */
p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
@@ -7536,11 +7526,13 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
p->novideo = TRUE;
p->notext = TRUE;
- if (p->vrtp)
- ast_rtp_pt_clear(newvideortp); /* Must be cleared in case no m=video line exists */
-
- if (p->trtp)
- ast_rtp_pt_clear(newtextrtp); /* Must be cleared in case no m=text line exists */
+ if (p->vrtp) {
+ ast_rtp_codecs_payloads_clear(&newvideortp, NULL);
+ }
+
+ if (p->trtp) {
+ ast_rtp_codecs_payloads_clear(&newtextrtp, NULL);
+ }
/* Find media streams in this SDP offer */
while ((m = get_sdp_iterate(&iterator, req, "m"))[0] != '\0') {
@@ -7565,7 +7557,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
}
if (debug)
ast_verbose("Found RTP audio format %d\n", codec);
- ast_rtp_set_m_type(newaudiortp, codec);
+
+ ast_rtp_codecs_payloads_set_m_type(&newaudiortp, NULL, codec);
}
} else if ((sscanf(m, "video %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
(sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1 && len >= 0)) {
@@ -7581,7 +7574,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
}
if (debug)
ast_verbose("Found RTP video format %d\n", codec);
- ast_rtp_set_m_type(newvideortp, codec);
+ ast_rtp_codecs_payloads_set_m_type(&newvideortp, NULL, codec);
}
} else if ((sscanf(m, "text %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
(sscanf(m, "text %d RTP/AVP %n", &x, &len) == 1 && len > 0)) {
@@ -7597,7 +7590,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
}
if (debug)
ast_verbose("Found RTP text format %d\n", codec);
- ast_rtp_set_m_type(newtextrtp, codec);
+ ast_rtp_codecs_payloads_set_m_type(&newtextrtp, NULL, codec);
}
} else if (p->udptl && ( (sscanf(m, "image %d udptl t38%n", &x, &len) == 1 && len > 0) ||
(sscanf(m, "image %d UDPTL t38%n", &x, &len) == 1 && len > 0) )) {
@@ -7662,10 +7655,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
if (udptlportno > 0) {
sin.sin_port = htons(udptlportno);
if (ast_test_flag(&p->flags[0], SIP_NAT) && ast_test_flag(&p->flags[1], SIP_PAGE2_UDPTL_DESTINATION)) {
- struct sockaddr_in peer;
- ast_rtp_get_peer(p->rtp, &peer);
- if (peer.sin_addr.s_addr) {
- memcpy(&sin.sin_addr, &peer.sin_addr, sizeof(sin.sin_addr));
+ struct sockaddr_in remote_address;
+ ast_rtp_instance_get_remote_address(p->rtp, &remote_address);
+ if (remote_address.sin_addr.s_addr) {
+ memcpy(&sin, &remote_address, sizeof(sin));
if (debug) {
ast_log(LOG_DEBUG, "Peer T.38 UDPTL is set behind NAT and with destination, destination address now %s\n", ast_inet_ntoa(sin.sin_addr));
}
@@ -7685,7 +7678,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
if (p->rtp) {
if (portno > 0) {
sin.sin_port = htons(portno);
- ast_rtp_set_peer(p->rtp, &sin);
+ ast_rtp_instance_set_remote_address(p->rtp, &sin);
if (debug)
ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
} else {
@@ -7693,7 +7686,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
if (debug)
ast_verbose("Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid %s\n", p->callid);
} else {
- ast_rtp_stop(p->rtp);
+ ast_rtp_instance_stop(p->rtp);
if (debug)
ast_verbose("Peer doesn't provide audio. Callid %s\n", p->callid);
}
@@ -7776,18 +7769,17 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
}
}
if (framing && p->autoframing) {
- struct ast_codec_pref *pref = ast_rtp_codec_getpref(p->rtp);
+ struct ast_codec_pref *pref = &ast_rtp_instance_get_codecs(p->rtp)->pref;
int codec_n;
- int format = 0;
- for (codec_n = 0; codec_n < MAX_RTP_PT; codec_n++) {
- format = ast_rtp_codec_getformat(codec_n);
- if (!format) /* non-codec or not found */
+ for (codec_n = 0; codec_n < AST_RTP_MAX_PT; codec_n++) {
+ struct ast_rtp_payload_type format = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(p->rtp), codec_n);
+ if (!format.asterisk_format || !format.code) /* non-codec or not found */
continue;
if (option_debug)
- ast_log(LOG_DEBUG, "Setting framing for %d to %ld\n", format, framing);
- ast_codec_pref_setsize(pref, format, framing);
+ ast_log(LOG_DEBUG, "Setting framing for %d to %ld\n", format.code, framing);
+ ast_codec_pref_setsize(pref, format.code, framing);
}
- ast_rtp_codec_setpref(p->rtp, pref);
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, pref);
}
continue;
}
@@ -7799,7 +7791,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
sscanf(red_cp, "%u", &red_data_pt[red_num_gen]);
red_cp = strtok(red_cp, "/");
- while (red_cp && red_num_gen++ < RED_MAX_GENERATION) {
+ while (red_cp && red_num_gen++ < AST_RED_MAX_GENERATION) {
sscanf(red_cp, "%u", &red_data_pt[red_num_gen]);
red_cp = strtok(NULL, "/");
}
@@ -7808,15 +7800,15 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
}
if (sscanf(a, "fmtp: %u %63s", &codec, fmtp_string) == 2) {
- struct rtpPayloadType payload;
+ struct ast_rtp_payload_type payload;
unsigned int handled = 0;
- payload = ast_rtp_lookup_pt(newaudiortp, codec);
+ payload = ast_rtp_codecs_payload_lookup(&newaudiortp, codec);
if (!payload.code) {
/* it wasn't found, try the video rtp */
- payload = ast_rtp_lookup_pt(newvideortp, codec);
+ payload = ast_rtp_codecs_payload_lookup(&newvideortp, codec);
}
- if (payload.code && payload.isAstFormat) {
+ if (payload.code && payload.asterisk_format) {
unsigned int bit_rate;
switch (payload.code) {
@@ -7824,7 +7816,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
if (sscanf(fmtp_string, "bitrate=%u", &bit_rate) == 1) {
if (bit_rate != 32000) {
ast_log(LOG_WARNING, "Got Siren7 offer at %d bps, but only 32000 bps supported; ignoring.\n", bit_rate);
- ast_rtp_unset_m_type(newaudiortp, codec);
+ ast_rtp_codecs_payloads_unset(&newaudiortp, NULL, codec);
} else {
handled = 1;
}
@@ -7834,7 +7826,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
if (sscanf(fmtp_string, "bitrate=%u", &bit_rate) == 1) {
if (bit_rate != 48000) {
ast_log(LOG_WARNING, "Got Siren14 offer at %d bps, but only 48000 bps supported; ignoring.\n", bit_rate);
- ast_rtp_unset_m_type(newaudiortp, codec);
+ ast_rtp_codecs_payloads_unset(&newaudiortp, NULL, codec);
} else {
handled = 1;
}
@@ -7856,24 +7848,24 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
/* Note: should really look at the '#chans' params too */
/* Note: This should all be done in the context of the m= above */
if (!strncasecmp(mimeSubtype, "H26", 3) || !strncasecmp(mimeSubtype, "MP4", 3)) { /* Video */
- if (ast_rtp_set_rtpmap_type_rate(newvideortp, codec, "video", mimeSubtype, 0, sample_rate) != -1) {
+ if (ast_rtp_codecs_payloads_set_rtpmap_type_rate(&newvideortp, NULL, codec, "video", mimeSubtype, 0, sample_rate) != -1) {
if (debug)
ast_verbose("Found video description format %s for ID %d\n", mimeSubtype, codec);
found_rtpmap_codecs[last_rtpmap_codec] = codec;
last_rtpmap_codec++;
} else {
- ast_rtp_unset_m_type(newvideortp, codec);
+ ast_rtp_codecs_payloads_unset(&newvideortp, NULL, codec);
if (debug)
ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
}
} else if (!strncasecmp(mimeSubtype, "T140", 4)) { /* Text */
if (p->trtp) {
/* ast_verbose("Adding t140 mimeSubtype to textrtp struct\n"); */
- ast_rtp_set_rtpmap_type(newtextrtp, codec, "text", mimeSubtype, 0);
+ ast_rtp_codecs_payloads_set_rtpmap_type_rate(&newtextrtp, NULL, codec, "text", mimeSubtype, 0, sample_rate);
}
} else if (!strncasecmp(mimeSubtype, "RED", 3)) { /* Text with Redudancy */
if (p->trtp) {
- ast_rtp_set_rtpmap_type(newtextrtp, codec, "text", mimeSubtype, 0);
+ ast_rtp_codecs_payloads_set_rtpmap_type_rate(&newtextrtp, NULL, codec, "text", mimeSubtype, 0, sample_rate);
red_pt = codec;
sprintf(red_fmtp, "fmtp:%d ", red_pt);
@@ -7881,15 +7873,14 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_verbose("RED submimetype has payload type: %d\n", red_pt);
}
} else { /* Must be audio?? */
- if (ast_rtp_set_rtpmap_type_rate(newaudiortp, codec, "audio", mimeSubtype,
- ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0,
- sample_rate) != -1) {
+ if (ast_rtp_codecs_payloads_set_rtpmap_type_rate(&newaudiortp, NULL, codec, "audio", mimeSubtype,
+ ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0, sample_rate) != -1) {
if (debug)
ast_verbose("Found audio description format %s for ID %d\n", mimeSubtype, codec);
found_rtpmap_codecs[last_rtpmap_codec] = codec;
last_rtpmap_codec++;
} else {
- ast_rtp_unset_m_type(newaudiortp, codec);
+ ast_rtp_codecs_payloads_unset(&newaudiortp, NULL, codec);
if (debug)
ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
}
@@ -8028,15 +8019,14 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
}
/* Now gather all of the codecs that we are asked for: */
- ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
- ast_rtp_get_current_formats(newvideortp, &vpeercapability, &vpeernoncodeccapability);
- ast_rtp_get_current_formats(newtextrtp, &tpeercapability, &tpeernoncodeccapability);
+ ast_rtp_codecs_payload_formats(&newaudiortp, &peercapability, &peernoncodeccapability);
+ ast_rtp_codecs_payload_formats(&newvideortp, &vpeercapability, &vpeernoncodeccapability);
+ ast_rtp_codecs_payload_formats(&newtextrtp, &tpeercapability, &tpeernoncodeccapability);
newjointcapability = p->capability & (peercapability | vpeercapability | tpeercapability);
newpeercapability = (peercapability | vpeercapability | tpeercapability);
newnoncodeccapability = p->noncodeccapability & peernoncodeccapability;
-
-
+
if (debug) {
/* shame on whoever coded this.... */
char s1[SIPBUFSIZE], s2[SIPBUFSIZE], s3[SIPBUFSIZE], s4[SIPBUFSIZE], s5[SIPBUFSIZE];
@@ -8047,11 +8037,17 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_getformatname_multiple(s3, SIPBUFSIZE, vpeercapability),
ast_getformatname_multiple(s4, SIPBUFSIZE, tpeercapability),
ast_getformatname_multiple(s5, SIPBUFSIZE, newjointcapability));
+ }
+
+ if (debug) {
+ struct ast_str *s1 = ast_str_alloca(SIPBUFSIZE);
+ struct ast_str *s2 = ast_str_alloca(SIPBUFSIZE);
+ struct ast_str *s3 = ast_str_alloca(SIPBUFSIZE);
ast_verbose("Non-codec capabilities (dtmf): us - %s, peer - %s, combined - %s\n",
- ast_rtp_lookup_mime_multiple(s1, SIPBUFSIZE, p->noncodeccapability, 0, 0),
- ast_rtp_lookup_mime_multiple(s2, SIPBUFSIZE, peernoncodeccapability, 0, 0),
- ast_rtp_lookup_mime_multiple(s3, SIPBUFSIZE, newnoncodeccapability, 0, 0));
+ ast_rtp_lookup_mime_multiple2(s1, p->noncodeccapability, 0, 0),
+ ast_rtp_lookup_mime_multiple2(s2, peernoncodeccapability, 0, 0),
+ ast_rtp_lookup_mime_multiple2(s3, newnoncodeccapability, 0, 0));
}
if (!newjointcapability) {
/* If T.38 was not negotiated either, totally bail out... */
@@ -8082,11 +8078,13 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
p->red = 0;
}
- ast_rtp_pt_copy(p->rtp, newaudiortp);
- if (p->vrtp)
- ast_rtp_pt_copy(p->vrtp, newvideortp);
- if (p->trtp)
- ast_rtp_pt_copy(p->trtp, newtextrtp);
+ ast_rtp_codecs_payloads_copy(&newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp);
+ if (p->vrtp) {
+ ast_rtp_codecs_payloads_copy(&newvideortp, ast_rtp_instance_get_codecs(p->vrtp), p->vrtp);
+ }
+ if (p->trtp) {
+ ast_rtp_codecs_payloads_copy(&newtextrtp, ast_rtp_instance_get_codecs(p->trtp), p->trtp);
+ }
if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
ast_clear_flag(&p->flags[0], SIP_DTMF);
@@ -8094,8 +8092,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
/* XXX Would it be reasonable to drop the DSP at this point? XXX */
ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833);
/* Since RFC2833 is now negotiated we need to change some properties of the RTP stream */
- ast_rtp_setdtmf(p->rtp, 1);
- ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF, 1);
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
} else {
ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
}
@@ -8103,21 +8101,21 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
/* Setup audio port number */
if (p->rtp && sin.sin_port) {
- ast_rtp_set_peer(p->rtp, &sin);
+ ast_rtp_instance_set_remote_address(p->rtp, &sin);
if (debug)
ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
}
/* Setup video port number */
if (p->vrtp && vsin.sin_port) {
- ast_rtp_set_peer(p->vrtp, &vsin);
+ ast_rtp_instance_set_remote_address(p->vrtp, &vsin);
if (debug)
ast_verbose("Peer video RTP is at port %s:%d\n", ast_inet_ntoa(vsin.sin_addr), ntohs(vsin.sin_port));
}
/* Setup text port number */
if (p->trtp && tsin.sin_port) {
- ast_rtp_set_peer(p->trtp, &tsin);
+ ast_rtp_instance_set_remote_address(p->trtp, &tsin);
if (debug)
ast_verbose("Peer text RTP is at port %s:%d\n", ast_inet_ntoa(tsin.sin_addr), ntohs(tsin.sin_port));
}
@@ -8164,7 +8162,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
S_OR(p->mohsuggest, NULL),
!ast_strlen_zero(p->mohsuggest) ? strlen(p->mohsuggest) + 1 : 0);
if (sendonly)
- ast_rtp_stop(p->rtp);
+ ast_rtp_instance_stop(p->rtp);
/* RTCP needs to go ahead, even if we're on hold!!! */
/* Activate a re-invite */
ast_queue_frame(p->owner, &ast_null_frame);
@@ -9001,19 +8999,19 @@ static void add_codec_to_sdp(const struct sip_pvt *p, int codec,
if (debug)
ast_verbose("Adding codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
- if ((rtp_code = ast_rtp_lookup_code(p->rtp, 1, codec)) == -1)
+ if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 1, codec)) == -1)
return;
if (p->rtp) {
- struct ast_codec_pref *pref = ast_rtp_codec_getpref(p->rtp);
+ struct ast_codec_pref *pref = &ast_rtp_instance_get_codecs(p->rtp)->pref;
fmt = ast_codec_pref_getsize(pref, codec);
} else /* I dont see how you couldn't have p->rtp, but good to check for and error out if not there like earlier code */
return;
ast_str_append(m_buf, 0, " %d", rtp_code);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
- ast_rtp_lookup_mime_subtype(1, codec,
+ ast_rtp_lookup_mime_subtype2(1, codec,
ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0),
- ast_rtp_lookup_sample_rate(1, codec));
+ ast_rtp_lookup_sample_rate2(1, codec));
switch (codec) {
case AST_FORMAT_G729A:
@@ -9060,13 +9058,13 @@ static void add_vcodec_to_sdp(const struct sip_pvt *p, int codec,
if (debug)
ast_verbose("Adding video codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
- if ((rtp_code = ast_rtp_lookup_code(p->vrtp, 1, codec)) == -1)
+ if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->vrtp), 1, codec)) == -1)
return;
ast_str_append(m_buf, 0, " %d", rtp_code);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
- ast_rtp_lookup_mime_subtype(1, codec, 0),
- ast_rtp_lookup_sample_rate(1, codec));
+ ast_rtp_lookup_mime_subtype2(1, codec, 0),
+ ast_rtp_lookup_sample_rate2(1, codec));
/* Add fmtp code here */
}
@@ -9083,20 +9081,21 @@ static void add_tcodec_to_sdp(const struct sip_pvt *p, int codec,
if (debug)
ast_verbose("Adding text codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
- if ((rtp_code = ast_rtp_lookup_code(p->trtp, 1, codec)) == -1)
+ if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, codec)) == -1)
return;
ast_str_append(m_buf, 0, " %d", rtp_code);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
- ast_rtp_lookup_mime_subtype(1, codec, 0),
- ast_rtp_lookup_sample_rate(1, codec));
+ ast_rtp_lookup_mime_subtype2(1, codec, 0),
+ ast_rtp_lookup_sample_rate2(1, codec));
/* Add fmtp code here */
if (codec == AST_FORMAT_T140RED) {
- ast_str_append(a_buf, 0, "a=fmtp:%d %d/%d/%d\r\n", rtp_code,
- ast_rtp_lookup_code(p->trtp, 1, AST_FORMAT_T140),
- ast_rtp_lookup_code(p->trtp, 1, AST_FORMAT_T140),
- ast_rtp_lookup_code(p->trtp, 1, AST_FORMAT_T140));
+ int t140code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, AST_FORMAT_T140);
+ ast_str_append(a_buf, 0, "a=fmtp:%d %d/%d/%d\r\n", rtp_code,
+ t140code,
+ t140code,
+ t140code);
}
}
@@ -9139,14 +9138,14 @@ static void add_noncodec_to_sdp(const struct sip_pvt *p, int format,
int rtp_code;
if (debug)
- ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", format, ast_rtp_lookup_mime_subtype(0, format, 0));
- if ((rtp_code = ast_rtp_lookup_code(p->rtp, 0, format)) == -1)
+ ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", format, ast_rtp_lookup_mime_subtype2(0, format, 0));
+ if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 0, format)) == -1)
return;
ast_str_append(m_buf, 0, " %d", rtp_code);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
- ast_rtp_lookup_mime_subtype(0, format, 0),
- ast_rtp_lookup_sample_rate(0, format));
+ ast_rtp_lookup_mime_subtype2(0, format, 0),
+ ast_rtp_lookup_sample_rate2(0, format));
if (format == AST_RTP_DTMF) /* Indicate we support DTMF and FLASH... */
ast_str_append(a_buf, 0, "a=fmtp:%d 0-16\r\n", rtp_code);
}
@@ -9159,11 +9158,11 @@ static void get_our_media_address(struct sip_pvt *p, int needvideo,
struct sockaddr_in *dest, struct sockaddr_in *vdest)
{
/* First, get our address */
- ast_rtp_get_us(p->rtp, sin);
+ ast_rtp_instance_get_local_address(p->rtp, sin);
if (p->vrtp)
- ast_rtp_get_us(p->vrtp, vsin);
+ ast_rtp_instance_get_local_address(p->vrtp, vsin);
if (p->trtp)
- ast_rtp_get_us(p->trtp, tsin);
+ ast_rtp_instance_get_local_address(p->trtp, tsin);
/* Now, try to figure out where we want them to send data */
/* Is this a re-invite to move the media out, then use the original offer from caller */
@@ -9594,7 +9593,7 @@ static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const
if (p->rtp) {
if (!p->autoframing && !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
ast_debug(1, "Setting framing from config on incoming call\n");
- ast_rtp_codec_setpref(p->rtp, &p->prefs);
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &p->prefs);
}
try_suggested_sip_codec(p);
if (p->t38.state == T38_PEER_DIRECT || p->t38.state == T38_ENABLED) {
@@ -12087,12 +12086,6 @@ static enum check_auth_result register_verify(struct sip_pvt *p, struct sockaddr
}
if (peer) {
- /*! \todo OEJ Remove this - there's never RTP in a REGISTER dialog... */
- /* Set Frame packetization */
- if (p->rtp) {
- ast_rtp_codec_setpref(p->rtp, &peer->prefs);
- p->autoframing = peer->autoframing;
- }
if (!peer->host_dynamic) {
ast_log(LOG_ERROR, "Peer '%s' is trying to register, but not configured as host=dynamic\n", peer->name);
res = AUTH_PEER_NOT_DYNAMIC;
@@ -13024,7 +13017,7 @@ static enum check_auth_result check_peer_ok(struct sip_pvt *p, char *of,
/* XXX what about p->prefs = peer->prefs; ? */
/* Set Frame packetization */
if (p->rtp) {
- ast_rtp_codec_setpref(p->rtp, &peer->prefs);
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &peer->prefs);
p->autoframing = peer->autoframing;
}
@@ -13046,6 +13039,7 @@ static enum check_auth_result check_peer_ok(struct sip_pvt *p, char *of,
ast_string_field_set(p, mohinterpret, peer->mohinterpret);
ast_string_field_set(p, mohsuggest, peer->mohsuggest);
ast_string_field_set(p, parkinglot, peer->parkinglot);
+ ast_string_field_set(p, engine, peer->engine);
if (peer->callingpres) /* Peer calling pres setting will override RPID */
p->callingpres = peer->callingpres;
if (peer->maxms && peer->lastms)
@@ -13113,17 +13107,6 @@ static enum check_auth_result check_peer_ok(struct sip_pvt *p, char *of,
if (p->peercapability)
p->jointcapability &= p->peercapability;
p->maxcallbitrate = peer->maxcallbitrate;
- if (!ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS) &&
- (!ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) ||
- !(p->capability & AST_FORMAT_VIDEO_MASK)) &&
- p->vrtp) {
- ast_rtp_destroy(p->vrtp);
- p->vrtp = NULL;
- }
- if ((!ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT) || !(p->capability & AST_FORMAT_TEXT_MASK)) && p->trtp) {
- ast_rtp_destroy(p->trtp);
- p->trtp = NULL;
- }
if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
(ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
p->noncodeccapability |= AST_RTP_DTMF;
@@ -13132,6 +13115,12 @@ static enum check_auth_result check_peer_ok(struct sip_pvt *p, char *of,
p->jointnoncodeccapability = p->noncodeccapability;
if (p->t38.peercapability)
p->t38.jointcapability &= p->t38.peercapability;
+ if (!dialog_initialize_rtp(p)) {
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &peer->prefs);
+ p->autoframing = peer->autoframing;
+ } else {
+ res = AUTH_RTP_FAILED;
+ }
}
unref_peer(peer, "check_peer_ok: unref_peer: tossing temp ptr to peer from find_peer");
return res;
@@ -13253,7 +13242,11 @@ static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_requ
/* Finally, apply the guest policy */
if (sip_cfg.allowguest) {
replace_cid(p, rpid_num, calleridname);
- res = AUTH_SUCCESSFUL;
+ if (!dialog_initialize_rtp(p)) {
+ res = AUTH_SUCCESSFUL;
+ } else {
+ res = AUTH_RTP_FAILED;
+ }
} else if (sip_cfg.alwaysauthreject)
res = AUTH_FAKE_AUTH; /* reject with fake authorization request */
else
@@ -14050,7 +14043,20 @@ static int dialog_needdestroy(void *dialogobj, void *arg, int flags)
*/
return 0;
}
-
+
+ /* We absolutely cannot destroy the rtp struct while a bridge is active or we WILL crash */
+ if (dialog->rtp && ast_rtp_instance_get_bridged(dialog->rtp)) {
+ ast_debug(2, "Bridge still active. Delaying destroy of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
+ sip_pvt_unlock(dialog);
+ return 0;
+ }
+
+ if (dialog->vrtp && ast_rtp_instance_get_bridged(dialog->vrtp)) {
+ ast_debug(2, "Bridge still active. Delaying destroy of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
+ sip_pvt_unlock(dialog);
+ return 0;
+ }
+
/* Check RTP timeouts and kill calls if we have a timeout set and do not get RTP */
check_rtp_timeout(dialog, *t);
@@ -14059,13 +14065,13 @@ static int dialog_needdestroy(void *dialogobj, void *arg, int flags)
- if that's the case, wait with destruction */
if (dialog->needdestroy && !dialog->packets && !dialog->owner) {
/* We absolutely cannot destroy the rtp struct while a bridge is active or we WILL crash */
- if (dialog->rtp && ast_rtp_get_bridged(dialog->rtp)) {
+ if (dialog->rtp && ast_rtp_instance_get_bridged(dialog->rtp)) {
ast_debug(2, "Bridge still active. Delaying destruction of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
sip_pvt_unlock(dialog);
return 0;
}
- if (dialog->vrtp && ast_rtp_get_bridged(dialog->vrtp)) {
+ if (dialog->vrtp && ast_rtp_instance_get_bridged(dialog->vrtp)) {
ast_debug(2, "Bridge still active. Delaying destroy of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
sip_pvt_unlock(dialog);
return 0;
@@ -14555,6 +14561,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
ast_cli(fd, " Sess-Refresh : %s\n", strefresher2str(peer->stimer.st_ref));
ast_cli(fd, " Sess-Expires : %d secs\n", peer->stimer.st_max_se);
ast_cli(fd, " Min-Sess : %d secs\n", peer->stimer.st_min_se);
+ ast_cli(fd, " RTP Engine : %s\n", peer->engine);
ast_cli(fd, "\n");
peer = unref_peer(peer, "sip_show_peer: unref_peer: done with peer ptr");
} else if (peer && type == 1) { /* manager listing */
@@ -14602,6 +14609,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
astman_append(s, "SIP-Sess-Refresh: %s\r\n", strefresher2str(peer->stimer.st_ref));
astman_append(s, "SIP-Sess-Expires: %d\r\n", peer->stimer.st_max_se);
astman_append(s, "SIP-Sess-Min: %d\r\n", peer->stimer.st_min_se);
+ astman_append(s, "SIP-RTP-Engine: %s\r\n", peer->engine);
/* - is enumerated */
astman_append(s, "SIP-DTMFmode: %s\r\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
@@ -14734,6 +14742,7 @@ static char *sip_show_user(struct ast_cli_entry *e, int cmd, struct ast_cli_args
ast_cli(a->fd, " Sess-Refresh : %s\n", strefresher2str(user->stimer.st_ref));
ast_cli(a->fd, " Sess-Expires : %d secs\n", user->stimer.st_max_se);
ast_cli(a->fd, " Sess-Min-SE : %d secs\n", user->stimer.st_min_se);
+ ast_cli(a->fd, " RTP Engine : %s\n", user->engine);
ast_cli(a->fd, " Codec Order : (");
print_codec_to_cli(a->fd, &user->prefs);
@@ -14888,11 +14897,10 @@ static int show_chanstats_cb(void *__cur, void *__arg, int flags)
#define FORMAT2 "%-15.15s %-11.11s %-8.8s %-10.10s %-10.10s (%-2.2s) %-6.6s %-10.10s %-10.10s ( %%) %-6.6s\n"
#define FORMAT "%-15.15s %-11.11s %-8.8s %-10.10u%-1.1s %-10.10u (%-2.2u%%) %-6.6u %-10.10u%-1.1s %-10.10u (%-2.2u%%) %-6.6u\n"
struct sip_pvt *cur = __cur;
- unsigned int rxcount;
- unsigned int txcount;
+ struct ast_rtp_instance_stats stats;
char durbuf[10];
- int duration;
- int durh, durm, durs;
+ int duration;
+ int durh, durm, durs;
struct ast_channel *c = cur->owner;
struct __show_chan_arg *arg = __arg;
int fd = arg->fd;
@@ -14906,10 +14914,9 @@ static int show_chanstats_cb(void *__cur, void *__arg, int flags)
ast_cli(fd, "%-15.15s %-11.11s (inv state: %s) -- %s\n", ast_inet_ntoa(cur->sa.sin_addr), cur->callid, invitestate2string[cur->invitestate].desc, "-- No RTP active");
return 0; /* don't care, we scan all channels */
}
- rxcount = ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXCOUNT);
- txcount = ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXCOUNT);
- /* Find the duration of this channel */
+ ast_rtp_instance_get_stats(cur->rtp, &stats, AST_RTP_INSTANCE_STAT_ALL);
+
if (c && c->cdr && !ast_tvzero(c->cdr->start)) {
duration = (int)(ast_tvdiff_ms(ast_tvnow(), c->cdr->start) / 1000);
durh = duration / 3600;
@@ -14919,21 +14926,21 @@ static int show_chanstats_cb(void *__cur, void *__arg, int flags)
} else {
durbuf[0] = '\0';
}
- /* Print stats for every call with RTP */
+
ast_cli(fd, FORMAT,
ast_inet_ntoa(cur->sa.sin_addr),
cur->callid,
durbuf,
- rxcount > (unsigned int) 100000 ? (unsigned int) (rxcount)/(unsigned int) 1000 : rxcount,
- rxcount > (unsigned int) 100000 ? "K":" ",
- ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXPLOSS),
- rxcount > ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXPLOSS) ? (unsigned int) (ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXPLOSS) / rxcount * 100) : 0,
- ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXJITTER),
- txcount > (unsigned int) 100000 ? (unsigned int) (txcount)/(unsigned int) 1000 : txcount,
- txcount > (unsigned int) 100000 ? "K":" ",
- ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXPLOSS),
- txcount > ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXPLOSS) ? (unsigned int) (ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXPLOSS)/ txcount * 100) : 0,
- ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXJITTER)
+ stats.rxcount > (unsigned int) 100000 ? (unsigned int) (stats.rxcount)/(unsigned int) 1000 : stats.rxcount,
+ stats.rxcount > (unsigned int) 100000 ? "K":" ",
+ stats.rxploss,
+ stats.rxcount > stats.rxploss ? (stats.rxploss / stats.rxcount * 100) : 0,
+ stats.rxjitter,
+ stats.txcount > (unsigned int) 100000 ? (unsigned int) (stats.txcount)/(unsigned int) 1000 : stats.txcount,
+ stats.txcount > (unsigned int) 100000 ? "K":" ",
+ stats.txploss,
+ stats.txcount > stats.txploss ? (stats.txploss / stats.txcount * 100) : 0,
+ stats.txjitter
);
arg->numchans++;
@@ -16880,7 +16887,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
if (p->udptl && p->t38.state == T38_LOCAL_REINVITE) {
change_t38_state(p, T38_DISABLED);
/* Try to reset RTP timers */
- ast_rtp_set_rtptimers_onhold(p->rtp);
+ //ast_rtp_set_rtptimers_onhold(p->rtp);
/* Trigger a reinvite back to audio */
transmit_reinvite_with_sdp(p, FALSE, FALSE);
@@ -17300,11 +17307,11 @@ static void stop_media_flows(struct sip_pvt *p)
{
/* Immediately stop RTP, VRTP and UDPTL as applicable */
if (p->rtp)
- ast_rtp_stop(p->rtp);
+ ast_rtp_instance_stop(p->rtp);
if (p->vrtp)
- ast_rtp_stop(p->vrtp);
+ ast_rtp_instance_stop(p->vrtp);
if (p->trtp)
- ast_rtp_stop(p->trtp);
+ ast_rtp_instance_stop(p->trtp);
if (p->udptl)
ast_udptl_stop(p->udptl);
}
@@ -19032,8 +19039,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
build_contact(p); /* Build our contact header */
if (p->rtp) {
- ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
- ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
}
if (!replace_id && gotdest) { /* No matching extension found */
@@ -19852,7 +19859,7 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
static int acf_channel_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen)
{
struct sip_pvt *p = chan->tech_pvt;
- char *all = "", *parse = ast_strdupa(preparse);
+ char *parse = ast_strdupa(preparse);
int res = 0;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(param);
@@ -19890,61 +19897,70 @@ static int acf_channel_read(struct ast_channel *chan, const char *funcname, char
args.type = "audio";
if (!strcasecmp(args.type, "audio"))
- ast_rtp_get_peer(p->rtp, &sin);
+ ast_rtp_instance_get_remote_address(p->rtp, &sin);
else if (!strcasecmp(args.type, "video"))
- ast_rtp_get_peer(p->vrtp, &sin);
+ ast_rtp_instance_get_remote_address(p->vrtp, &sin);
else if (!strcasecmp(args.type, "text"))
- ast_rtp_get_peer(p->trtp, &sin);
+ ast_rtp_instance_get_remote_address(p->trtp, &sin);
else
return -1;
snprintf(buf, buflen, "%s:%d", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
} else if (!strcasecmp(args.param, "rtpqos")) {
- struct ast_rtp_quality qos;
- struct ast_rtp *rtp = p->rtp;
-
- memset(&qos, 0, sizeof(qos));
+ struct ast_rtp_instance *rtp = NULL;
- if (ast_strlen_zero(args.type))
+ if (ast_strlen_zero(args.type)) {
args.type = "audio";
- if (ast_strlen_zero(args.field))
- args.field = "all";
-
- if (!strcasecmp(args.type, "AUDIO")) {
- all = ast_rtp_get_quality(rtp = p->rtp, &qos, RTPQOS_SUMMARY);
- } else if (!strcasecmp(args.type, "VIDEO")) {
- all = ast_rtp_get_quality(rtp = p->vrtp, &qos, RTPQOS_SUMMARY);
- } else if (!strcasecmp(args.type, "TEXT")) {
- all = ast_rtp_get_quality(rtp = p->trtp, &qos, RTPQOS_SUMMARY);
+ }
+
+ if (!strcasecmp(args.type, "audio")) {
+ rtp = p->rtp;
+ } else if (!strcasecmp(args.type, "video")) {
+ rtp = p->vrtp;
+ } else if (!strcasecmp(args.type, "text")) {
+ rtp = p->trtp;
} else {
- return -1;
+ return -1;
}
-
- if (!strcasecmp(args.field, "local_ssrc"))
- snprintf(buf, buflen, "%u", qos.local_ssrc);
- else if (!strcasecmp(args.field, "local_lostpackets"))
- snprintf(buf, buflen, "%u", qos.local_lostpackets);
- else if (!strcasecmp(args.field, "local_jitter"))
- snprintf(buf, buflen, "%.0f", qos.local_jitter * 1000.0);
- else if (!strcasecmp(args.field, "local_count"))
- snprintf(buf, buflen, "%u", qos.local_count);
- else if (!strcasecmp(args.field, "remote_ssrc"))
- snprintf(buf, buflen, "%u", qos.remote_ssrc);
- else if (!strcasecmp(args.field, "remote_lostpackets"))
- snprintf(buf, buflen, "%u", qos.remote_lostpackets);
- else if (!strcasecmp(args.field, "remote_jitter"))
- snprintf(buf, buflen, "%.0f", qos.remote_jitter * 1000.0);
- else if (!strcasecmp(args.field, "remote_count"))
- snprintf(buf, buflen, "%u", qos.remote_count);
- else if (!strcasecmp(args.field, "rtt"))
- snprintf(buf, buflen, "%.0f", qos.rtt * 1000.0);
- else if (!strcasecmp(args.field, "all"))
- ast_copy_string(buf, all, buflen);
- else if (!ast_rtp_get_qos(rtp, args.field, buf, buflen))
- ;
- else {
- ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname);
- return -1;
+
+ if (ast_strlen_zero(args.field) || !strcasecmp(args.field, "all")) {
+ char quality_buf[AST_MAX_USER_FIELD], *quality;
+
+ if (!(quality = ast_rtp_instance_get_quality(rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
+ return -1;
+ }
+
+ ast_copy_string(buf, quality_buf, buflen);
+ return res;
+ } else {
+ struct ast_rtp_instance_stats stats;
+
+ if (ast_rtp_instance_get_stats(rtp, &stats, AST_RTP_INSTANCE_STAT_ALL)) {
+ return -1;
+ }
+
+ if (!strcasecmp(args.field, "local_ssrc")) {
+ snprintf(buf, buflen, "%u", stats.local_ssrc);
+ } else if (!strcasecmp(args.field, "local_lostpackets")) {
+ snprintf(buf, buflen, "%u", stats.rxploss);
+ } else if (!strcasecmp(args.field, "local_jitter")) {
+ snprintf(buf, buflen, "%u", stats.rxjitter);
+ } else if (!strcasecmp(args.field, "local_count")) {
+ snprintf(buf, buflen, "%u", stats.rxcount);
+ } else if (!strcasecmp(args.field, "remote_ssrc")) {
+ snprintf(buf, buflen, "%u", stats.remote_ssrc);
+ } else if (!strcasecmp(args.field, "remote_lostpackets")) {
+ snprintf(buf, buflen, "%u", stats.txploss);
+ } else if (!strcasecmp(args.field, "remote_jitter")) {
+ snprintf(buf, buflen, "%u", stats.txjitter);
+ } else if (!strcasecmp(args.field, "remote_count")) {
+ snprintf(buf, buflen, "%u", stats.txcount);
+ } else if (!strcasecmp(args.field, "rtt")) {
+ snprintf(buf, buflen, "%u", stats.rtt);
+ } else {
+ ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname);
+ return -1;
+ }
}
} else {
res = -1;
@@ -19976,53 +19992,53 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
/* Get RTCP quality before end of call */
if (p->do_history || p->owner) {
+ char quality_buf[AST_MAX_USER_FIELD], *quality;
struct ast_channel *bridge = p->owner ? ast_bridged_channel(p->owner) : NULL;
- char *videoqos, *textqos;
- if (p->rtp) {
+ if (p->rtp && (quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
if (p->do_history) {
- char *audioqos,
- *audioqos_jitter,
- *audioqos_loss,
- *audioqos_rtt;
-
- audioqos = ast_rtp_get_quality(p->rtp, NULL, RTPQOS_SUMMARY);
- audioqos_jitter = ast_rtp_get_quality(p->rtp, NULL, RTPQOS_JITTER);
- audioqos_loss = ast_rtp_get_quality(p->rtp, NULL, RTPQOS_LOSS);
- audioqos_rtt = ast_rtp_get_quality(p->rtp, NULL, RTPQOS_RTT);
-
- append_history(p, "RTCPaudio", "Quality:%s", audioqos);
- append_history(p, "RTCPaudioJitter", "Quality:%s", audioqos_jitter);
- append_history(p, "RTCPaudioLoss", "Quality:%s", audioqos_loss);
- append_history(p, "RTCPaudioRTT", "Quality:%s", audioqos_rtt);
+ append_history(p, "RTCPaudio", "Quality:%s", quality);
+
+ if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf)))) {
+ append_history(p, "RTCPaudioJitter", "Quality:%s", quality);
+ }
+ if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf)))) {
+ append_history(p, "RTCPaudioLoss", "Quality:%s", quality);
+ }
+ if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf)))) {
+ append_history(p, "RTCPaudioRTT", "Quality:%s", quality);
+ }
}
-
+
if (p->owner) {
- ast_rtp_set_vars(p->owner, p->rtp);
+ ast_rtp_instance_set_stats_vars(p->owner, p->rtp);
}
+
}
if (bridge) {
struct sip_pvt *q = bridge->tech_pvt;
- if (IS_SIP_TECH(bridge->tech) && q->rtp)
- ast_rtp_set_vars(bridge, q->rtp);
+ if (IS_SIP_TECH(bridge->tech) && q->rtp) {
+ ast_rtp_instance_set_stats_vars(bridge, q->rtp);
+ }
}
- if (p->vrtp) {
- videoqos = ast_rtp_get_quality(p->vrtp, NULL, RTPQOS_SUMMARY);
- if (p->do_history)
- append_history(p, "RTCPvideo", "Quality:%s", videoqos);
- if (p->owner)
- pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos);
+ if (p->vrtp && (quality = ast_rtp_instance_get_quality(p->vrtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
+ if (p->do_history) {
+ append_history(p, "RTCPvideo", "Quality:%s", quality);
+ }
+ if (p->owner) {
+ pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", quality);
+ }
}
-
- if (p->trtp) {
- textqos = ast_rtp_get_quality(p->trtp, NULL, RTPQOS_SUMMARY);
- if (p->do_history)
- append_history(p, "RTCPtext", "Quality:%s", textqos);
- if (p->owner)
- pbx_builtin_setvar_helper(p->owner, "RTPTEXTQOS", textqos);
+ if (p->trtp && (quality = ast_rtp_instance_get_quality(p->trtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
+ if (p->do_history) {
+ append_history(p, "RTCPtext", "Quality:%s", quality);
+ }
+ if (p->owner) {
+ pbx_builtin_setvar_helper(p->owner, "RTPTEXTQOS", quality);
+ }
}
}
@@ -21211,15 +21227,8 @@ static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
return;
/* If we have no timers set, return now */
- if ((ast_rtp_get_rtpkeepalive(dialog->rtp) == 0) && (ast_rtp_get_rtptimeout(dialog->rtp) == 0) && (ast_rtp_get_rtpholdtimeout(dialog->rtp) == 0))
+ if (!ast_rtp_instance_get_timeout(dialog->rtp) && !ast_rtp_instance_get_hold_timeout(dialog->rtp)) {
return;
-
- /* Check AUDIO RTP keepalives */
- if (dialog->lastrtptx && ast_rtp_get_rtpkeepalive(dialog->rtp) &&
- (t > dialog->lastrtptx + ast_rtp_get_rtpkeepalive(dialog->rtp))) {
- /* Need to send an empty RTP packet */
- dialog->lastrtptx = time(NULL);
- ast_rtp_sendcng(dialog->rtp, 0);
}
/*! \todo Check video RTP keepalives
@@ -21229,16 +21238,10 @@ static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
*/
/* Check AUDIO RTP timers */
- if (dialog->lastrtprx && (ast_rtp_get_rtptimeout(dialog->rtp) || ast_rtp_get_rtpholdtimeout(dialog->rtp)) &&
- (t > dialog->lastrtprx + ast_rtp_get_rtptimeout(dialog->rtp))) {
-
- /* Might be a timeout now -- see if we're on hold */
- struct sockaddr_in sin;
- ast_rtp_get_peer(dialog->rtp, &sin);
- if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD) || (ast_rtp_get_rtpholdtimeout(dialog->rtp) &&
- (t > dialog->lastrtprx + ast_rtp_get_rtpholdtimeout(dialog->rtp)))) {
+ if (dialog->lastrtprx && (ast_rtp_instance_get_timeout(dialog->rtp) || ast_rtp_instance_get_hold_timeout(dialog->rtp)) && (t > dialog->lastrtprx + ast_rtp_instance_get_timeout(dialog->rtp))) {
+ if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD) || (ast_rtp_instance_get_hold_timeout(dialog->rtp) && (t > dialog->lastrtprx + ast_rtp_instance_get_hold_timeout(dialog->rtp)))) {
/* Needs a hangup */
- if (ast_rtp_get_rtptimeout(dialog->rtp)) {
+ if (ast_rtp_instance_get_timeout(dialog->rtp)) {
while (dialog->owner && ast_channel_trylock(dialog->owner)) {
sip_pvt_unlock(dialog);
usleep(1);
@@ -21253,11 +21256,11 @@ static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
has already been requested and we don't want to
repeatedly request hangups
*/
- ast_rtp_set_rtptimeout(dialog->rtp, 0);
- ast_rtp_set_rtpholdtimeout(dialog->rtp, 0);
+ ast_rtp_instance_set_timeout(dialog->rtp, 0);
+ ast_rtp_instance_set_hold_timeout(dialog->rtp, 0);
if (dialog->vrtp) {
- ast_rtp_set_rtptimeout(dialog->vrtp, 0);
- ast_rtp_set_rtpholdtimeout(dialog->vrtp, 0);
+ ast_rtp_instance_set_timeout(dialog->vrtp, 0);
+ ast_rtp_instance_set_hold_timeout(dialog->vrtp, 0);
}
}
}
@@ -22417,6 +22420,7 @@ static void set_peer_defaults(struct sip_peer *peer)
ast_string_field_set(peer, language, default_language);
ast_string_field_set(peer, mohinterpret, default_mohinterpret);
ast_string_field_set(peer, mohsuggest, default_mohsuggest);
+ ast_string_field_set(peer, engine, default_engine);
peer->addr.sin_family = AF_INET;
peer->defaddr.sin_family = AF_INET;
peer->capability = global_capability;
@@ -22756,6 +22760,8 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str
ast_string_field_set(peer, mohsuggest, v->value);
} else if (!strcasecmp(v->name, "parkinglot")) {
ast_string_field_set(peer, parkinglot, v->value);
+ } else if (!strcasecmp(v->name, "rtp_engine")) {
+ ast_string_field_set(peer, engine, v->value);
} else if (!strcasecmp(v->name, "mailbox")) {
add_peer_mailboxes(peer, v->value);
} else if (!strcasecmp(v->name, "hasvoicemail")) {
@@ -23205,6 +23211,7 @@ static int reload_config(enum channelreloadreason reason)
ast_set_flag(&global_flags[0], SIP_DTMF_RFC2833); /*!< Default DTMF setting: RFC2833 */
ast_set_flag(&global_flags[0], SIP_NAT_RFC3581); /*!< NAT support if requested by device with rport */
ast_set_flag(&global_flags[0], SIP_CAN_REINVITE); /*!< Allow re-invites */
+ ast_copy_string(default_engine, DEFAULT_ENGINE, sizeof(default_engine));
/* Debugging settings, always default to off */
dumphistory = FALSE;
@@ -23945,156 +23952,176 @@ static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl)
return 0;
}
-/*! \brief Returns null if we can't reinvite audio (part of RTP interface) */
-static enum ast_rtp_get_result sip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
+static enum ast_rtp_glue_result sip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
- struct sip_pvt *p = NULL;
- enum ast_rtp_get_result res = AST_RTP_TRY_PARTIAL;
+ struct sip_pvt *p = NULL;
+ enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_LOCAL;
- if (!(p = chan->tech_pvt))
- return AST_RTP_GET_FAILED;
-
- sip_pvt_lock(p);
- if (!(p->rtp)) {
- sip_pvt_unlock(p);
- return AST_RTP_GET_FAILED;
+ if (!(p = chan->tech_pvt)) {
+ return AST_RTP_GLUE_RESULT_FORBID;
}
- *rtp = p->rtp;
+ sip_pvt_lock(p);
+ if (!(p->rtp)) {
+ sip_pvt_unlock(p);
+ return AST_RTP_GLUE_RESULT_FORBID;
+ }
- if (ast_rtp_getnat(*rtp) && !ast_test_flag(&p->flags[0], SIP_CAN_REINVITE_NAT))
- res = AST_RTP_TRY_PARTIAL;
- else if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
- res = AST_RTP_TRY_NATIVE;
- else if (ast_test_flag(&global_jbconf, AST_JB_FORCED))
- res = AST_RTP_GET_FAILED;
+ ao2_ref(p->rtp, +1);
+ *instance = p->rtp;
- sip_pvt_unlock(p);
+ if (!ast_test_flag(&p->flags[0], SIP_CAN_REINVITE_NAT)) {
+ res = AST_RTP_GLUE_RESULT_LOCAL;
+ } else if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE)) {
+ res = AST_RTP_GLUE_RESULT_REMOTE;
+ } else if (ast_test_flag(&global_jbconf, AST_JB_FORCED)) {
+ res = AST_RTP_GLUE_RESULT_FORBID;
+ }
- return res;
+ sip_pvt_unlock(p);
+
+ return res;
}
-/*! \brief Returns null if we can't reinvite video (part of RTP interface) */
-static enum ast_rtp_get_result sip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
+static enum ast_rtp_glue_result sip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
struct sip_pvt *p = NULL;
- enum ast_rtp_get_result res = AST_RTP_TRY_PARTIAL;
-
- if (!(p = chan->tech_pvt))
- return AST_RTP_GET_FAILED;
+ enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_FORBID;
+
+ if (!(p = chan->tech_pvt)) {
+ return AST_RTP_GLUE_RESULT_FORBID;
+ }
sip_pvt_lock(p);
if (!(p->vrtp)) {
sip_pvt_unlock(p);
- return AST_RTP_GET_FAILED;
+ return AST_RTP_GLUE_RESULT_FORBID;
}
- *rtp = p->vrtp;
+ ao2_ref(p->vrtp, +1);
+ *instance = p->vrtp;
- if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
- res = AST_RTP_TRY_NATIVE;
+ if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE)) {
+ res = AST_RTP_GLUE_RESULT_REMOTE;
+ }
sip_pvt_unlock(p);
return res;
}
-/*! \brief Returns null if we can't reinvite text (part of RTP interface) */
-static enum ast_rtp_get_result sip_get_trtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
+static enum ast_rtp_glue_result sip_get_trtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
- struct sip_pvt *p = NULL;
- enum ast_rtp_get_result res = AST_RTP_TRY_PARTIAL;
-
- if (!(p = chan->tech_pvt))
- return AST_RTP_GET_FAILED;
+ struct sip_pvt *p = NULL;
+ enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_FORBID;
- sip_pvt_lock(p);
- if (!(p->trtp)) {
- sip_pvt_unlock(p);
- return AST_RTP_GET_FAILED;
- }
+ if (!(p = chan->tech_pvt)) {
+ return AST_RTP_GLUE_RESULT_FORBID;
+ }
- *rtp = p->trtp;
+ sip_pvt_lock(p);
+ if (!(p->trtp)) {
+ sip_pvt_unlock(p);
+ return AST_RTP_GLUE_RESULT_FORBID;
+ }
- if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
- res = AST_RTP_TRY_NATIVE;
+ ao2_ref(p->trtp, +1);
+ *instance = p->trtp;
- sip_pvt_unlock(p);
+ if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE)) {
+ res = AST_RTP_GLUE_RESULT_REMOTE;
+ }
- return res;
+ sip_pvt_unlock(p);
+
+ return res;
}
-/*! \brief Set the RTP peer for this call */
-static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active)
+static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, int codecs, int nat_active)
{
- struct sip_pvt *p;
- int changed = 0;
+ struct sip_pvt *p;
+ int changed = 0;
- p = chan->tech_pvt;
- if (!p)
- return -1;
+ p = chan->tech_pvt;
+ if (!p)
+ return -1;
/* Disable early RTP bridge */
if (chan->_state != AST_STATE_UP && !sip_cfg.directrtpsetup) /* We are in early state */
return 0;
- sip_pvt_lock(p);
- if (p->alreadygone) {
- /* If we're destroyed, don't bother */
- sip_pvt_unlock(p);
- return 0;
- }
+ sip_pvt_lock(p);
+ if (p->alreadygone) {
+ /* If we're destroyed, don't bother */
+ sip_pvt_unlock(p);
+ return 0;
+ }
- /* if this peer cannot handle reinvites of the media stream to devices
- that are known to be behind a NAT, then stop the process now
+ /* if this peer cannot handle reinvites of the media stream to devices
+ that are known to be behind a NAT, then stop the process now
*/
- if (nat_active && !ast_test_flag(&p->flags[0], SIP_CAN_REINVITE_NAT)) {
- sip_pvt_unlock(p);
- return 0;
- }
+ if (nat_active && !ast_test_flag(&p->flags[0], SIP_CAN_REINVITE_NAT)) {
+ sip_pvt_unlock(p);
+ return 0;
+ }
- if (rtp) {
- changed |= ast_rtp_get_peer(rtp, &p->redirip);
- } else if (p->redirip.sin_addr.s_addr || ntohs(p->redirip.sin_port) != 0) {
- memset(&p->redirip, 0, sizeof(p->redirip));
- changed = 1;
- }
- if (vrtp) {
- changed |= ast_rtp_get_peer(vrtp, &p->vredirip);
- } else if (p->vredirip.sin_addr.s_addr || ntohs(p->vredirip.sin_port) != 0) {
- memset(&p->vredirip, 0, sizeof(p->vredirip));
- changed = 1;
- }
- if (trtp) {
- changed |= ast_rtp_get_peer(trtp, &p->tredirip);
- } else if (p->tredirip.sin_addr.s_addr || ntohs(p->tredirip.sin_port) != 0) {
- memset(&p->tredirip, 0, sizeof(p->tredirip));
- changed = 1;
- }
- if (codecs && (p->redircodecs != codecs)) {
- p->redircodecs = codecs;
- changed = 1;
- }
- if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER) && !ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
- if (chan->_state != AST_STATE_UP) { /* We are in early state */
- if (p->do_history)
- append_history(p, "ExtInv", "Initial invite sent with remote bridge proposal.");
- ast_debug(1, "Early remote bridge setting SIP '%s' - Sending media to %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip.sin_addr));
- } else if (!p->pendinginvite) { /* We are up, and have no outstanding invite */
- ast_debug(3, "Sending reinvite on SIP '%s' - It's audio soon redirected to IP %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip.sin_addr));
- transmit_reinvite_with_sdp(p, FALSE, FALSE);
- } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
- ast_debug(3, "Deferring reinvite on SIP '%s' - It's audio will be redirected to IP %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip.sin_addr));
- /* We have a pending Invite. Send re-invite when we're done with the invite */
- ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
- }
- }
- /* Reset lastrtprx timer */
- p->lastrtprx = p->lastrtptx = time(NULL);
- sip_pvt_unlock(p);
- return 0;
+ if (instance) {
+ changed |= ast_rtp_instance_get_remote_address(instance, &p->redirip);
+ } else if (p->redirip.sin_addr.s_addr || ntohs(p->redirip.sin_port) != 0) {
+ memset(&p->redirip, 0, sizeof(p->redirip));
+ changed = 1;
+ }
+ if (vinstance) {
+ changed |= ast_rtp_instance_get_remote_address(vinstance, &p->vredirip);
+ } else if (p->vredirip.sin_addr.s_addr || ntohs(p->vredirip.sin_port) != 0) {
+ memset(&p->vredirip, 0, sizeof(p->vredirip));
+ changed = 1;
+ }
+ if (tinstance) {
+ changed |= ast_rtp_instance_get_remote_address(tinstance, &p->tredirip);
+ } else if (p->tredirip.sin_addr.s_addr || ntohs(p->tredirip.sin_port) != 0) {
+ memset(&p->tredirip, 0, sizeof(p->tredirip));
+ changed = 1;
+ }
+ if (codecs && (p->redircodecs != codecs)) {
+ p->redircodecs = codecs;
+ changed = 1;
+ }
+ if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER) && !ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
+ if (chan->_state != AST_STATE_UP) { /* We are in early state */
+ if (p->do_history)
+ append_history(p, "ExtInv", "Initial invite sent with remote bridge proposal.");
+ ast_debug(1, "Early remote bridge setting SIP '%s' - Sending media to %s\n", p->callid, ast_inet_ntoa(instance ? p->redirip.sin_addr : p->ourip.sin_addr));
+ } else if (!p->pendinginvite) { /* We are up, and have no outstanding invite */
+ ast_debug(3, "Sending reinvite on SIP '%s' - It's audio soon redirected to IP %s\n", p->callid, ast_inet_ntoa(instance ? p->redirip.sin_addr : p->ourip.sin_addr));
+ transmit_reinvite_with_sdp(p, FALSE, FALSE);
+ } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
+ ast_debug(3, "Deferring reinvite on SIP '%s' - It's audio will be redirected to IP %s\n", p->callid, ast_inet_ntoa(instance ? p->redirip.sin_addr : p->ourip.sin_addr));
+ /* We have a pending Invite. Send re-invite when we're done with the invite */
+ ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
+ }
+ }
+ /* Reset lastrtprx timer */
+ p->lastrtprx = p->lastrtptx = time(NULL);
+ sip_pvt_unlock(p);
+ return 0;
}
+static int sip_get_codec(struct ast_channel *chan)
+{
+ struct sip_pvt *p = chan->tech_pvt;
+ return p->peercapability ? p->peercapability : p->capability;
+}
+
+static struct ast_rtp_glue sip_rtp_glue = {
+ .type = "SIP",
+ .get_rtp_info = sip_get_rtp_peer,
+ .get_vrtp_info = sip_get_vrtp_peer,
+ .get_trtp_info = sip_get_trtp_peer,
+ .update_peer = sip_set_rtp_peer,
+ .get_codec = sip_get_codec,
+};
+
static char *app_dtmfmode = "SIPDtmfMode";
static char *app_sipaddheader = "SIPAddHeader";
static char *app_sipremoveheader = "SIPRemoveHeader";
@@ -24140,7 +24167,7 @@ static int sip_dtmfmode(struct ast_channel *chan, void *data)
} else
ast_log(LOG_WARNING, "I don't know about this dtmf mode: %s\n", mode);
if (p->rtp)
- ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) {
if (!p->vad) {
p->vad = ast_dsp_new();
@@ -24288,13 +24315,6 @@ static int sip_sipredirect(struct sip_pvt *p, const char *dest)
return 0;
}
-/*! \brief Return SIP UA's codec (part of the RTP interface) */
-static int sip_get_codec(struct ast_channel *chan)
-{
- struct sip_pvt *p = chan->tech_pvt;
- return p->jointcapability ? p->jointcapability : p->capability;
-}
-
/*! \brief Send a poke to all known peers */
static void sip_poke_all_peers(void)
{
@@ -24502,12 +24522,12 @@ static int load_module(void)
/* Register all CLI functions for SIP */
ast_cli_register_multiple(cli_sip, ARRAY_LEN(cli_sip));
- /* Tell the RTP subdriver that we're here */
- ast_rtp_proto_register(&sip_rtp);
-
/* Tell the UDPTL subdriver that we're here */
ast_udptl_proto_register(&sip_udptl);
+ /* Tell the RTP engine about our RTP glue */
+ ast_rtp_glue_register(&sip_rtp_glue);
+
/* Register dialplan applications */
ast_register_application_xml(app_dtmfmode, sip_dtmfmode);
ast_register_application_xml(app_sipaddheader, sip_addheader);
@@ -24578,12 +24598,12 @@ static int unload_module(void)
/* Unregister CLI commands */
ast_cli_unregister_multiple(cli_sip, ARRAY_LEN(cli_sip));
- /* Disconnect from the RTP subsystem */
- ast_rtp_proto_unregister(&sip_rtp);
-
/* Disconnect from UDPTL */
ast_udptl_proto_unregister(&sip_udptl);
+ /* Disconnect from RTP engine */
+ ast_rtp_glue_unregister(&sip_rtp_glue);
+
/* Unregister AMI actions */
ast_manager_unregister("SIPpeers");
ast_manager_unregister("SIPshowpeer");
diff --git a/channels/chan_skinny.c b/channels/chan_skinny.c
index e8330fa824..f4104a89eb 100644
--- a/channels/chan_skinny.c
+++ b/channels/chan_skinny.c
@@ -49,7 +49,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/pbx.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
-#include "asterisk/rtp.h"
+#include "asterisk/rtp_engine.h"
#include "asterisk/netsock.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
@@ -1111,8 +1111,8 @@ static int matchdigittimeout = 3000;
struct skinny_subchannel {
ast_mutex_t lock;
struct ast_channel *owner;
- struct ast_rtp *rtp;
- struct ast_rtp *vrtp;
+ struct ast_rtp_instance *rtp;
+ struct ast_rtp_instance *vrtp;
unsigned int callid;
/* time_t lastouttime; */ /* Unused */
int progress;
@@ -1347,7 +1347,7 @@ static const struct ast_channel_tech skinny_tech = {
.fixup = skinny_fixup,
.send_digit_begin = skinny_senddigit_begin,
.send_digit_end = skinny_senddigit_end,
- .bridge = ast_rtp_bridge,
+ .bridge = ast_rtp_instance_bridge,
};
static int skinny_extensionstate_cb(char *context, char* exten, int state, void *data);
@@ -2557,46 +2557,48 @@ static void mwi_event_cb(const struct ast_event *event, void *userdata)
/* I do not believe skinny can deal with video.
Anyone know differently? */
/* Yes, it can. Currently 7985 and Cisco VT Advantage do video. */
-static enum ast_rtp_get_result skinny_get_vrtp_peer(struct ast_channel *c, struct ast_rtp **rtp)
+static enum ast_rtp_glue_result skinny_get_vrtp_peer(struct ast_channel *c, struct ast_rtp_instance **instance)
{
struct skinny_subchannel *sub = NULL;
if (!(sub = c->tech_pvt) || !(sub->vrtp))
- return AST_RTP_GET_FAILED;
+ return AST_RTP_GLUE_RESULT_FORBID;
- *rtp = sub->vrtp;
+ ao2_ref(sub->vrtp, +1);
+ *instance = sub->vrtp;
- return AST_RTP_TRY_NATIVE;
+ return AST_RTP_GLUE_RESULT_REMOTE;
}
-static enum ast_rtp_get_result skinny_get_rtp_peer(struct ast_channel *c, struct ast_rtp **rtp)
+static enum ast_rtp_glue_result skinny_get_rtp_peer(struct ast_channel *c, struct ast_rtp_instance **instance)
{
struct skinny_subchannel *sub = NULL;
struct skinny_line *l;
- enum ast_rtp_get_result res = AST_RTP_TRY_NATIVE;
+ enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_REMOTE;
if (skinnydebug)
ast_verb(1, "skinny_get_rtp_peer() Channel = %s\n", c->name);
if (!(sub = c->tech_pvt))
- return AST_RTP_GET_FAILED;
+ return AST_RTP_GLUE_RESULT_FORBID;
ast_mutex_lock(&sub->lock);
if (!(sub->rtp)){
ast_mutex_unlock(&sub->lock);
- return AST_RTP_GET_FAILED;
+ return AST_RTP_GLUE_RESULT_FORBID;
}
-
- *rtp = sub->rtp;
+
+ ao2_ref(sub->rtp, +1);
+ *instance = sub->rtp;
l = sub->parent;
if (!l->canreinvite || l->nat){
- res = AST_RTP_TRY_PARTIAL;
+ res = AST_RTP_GLUE_RESULT_LOCAL;
if (skinnydebug)
- ast_verb(1, "skinny_get_rtp_peer() Using AST_RTP_TRY_PARTIAL \n");
+ ast_verb(1, "skinny_get_rtp_peer() Using AST_RTP_GLUE_RESULT_LOCAL \n");
}
ast_mutex_unlock(&sub->lock);
@@ -2605,7 +2607,7 @@ static enum ast_rtp_get_result skinny_get_rtp_peer(struct ast_channel *c, struct
}
-static int skinny_set_rtp_peer(struct ast_channel *c, struct ast_rtp *rtp, struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active)
+static int skinny_set_rtp_peer(struct ast_channel *c, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *trtp, int codecs, int nat_active)
{
struct skinny_subchannel *sub;
struct skinny_line *l;
@@ -2630,7 +2632,7 @@ static int skinny_set_rtp_peer(struct ast_channel *c, struct ast_rtp *rtp, struc
s = d->session;
if (rtp){
- ast_rtp_get_peer(rtp, &them);
+ ast_rtp_instance_get_remote_address(rtp, &them);
/* Shutdown any early-media or previous media on re-invite */
if (!(req = req_alloc(sizeof(struct stop_media_transmission_message), STOP_MEDIA_TRANSMISSION_MESSAGE)))
@@ -2654,7 +2656,7 @@ static int skinny_set_rtp_peer(struct ast_channel *c, struct ast_rtp *rtp, struc
req->data.startmedia.conferenceId = htolel(sub->callid);
req->data.startmedia.passThruPartyId = htolel(sub->callid);
if (!(l->canreinvite) || (l->nat)){
- ast_rtp_get_us(rtp, &us);
+ ast_rtp_instance_get_local_address(rtp, &us);
req->data.startmedia.remoteIp = htolel(d->ourip.s_addr);
req->data.startmedia.remotePort = htolel(ntohs(us.sin_port));
} else {
@@ -2675,11 +2677,11 @@ static int skinny_set_rtp_peer(struct ast_channel *c, struct ast_rtp *rtp, struc
return 0;
}
-static struct ast_rtp_protocol skinny_rtp = {
+static struct ast_rtp_glue skinny_rtp_glue = {
.type = "Skinny",
.get_rtp_info = skinny_get_rtp_peer,
.get_vrtp_info = skinny_get_vrtp_peer,
- .set_rtp_peer = skinny_set_rtp_peer,
+ .update_peer = skinny_set_rtp_peer,
};
static char *handle_skinny_set_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
@@ -3559,29 +3561,36 @@ static void start_rtp(struct skinny_subchannel *sub)
ast_mutex_lock(&sub->lock);
/* Allocate the RTP */
- sub->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
+ sub->rtp = ast_rtp_instance_new(NULL, sched, &bindaddr, NULL);
if (hasvideo)
- sub->vrtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
-
+ sub->vrtp = ast_rtp_instance_new(NULL, sched, &bindaddr, NULL);
+
+ if (sub->rtp) {
+ ast_rtp_instance_set_prop(sub->rtp, AST_RTP_PROPERTY_RTCP, 1);
+ }
+ if (sub->vrtp) {
+ ast_rtp_instance_set_prop(sub->vrtp, AST_RTP_PROPERTY_RTCP, 1);
+ }
+
if (sub->rtp && sub->owner) {
- ast_channel_set_fd(sub->owner, 0, ast_rtp_fd(sub->rtp));
- ast_channel_set_fd(sub->owner, 1, ast_rtcp_fd(sub->rtp));
+ ast_channel_set_fd(sub->owner, 0, ast_rtp_instance_fd(sub->rtp, 0));
+ ast_channel_set_fd(sub->owner, 1, ast_rtp_instance_fd(sub->rtp, 1));
}
if (hasvideo && sub->vrtp && sub->owner) {
- ast_channel_set_fd(sub->owner, 2, ast_rtp_fd(sub->vrtp));
- ast_channel_set_fd(sub->owner, 3, ast_rtcp_fd(sub->vrtp));
+ ast_channel_set_fd(sub->owner, 2, ast_rtp_instance_fd(sub->vrtp, 0));
+ ast_channel_set_fd(sub->owner, 3, ast_rtp_instance_fd(sub->vrtp, 1));
}
if (sub->rtp) {
- ast_rtp_setqos(sub->rtp, qos.tos_audio, qos.cos_audio, "Skinny RTP");
- ast_rtp_setnat(sub->rtp, l->nat);
+ ast_rtp_instance_set_qos(sub->rtp, qos.tos_audio, qos.cos_audio, "Skinny RTP");
+ ast_rtp_instance_set_prop(sub->rtp, AST_RTP_PROPERTY_NAT, l->nat);
}
if (sub->vrtp) {
- ast_rtp_setqos(sub->vrtp, qos.tos_video, qos.cos_video, "Skinny VRTP");
- ast_rtp_setnat(sub->vrtp, l->nat);
+ ast_rtp_instance_set_qos(sub->vrtp, qos.tos_video, qos.cos_video, "Skinny VRTP");
+ ast_rtp_instance_set_prop(sub->vrtp, AST_RTP_PROPERTY_NAT, l->nat);
}
/* Set Frame packetization */
if (sub->rtp)
- ast_rtp_codec_setpref(sub->rtp, &l->prefs);
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(sub->rtp), sub->rtp, &l->prefs);
/* Create the RTP connection */
transmit_connect(d, sub);
@@ -3852,7 +3861,7 @@ static int skinny_hangup(struct ast_channel *ast)
sub->alreadygone = 0;
sub->outgoing = 0;
if (sub->rtp) {
- ast_rtp_destroy(sub->rtp);
+ ast_rtp_instance_destroy(sub->rtp);
sub->rtp = NULL;
}
ast_mutex_unlock(&sub->lock);
@@ -3913,16 +3922,16 @@ static struct ast_frame *skinny_rtp_read(struct skinny_subchannel *sub)
switch(ast->fdno) {
case 0:
- f = ast_rtp_read(sub->rtp); /* RTP Audio */
+ f = ast_rtp_instance_read(sub->rtp, 0); /* RTP Audio */
break;
case 1:
- f = ast_rtcp_read(sub->rtp); /* RTCP Control Channel */
+ f = ast_rtp_instance_read(sub->rtp, 1); /* RTCP Control Channel */
break;
case 2:
- f = ast_rtp_read(sub->vrtp); /* RTP Video */
+ f = ast_rtp_instance_read(sub->vrtp, 0); /* RTP Video */
break;
case 3:
- f = ast_rtcp_read(sub->vrtp); /* RTCP Control Channel for video */
+ f = ast_rtp_instance_read(sub->vrtp, 1); /* RTCP Control Channel for video */
break;
#if 0
case 5:
@@ -3979,7 +3988,7 @@ static int skinny_write(struct ast_channel *ast, struct ast_frame *frame)
if (sub) {
ast_mutex_lock(&sub->lock);
if (sub->rtp) {
- res = ast_rtp_write(sub->rtp, frame);
+ res = ast_rtp_instance_write(sub->rtp, frame);
}
ast_mutex_unlock(&sub->lock);
}
@@ -4253,7 +4262,7 @@ static int skinny_indicate(struct ast_channel *ast, int ind, const void *data, s
case AST_CONTROL_PROCEEDING:
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_new_source(sub->rtp);
+ ast_rtp_instance_new_source(sub->rtp);
break;
default:
ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", ind);
@@ -4312,7 +4321,7 @@ static struct ast_channel *skinny_new(struct skinny_line *l, int state)
if (skinnydebug)
ast_verb(1, "skinny_new: tmp->nativeformats=%d fmt=%d\n", tmp->nativeformats, fmt);
if (sub->rtp) {
- ast_channel_set_fd(tmp, 0, ast_rtp_fd(sub->rtp));
+ ast_channel_set_fd(tmp, 0, ast_rtp_instance_fd(sub->rtp, 0));
}
if (state == AST_STATE_RING) {
tmp->rings = 1;
@@ -5537,8 +5546,8 @@ static int handle_open_receive_channel_ack_message(struct skinny_req *req, struc
l = sub->parent;
if (sub->rtp) {
- ast_rtp_set_peer(sub->rtp, &sin);
- ast_rtp_get_us(sub->rtp, &us);
+ ast_rtp_instance_set_remote_address(sub->rtp, &sin);
+ ast_rtp_instance_get_local_address(sub->rtp, &us);
} else {
ast_log(LOG_ERROR, "No RTP structure, this is very bad\n");
return 0;
@@ -7289,7 +7298,7 @@ static int load_module(void)
return -1;
}
- ast_rtp_proto_register(&skinny_rtp);
+ ast_rtp_glue_register(&skinny_rtp_glue);
ast_cli_register_multiple(cli_skinny, ARRAY_LEN(cli_skinny));
ast_manager_register2("SKINNYdevices", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_skinny_show_devices,
@@ -7323,7 +7332,7 @@ static int unload_module(void)
struct skinny_subchannel *sub;
struct ast_context *con;
- ast_rtp_proto_unregister(&skinny_rtp);
+ ast_rtp_glue_unregister(&skinny_rtp_glue);
ast_channel_unregister(&skinny_tech);
ast_cli_unregister_multiple(cli_skinny, ARRAY_LEN(cli_skinny));
diff --git a/channels/chan_unistim.c b/channels/chan_unistim.c
index 818a32d71d..1cd94e02fc 100644
--- a/channels/chan_unistim.c
+++ b/channels/chan_unistim.c
@@ -60,7 +60,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/event.h"
-#include "asterisk/rtp.h"
+#include "asterisk/rtp_engine.h"
#include "asterisk/netsock.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
@@ -365,7 +365,7 @@ struct unistim_subchannel {
/*! Unistim line */
struct unistim_line *parent;
/*! RTP handle */
- struct ast_rtp *rtp;
+ struct ast_rtp_instance *rtp;
int alreadygone;
char ringvolume;
char ringstyle;
@@ -711,7 +711,7 @@ static const struct ast_channel_tech unistim_tech = {
.send_digit_begin = unistim_senddigit_begin,
.send_digit_end = unistim_senddigit_end,
.send_text = unistim_sendtext,
-/* .bridge = ast_rtp_bridge, */
+ .bridge = ast_rtp_instance_bridge,
};
static void display_last_error(const char *sz_msg)
@@ -1854,7 +1854,7 @@ static void cancel_dial(struct unistimsession *pte)
static void swap_subs(struct unistim_line *p, int a, int b)
{
/* struct ast_channel *towner; */
- struct ast_rtp *rtp;
+ struct ast_rtp_instance *rtp;
int fds;
if (unistimdebug)
@@ -2056,30 +2056,29 @@ static void start_rtp(struct unistim_subchannel *sub)
/* Allocate the RTP */
if (unistimdebug)
ast_verb(0, "Starting RTP. Bind on %s\n", ast_inet_ntoa(sout.sin_addr));
- sub->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, sout.sin_addr);
+ sub->rtp = ast_rtp_instance_new(NULL, sched, &sout, NULL);
if (!sub->rtp) {
ast_log(LOG_WARNING, "Unable to create RTP session: %s binaddr=%s\n",
strerror(errno), ast_inet_ntoa(sout.sin_addr));
ast_mutex_unlock(&sub->lock);
return;
}
- if (sub->rtp && sub->owner) {
- sub->owner->fds[0] = ast_rtp_fd(sub->rtp);
- sub->owner->fds[1] = ast_rtcp_fd(sub->rtp);
- }
- if (sub->rtp) {
- ast_rtp_setqos(sub->rtp, qos.tos_audio, qos.cos_audio, "UNISTIM RTP");
- ast_rtp_setnat(sub->rtp, sub->parent->parent->nat);
+ ast_rtp_instance_set_prop(sub->rtp, AST_RTP_PROPERTY_RTCP, 1);
+ if (sub->owner) {
+ sub->owner->fds[0] = ast_rtp_instance_fd(sub->rtp, 0);
+ sub->owner->fds[1] = ast_rtp_instance_fd(sub->rtp, 1);
}
+ ast_rtp_instance_set_qos(sub->rtp, qos.tos_audio, qos.cos_audio, "UNISTIM RTP");
+ ast_rtp_instance_set_prop(sub->rtp, AST_RTP_PROPERTY_NAT, sub->parent->parent->nat);
/* Create the RTP connection */
- ast_rtp_get_us(sub->rtp, &us);
+ ast_rtp_instance_get_local_address(sub->rtp, &us);
sin.sin_family = AF_INET;
/* Setting up RTP for our side */
memcpy(&sin.sin_addr, &sub->parent->parent->session->sin.sin_addr,
sizeof(sin.sin_addr));
sin.sin_port = htons(sub->parent->parent->rtp_port);
- ast_rtp_set_peer(sub->rtp, &sin);
+ ast_rtp_instance_set_remote_address(sub->rtp, &sin);
if (!(sub->owner->nativeformats & sub->owner->readformat)) {
int fmt;
fmt = ast_best_codec(sub->owner->nativeformats);
@@ -2091,7 +2090,7 @@ static void start_rtp(struct unistim_subchannel *sub)
sub->owner->readformat = fmt;
sub->owner->writeformat = fmt;
}
- codec = ast_rtp_lookup_code(sub->rtp, 1, sub->owner->readformat);
+ codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(sub->rtp), 1, sub->owner->readformat);
/* Setting up RTP of the phone */
if (public_ip.sin_family == 0) /* NAT IP override ? */
memcpy(&public, &us, sizeof(public)); /* No defined, using IP from recvmsg */
@@ -3724,7 +3723,7 @@ static int unistim_hangup(struct ast_channel *ast)
if (sub->rtp) {
if (unistimdebug)
ast_verb(0, "Destroying RTP session\n");
- ast_rtp_destroy(sub->rtp);
+ ast_rtp_instance_destroy(sub->rtp);
sub->rtp = NULL;
}
return 0;
@@ -3769,7 +3768,7 @@ static int unistim_hangup(struct ast_channel *ast)
if (sub->rtp) {
if (unistimdebug)
ast_verb(0, "Destroying RTP session\n");
- ast_rtp_destroy(sub->rtp);
+ ast_rtp_instance_destroy(sub->rtp);
sub->rtp = NULL;
}
return 0;
@@ -3794,7 +3793,7 @@ static int unistim_hangup(struct ast_channel *ast)
if (sub->rtp) {
if (unistimdebug)
ast_verb(0, "Destroying RTP session\n");
- ast_rtp_destroy(sub->rtp);
+ ast_rtp_instance_destroy(sub->rtp);
sub->rtp = NULL;
} else if (unistimdebug)
ast_verb(0, "No RTP session to destroy\n");
@@ -3921,10 +3920,10 @@ static struct ast_frame *unistim_rtp_read(const struct ast_channel *ast,
switch (ast->fdno) {
case 0:
- f = ast_rtp_read(sub->rtp); /* RTP Audio */
+ f = ast_rtp_instance_read(sub->rtp, 0); /* RTP Audio */
break;
case 1:
- f = ast_rtcp_read(sub->rtp); /* RTCP Control Channel */
+ f = ast_rtp_instance_read(sub->rtp, 1); /* RTCP Control Channel */
break;
default:
f = &ast_null_frame;
@@ -3990,7 +3989,7 @@ static int unistim_write(struct ast_channel *ast, struct ast_frame *frame)
if (sub) {
ast_mutex_lock(&sub->lock);
if (sub->rtp) {
- res = ast_rtp_write(sub->rtp, frame);
+ res = ast_rtp_instance_write(sub->rtp, frame);
}
ast_mutex_unlock(&sub->lock);
}
@@ -4455,8 +4454,8 @@ static struct ast_channel *unistim_new(struct unistim_subchannel *sub, int state
if ((sub->rtp) && (sub->subtype == 0)) {
if (unistimdebug)
ast_verb(0, "New unistim channel with a previous rtp handle ?\n");
- tmp->fds[0] = ast_rtp_fd(sub->rtp);
- tmp->fds[1] = ast_rtcp_fd(sub->rtp);
+ tmp->fds[0] = ast_rtp_instance_fd(sub->rtp, 0);
+ tmp->fds[1] = ast_rtp_instance_fd(sub->rtp, 1);
}
if (sub->rtp)
ast_jb_configure(tmp, &global_jbconf);
@@ -5526,51 +5525,19 @@ static int reload_config(void)
return 0;
}
-static enum ast_rtp_get_result unistim_get_vrtp_peer(struct ast_channel *chan,
- struct ast_rtp **rtp)
-{
- return AST_RTP_TRY_NATIVE;
-}
-
-static enum ast_rtp_get_result unistim_get_rtp_peer(struct ast_channel *chan,
- struct ast_rtp **rtp)
-{
- struct unistim_subchannel *sub;
- enum ast_rtp_get_result res = AST_RTP_GET_FAILED;
-
- if (unistimdebug)
- ast_verb(0, "unistim_get_rtp_peer called\n");
-
- sub = chan->tech_pvt;
- if (sub && sub->rtp) {
- *rtp = sub->rtp;
- res = AST_RTP_TRY_NATIVE;
- }
-
- return res;
-}
-
-static int unistim_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp,
- struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active)
+static enum ast_rtp_glue_result unistim_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
- struct unistim_subchannel *sub;
-
- if (unistimdebug)
- ast_verb(0, "unistim_set_rtp_peer called\n");
-
- sub = chan->tech_pvt;
+ struct unistim_subchannel *sub = chan->tech_pvt;
- if (sub)
- return 0;
+ ao2_ref(sub->rtp, +1);
+ *instance = sub->rtp;
- return -1;
+ return AST_RTP_GLUE_RESULT_LOCAL;
}
-static struct ast_rtp_protocol unistim_rtp = {
+static struct ast_rtp_glue unistim_rtp_glue = {
.type = channel_type,
.get_rtp_info = unistim_get_rtp_peer,
- .get_vrtp_info = unistim_get_vrtp_peer,
- .set_rtp_peer = unistim_set_rtp_peer,
};
/*--- load_module: PBX load module - initialization ---*/
@@ -5603,7 +5570,7 @@ int load_module(void)
goto chanreg_failed;
}
- ast_rtp_proto_register(&unistim_rtp);
+ ast_rtp_glue_register(&unistim_rtp_glue);
ast_cli_register_multiple(unistim_cli, ARRAY_LEN(unistim_cli));
@@ -5634,7 +5601,7 @@ static int unload_module(void)
ast_cli_unregister_multiple(unistim_cli, ARRAY_LEN(unistim_cli));
ast_channel_unregister(&unistim_tech);
- ast_rtp_proto_unregister(&unistim_rtp);
+ ast_rtp_glue_unregister(&unistim_rtp_glue);
ast_mutex_lock(&monlock);
if (monitor_thread && (monitor_thread != AST_PTHREADT_STOP) && (monitor_thread != AST_PTHREADT_NULL)) {
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 37fcb7405a..3785618e3c 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -292,6 +292,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;contactpermit=172.16.0.0/255.255.0.0 ; restrict at what IPs your users may
; register their phones.
+;engine=asterisk ; RTP engine to use when communicating with the device
+
;
; If regcontext is specified, Asterisk will dynamically create and destroy a
; NoOp priority 1 extension for a given peer who registers or unregisters with
diff --git a/include/asterisk/_private.h b/include/asterisk/_private.h
index c709b1f152..c7e195fe6a 100644
--- a/include/asterisk/_private.h
+++ b/include/asterisk/_private.h
@@ -41,6 +41,7 @@ int ast_tps_init(void); /*!< Provided by taskprocessor.c */
int ast_timing_init(void); /*!< Provided by timing.c */
int ast_indications_init(void); /*!< Provided by indications.c */
int ast_indications_reload(void);/*!< Provided by indications.c */
+void ast_stun_init(void); /*!< Provided by stun.c */
/*!
* \brief Reload asterisk modules.
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
deleted file mode 100644
index c42906a064..0000000000
--- a/include/asterisk/rtp.h
+++ /dev/null
@@ -1,416 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 1999 - 2006, Digium, Inc.
- *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*!
- * \file rtp.h
- * \brief Supports RTP and RTCP with Symmetric RTP support for NAT traversal.
- *
- * RTP is defined in RFC 3550.
- */
-
-#ifndef _ASTERISK_RTP_H
-#define _ASTERISK_RTP_H
-
-#include "asterisk/network.h"
-
-#include "asterisk/frame.h"
-#include "asterisk/io.h"
-#include "asterisk/sched.h"
-#include "asterisk/channel.h"
-#include "asterisk/linkedlists.h"
-
-#if defined(__cplusplus) || defined(c_plusplus)
-extern "C" {
-#endif
-
-/* Codes for RTP-specific data - not defined by our AST_FORMAT codes */
-/*! DTMF (RFC2833) */
-#define AST_RTP_DTMF (1 << 0)
-/*! 'Comfort Noise' (RFC3389) */
-#define AST_RTP_CN (1 << 1)
-/*! DTMF (Cisco Proprietary) */
-#define AST_RTP_CISCO_DTMF (1 << 2)
-/*! Maximum RTP-specific code */
-#define AST_RTP_MAX AST_RTP_CISCO_DTMF
-
-/*! Maxmum number of payload defintions for a RTP session */
-#define MAX_RTP_PT 256
-
-/*! T.140 Redundancy Maxium number of generations */
-#define RED_MAX_GENERATION 5
-
-#define FLAG_3389_WARNING (1 << 0)
-
-enum ast_rtp_options {
- AST_RTP_OPT_G726_NONSTANDARD = (1 << 0),
-};
-
-enum ast_rtp_get_result {
- /*! Failed to find the RTP structure */
- AST_RTP_GET_FAILED = 0,
- /*! RTP structure exists but true native bridge can not occur so try partial */
- AST_RTP_TRY_PARTIAL,
- /*! RTP structure exists and native bridge can occur */
- AST_RTP_TRY_NATIVE,
-};
-
-/*! \brief Variables used in ast_rtcp_get function */
-enum ast_rtp_qos_vars {
- AST_RTP_TXCOUNT,
- AST_RTP_RXCOUNT,
- AST_RTP_TXJITTER,
- AST_RTP_RXJITTER,
- AST_RTP_RXPLOSS,
- AST_RTP_TXPLOSS,
- AST_RTP_RTT
-};
-
-struct ast_rtp;
-/*! T.140 Redundancy structure*/
-struct rtp_red;
-
-/*! \brief The value of each payload format mapping: */
-struct rtpPayloadType {
- int isAstFormat; /*!< whether the following code is an AST_FORMAT */
- int code;
-};
-
-/*! \brief This is the structure that binds a channel (SIP/Jingle/H.323) to the RTP subsystem
-*/
-struct ast_rtp_protocol {
- /*! Get RTP struct, or NULL if unwilling to transfer */
- enum ast_rtp_get_result (* const get_rtp_info)(struct ast_channel *chan, struct ast_rtp **rtp);
- /*! Get RTP struct, or NULL if unwilling to transfer */
- enum ast_rtp_get_result (* const get_vrtp_info)(struct ast_channel *chan, struct ast_rtp **rtp);
- /*! Get RTP struct, or NULL if unwilling to transfer */
- enum ast_rtp_get_result (* const get_trtp_info)(struct ast_channel *chan, struct ast_rtp **rtp);
- /*! Set RTP peer */
- int (* const set_rtp_peer)(struct ast_channel *chan, struct ast_rtp *peer, struct ast_rtp *vpeer, struct ast_rtp *tpeer, int codecs, int nat_active);
- int (* const get_codec)(struct ast_channel *chan);
- const char * const type;
- AST_LIST_ENTRY(ast_rtp_protocol) list;
-};
-
-enum ast_rtp_quality_type {
- RTPQOS_SUMMARY = 0,
- RTPQOS_JITTER,
- RTPQOS_LOSS,
- RTPQOS_RTT
-};
-
-/*! \brief RTCP quality report storage */
-struct ast_rtp_quality {
- unsigned int local_ssrc; /*!< Our SSRC */
- unsigned int local_lostpackets; /*!< Our lost packets */
- double local_jitter; /*!< Our calculated jitter */
- unsigned int local_count; /*!< Number of received packets */
- unsigned int remote_ssrc; /*!< Their SSRC */
- unsigned int remote_lostpackets; /*!< Their lost packets */
- double remote_jitter; /*!< Their reported jitter */
- unsigned int remote_count; /*!< Number of transmitted packets */
- double rtt; /*!< Round trip time */
-};
-
-/*! RTP callback structure */
-typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data);
-
-/*!
- * \brief Get the amount of space required to hold an RTP session
- * \return number of bytes required
- */
-size_t ast_rtp_alloc_size(void);
-
-/*!
- * \brief Initializate a RTP session.
- *
- * \param sched
- * \param io
- * \param rtcpenable
- * \param callbackmode
- * \return A representation (structure) of an RTP session.
- */
-struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode);
-
-/*!
- * \brief Initializate a RTP session using an in_addr structure.
- *
- * This fuction gets called by ast_rtp_new().
- *
- * \param sched
- * \param io
- * \param rtcpenable
- * \param callbackmode
- * \param in
- * \return A representation (structure) of an RTP session.
- */
-struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode, struct in_addr in);
-
-void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
-
-/* Copies from rtp to them and returns 1 if there was a change or 0 if it was already the same */
-int ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
-
-void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us);
-
-struct ast_rtp *ast_rtp_get_bridged(struct ast_rtp *rtp);
-
-/*! Destroy RTP session */
-void ast_rtp_destroy(struct ast_rtp *rtp);
-
-void ast_rtp_reset(struct ast_rtp *rtp);
-
-/*! Stop RTP session, do not destroy structure */
-void ast_rtp_stop(struct ast_rtp *rtp);
-
-void ast_rtp_set_callback(struct ast_rtp *rtp, ast_rtp_callback callback);
-
-void ast_rtp_set_data(struct ast_rtp *rtp, void *data);
-
-int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *f);
-
-struct ast_frame *ast_rtp_read(struct ast_rtp *rtp);
-
-struct ast_frame *ast_rtcp_read(struct ast_rtp *rtp);
-
-int ast_rtp_fd(struct ast_rtp *rtp);
-
-int ast_rtcp_fd(struct ast_rtp *rtp);
-
-int ast_rtp_senddigit_begin(struct ast_rtp *rtp, char digit);
-
-int ast_rtp_senddigit_end(struct ast_rtp *rtp, char digit);
-
-int ast_rtp_sendcng(struct ast_rtp *rtp, int level);
-
-int ast_rtp_setqos(struct ast_rtp *rtp, int tos, int cos, char *desc);
-
-void ast_rtp_new_source(struct ast_rtp *rtp);
-
-/*! \brief Setting RTP payload types from lines in a SDP description: */
-void ast_rtp_pt_clear(struct ast_rtp* rtp);
-/*! \brief Set payload types to defaults */
-void ast_rtp_pt_default(struct ast_rtp* rtp);
-
-/*! \brief Copy payload types between RTP structures */
-void ast_rtp_pt_copy(struct ast_rtp *dest, struct ast_rtp *src);
-
-/*! \brief Activate payload type */
-void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt);
-
-/*! \brief clear payload type */
-void ast_rtp_unset_m_type(struct ast_rtp* rtp, int pt);
-
-/*! \brief Set payload type to a known MIME media type for a codec
- *
- * \param rtp RTP structure to modify
- * \param pt Payload type entry to modify
- * \param mimeType top-level MIME type of media stream (typically "audio", "video", "text", etc.)
- * \param mimeSubtype MIME subtype of media stream (typically a codec name)
- * \param options Zero or more flags from the ast_rtp_options enum
- *
- * This function 'fills in' an entry in the list of possible formats for
- * a media stream associated with an RTP structure.
- *
- * \retval 0 on success
- * \retval -1 if the payload type is out of range
- * \retval -2 if the mimeType/mimeSubtype combination was not found
- */
-int ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
- char *mimeType, char *mimeSubtype,
- enum ast_rtp_options options);
-
-/*! \brief Set payload type to a known MIME media type for a codec with a specific sample rate
- *
- * \param rtp RTP structure to modify
- * \param pt Payload type entry to modify
- * \param mimeType top-level MIME type of media stream (typically "audio", "video", "text", etc.)
- * \param mimeSubtype MIME subtype of media stream (typically a codec name)
- * \param options Zero or more flags from the ast_rtp_options enum
- * \param sample_rate The sample rate of the media stream
- *
- * This function 'fills in' an entry in the list of possible formats for
- * a media stream associated with an RTP structure.
- *
- * \retval 0 on success
- * \retval -1 if the payload type is out of range
- * \retval -2 if the mimeType/mimeSubtype combination was not found
- */
-int ast_rtp_set_rtpmap_type_rate(struct ast_rtp* rtp, int pt,
- char *mimeType, char *mimeSubtype,
- enum ast_rtp_options options,
- unsigned int sample_rate);
-
-/*! \brief Mapping between RTP payload format codes and Asterisk codes: */
-struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt);
-int ast_rtp_lookup_code(struct ast_rtp* rtp, int isAstFormat, int code);
-
-void ast_rtp_get_current_formats(struct ast_rtp* rtp,
- int* astFormats, int* nonAstFormats);
-
-/*! \brief Mapping an Asterisk code into a MIME subtype (string): */
-const char *ast_rtp_lookup_mime_subtype(int isAstFormat, int code,
- enum ast_rtp_options options);
-
-/*! \brief Get the sample rate associated with known RTP payload types
- *
- * \param isAstFormat True if the value in the 'code' parameter is an AST_FORMAT value
- * \param code Format code, either from AST_FORMAT list or from AST_RTP list
- *
- * \return the sample rate if the format was found, zero if it was not found
- */
-unsigned int ast_rtp_lookup_sample_rate(int isAstFormat, int code);
-
-/*! \brief Build a string of MIME subtype names from a capability list */
-char *ast_rtp_lookup_mime_multiple(char *buf, size_t size, const int capability,
- const int isAstFormat, enum ast_rtp_options options);
-
-void ast_rtp_setnat(struct ast_rtp *rtp, int nat);
-
-int ast_rtp_getnat(struct ast_rtp *rtp);
-
-/*! \brief Indicate whether this RTP session is carrying DTMF or not */
-void ast_rtp_setdtmf(struct ast_rtp *rtp, int dtmf);
-
-/*! \brief Compensate for devices that send RFC2833 packets all at once */
-void ast_rtp_setdtmfcompensate(struct ast_rtp *rtp, int compensate);
-
-/*! \brief Enable STUN capability */
-void ast_rtp_setstun(struct ast_rtp *rtp, int stun_enable);
-
-/*! \brief Generic STUN request
- * send a generic stun request to the server specified.
- * \param s the socket used to send the request
- * \param dst the address of the STUN server
- * \param username if non null, add the username in the request
- * \param answer if non null, the function waits for a response and
- * puts here the externally visible address.
- * \return 0 on success, other values on error.
- * The interface it may change in the future.
- */
-int ast_stun_request(int s, struct sockaddr_in *dst,
- const char *username, struct sockaddr_in *answer);
-
-/*! \brief Send STUN request for an RTP socket
- * Deprecated, this is just a wrapper for ast_rtp_stun_request()
- */
-void ast_rtp_stun_request(struct ast_rtp *rtp, struct sockaddr_in *suggestion, const char *username);
-
-/*! \brief The RTP bridge.
- \arg \ref AstRTPbridge
-*/
-int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms);
-
-/*! \brief Register an RTP channel client */
-int ast_rtp_proto_register(struct ast_rtp_protocol *proto);
-
-/*! \brief Unregister an RTP channel client */
-void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto);
-
-int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media);
-
-/*! \brief If possible, create an early bridge directly between the devices without
- having to send a re-invite later */
-int ast_rtp_early_bridge(struct ast_channel *c0, struct ast_channel *c1);
-
-/*! \brief Get QOS stats on a RTP channel
- * \since 1.6.1
- */
-int ast_rtp_get_qos(struct ast_rtp *rtp, const char *qos, char *buf, unsigned int buflen);
-
-/*! \brief Return RTP and RTCP QoS values
- * \since 1.6.1
- */
-unsigned int ast_rtp_get_qosvalue(struct ast_rtp *rtp, enum ast_rtp_qos_vars value);
-
-/*! \brief Set RTPAUDIOQOS(...) variables on a channel when it is being hung up
- * \since 1.6.1
- */
-void ast_rtp_set_vars(struct ast_channel *chan, struct ast_rtp *rtp);
-
-/*! \brief Return RTCP quality string
- *
- * \param rtp An rtp structure to get qos information about.
- *
- * \param qual An (optional) rtp quality structure that will be
- * filled with the quality information described in
- * the ast_rtp_quality structure. This structure is
- * not dependent on any qtype, so a call for any
- * type of information would yield the same results
- * because ast_rtp_quality is not a data type
- * specific to any qos type.
- *
- * \param qtype The quality type you'd like, default should be
- * RTPQOS_SUMMARY which returns basic information
- * about the call. The return from RTPQOS_SUMMARY
- * is basically ast_rtp_quality in a string. The
- * other types are RTPQOS_JITTER, RTPQOS_LOSS and
- * RTPQOS_RTT which will return more specific
- * statistics.
- * \version 1.6.1 added qtype parameter
- */
-char *ast_rtp_get_quality(struct ast_rtp *rtp, struct ast_rtp_quality *qual, enum ast_rtp_quality_type qtype);
-/*! \brief Send an H.261 fast update request. Some devices need this rather than the XML message in SIP */
-int ast_rtcp_send_h261fur(void *data);
-
-void ast_rtp_init(void); /*! Initialize RTP subsystem */
-int ast_rtp_reload(void); /*! reload rtp configuration */
-void ast_rtp_new_init(struct ast_rtp *rtp);
-
-/*! \brief Set codec preference */
-void ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs);
-
-/*! \brief Get codec preference */
-struct ast_codec_pref *ast_rtp_codec_getpref(struct ast_rtp *rtp);
-
-/*! \brief get format from predefined dynamic payload format */
-int ast_rtp_codec_getformat(int pt);
-
-/*! \brief Set rtp timeout */
-void ast_rtp_set_rtptimeout(struct ast_rtp *rtp, int timeout);
-/*! \brief Set rtp hold timeout */
-void ast_rtp_set_rtpholdtimeout(struct ast_rtp *rtp, int timeout);
-/*! \brief set RTP keepalive interval */
-void ast_rtp_set_rtpkeepalive(struct ast_rtp *rtp, int period);
-/*! \brief Get RTP keepalive interval */
-int ast_rtp_get_rtpkeepalive(struct ast_rtp *rtp);
-/*! \brief Get rtp hold timeout */
-int ast_rtp_get_rtpholdtimeout(struct ast_rtp *rtp);
-/*! \brief Get rtp timeout */
-int ast_rtp_get_rtptimeout(struct ast_rtp *rtp);
-/* \brief Put RTP timeout timers on hold during another transaction, like T.38 */
-void ast_rtp_set_rtptimers_onhold(struct ast_rtp *rtp);
-
-/*! \brief Initalize t.140 redudancy
- * \param ti time between each t140red frame is sent
- * \param red_pt payloadtype for RTP packet
- * \param pt payloadtype numbers for each generation including primary data
- * \param num_gen number of redundant generations, primary data excluded
- * \since 1.6.1
- */
-int ast_rtp_red_init(struct ast_rtp *rtp, int ti, int *pt, int num_gen);
-
-/*! \brief Buffer t.140 data */
-void ast_red_buffer_t140(struct ast_rtp *rtp, struct ast_frame *f);
-
-
-
-#if defined(__cplusplus) || defined(c_plusplus)
-}
-#endif
-
-#endif /* _ASTERISK_RTP_H */
diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h
new file mode 100644
index 0000000000..edd7d1c47c
--- /dev/null
+++ b/include/asterisk/rtp_engine.h
@@ -0,0 +1,1594 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2009, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ * Joshua Colp <jcolp@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ * \brief Pluggable RTP Architecture
+ * \author Joshua Colp <jcolp@digium.com>
+ * \ref AstRTPEngine
+ */
+
+/*!
+ * \page AstRTPEngine Asterisk RTP Engine API
+ *
+ * The purpose of this API is to provide a way for multiple RTP stacks to be used inside
+ * of Asterisk without any module that uses RTP knowing any different. To the module each RTP
+ * stack behaves the same.
+ *
+ * An RTP session is called an instance and is made up of a combination of codec information,
+ * RTP engine, RTP properties, and address information. An engine name may be passed in to explicitly
+ * choose an RTP stack to be used but a default one will be used if none is provided. An address to use
+ * for RTP may also be provided but the underlying RTP engine may choose a different address depending on
+ * it's configuration.
+ *
+ * An RTP engine is the layer between the RTP engine core and the RTP stack itself. The RTP engine core provides
+ * a set of callbacks to do various things (such as write audio out) that the RTP engine has to have implemented.
+ *
+ * Glue is what binds an RTP instance to a channel. It is used to retrieve RTP instance information when
+ * performing remote or local bridging and is used to have the channel driver tell the remote side to change
+ * destination of the RTP stream.
+ *
+ * Statistics from an RTP instance can be retrieved using the ast_rtp_instance_get_stats API call. This essentially
+ * asks the RTP engine in use to fill in a structure with the requested values. It is not required for an RTP engine
+ * to support all statistic values.
+ *
+ * Properties allow behavior of the RTP engine and RTP engine core to be changed. For example, there is a property named
+ * AST_RTP_PROPERTY_NAT which is used to tell the RTP engine to enable symmetric RTP if it supports it. It is not required
+ * for an RTP engine to support all properties.
+ *
+ * Codec information is stored using a separate data structure which has it's own set of API calls to add/remove/retrieve
+ * information. They are used by the module after an RTP instance is created so that payload information is available for
+ * the RTP engine.
+ */
+
+#ifndef _ASTERISK_RTP_ENGINE_H
+#define _ASTERISK_RTP_ENGINE_H
+
+#if defined(__cplusplus) || defined(c_plusplus)
+extern "C" {
+#endif
+
+#include "asterisk/astobj2.h"
+
+/* Maximum number of payloads supported */
+#define AST_RTP_MAX_PT 256
+
+/* Maximum number of generations */
+#define AST_RED_MAX_GENERATION 5
+
+struct ast_rtp_instance;
+struct ast_rtp_glue;
+
+/*! RTP Properties that can be set on an RTP instance */
+enum ast_rtp_property {
+ /*! Enable symmetric RTP support */
+ AST_RTP_PROPERTY_NAT = 0,
+ /*! RTP instance will be carrying DTMF (using RFC2833) */
+ AST_RTP_PROPERTY_DTMF,
+ /*! Expect unreliable DTMF from remote party */
+ AST_RTP_PROPERTY_DTMF_COMPENSATE,
+ /*! Enable STUN support */
+ AST_RTP_PROPERTY_STUN,
+ /*! Enable RTCP support */
+ AST_RTP_PROPERTY_RTCP,
+ /*! Maximum number of RTP properties supported */
+ AST_RTP_PROPERTY_MAX,
+};
+
+/*! Additional RTP options */
+enum ast_rtp_options {
+ /*! Remote side is using non-standard G.726 */
+ AST_RTP_OPT_G726_NONSTANDARD = (1 << 0),
+};
+
+/*! RTP DTMF Modes */
+enum ast_rtp_dtmf_mode {
+ /*! No DTMF is being carried over the RTP stream */
+ AST_RTP_DTMF_MODE_NONE = 0,
+ /*! DTMF is being carried out of band using RFC2833 */
+ AST_RTP_DTMF_MODE_RFC2833,
+ /*! DTMF is being carried inband over the RTP stream */
+ AST_RTP_DTMF_MODE_INBAND,
+};
+
+/*! Result codes when RTP glue is queried for information */
+enum ast_rtp_glue_result {
+ /*! No remote or local bridging is permitted */
+ AST_RTP_GLUE_RESULT_FORBID = 0,
+ /*! Move RTP stream to be remote between devices directly */
+ AST_RTP_GLUE_RESULT_REMOTE,
+ /*! Perform RTP engine level bridging if possible */
+ AST_RTP_GLUE_RESULT_LOCAL,
+};
+
+/*! Field statistics that can be retrieved from an RTP instance */
+enum ast_rtp_instance_stat_field {
+ /*! Retrieve quality information */
+ AST_RTP_INSTANCE_STAT_FIELD_QUALITY = 0,
+ /*! Retrieve quality information about jitter */
+ AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER,
+ /*! Retrieve quality information about packet loss */
+ AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS,
+ /*! Retrieve quality information about round trip time */
+ AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT,
+};
+
+/*! Statistics that can be retrieved from an RTP instance */
+enum ast_rtp_instance_stat {
+ /*! Retrieve all statistics */
+ AST_RTP_INSTANCE_STAT_ALL = 0,
+ /*! Retrieve number of packets transmitted */
+ AST_RTP_INSTANCE_STAT_TXCOUNT,
+ /*! Retrieve number of packets received */
+ AST_RTP_INSTANCE_STAT_RXCOUNT,
+ /*! Retrieve ALL statistics relating to packet loss */
+ AST_RTP_INSTANCE_STAT_COMBINED_LOSS,
+ /*! Retrieve number of packets lost for transmitting */
+ AST_RTP_INSTANCE_STAT_TXPLOSS,
+ /*! Retrieve number of packets lost for receiving */
+ AST_RTP_INSTANCE_STAT_RXPLOSS,
+ /*! Retrieve maximum number of packets lost on remote side */
+ AST_RTP_INSTANCE_STAT_REMOTE_MAXRXPLOSS,
+ /*! Retrieve minimum number of packets lost on remote side */
+ AST_RTP_INSTANCE_STAT_REMOTE_MINRXPLOSS,
+ /*! Retrieve average number of packets lost on remote side */
+ AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVRXPLOSS,
+ /*! Retrieve standard deviation of packets lost on remote side */
+ AST_RTP_INSTANCE_STAT_REMOTE_STDEVRXPLOSS,
+ /*! Retrieve maximum number of packets lost on local side */
+ AST_RTP_INSTANCE_STAT_LOCAL_MAXRXPLOSS,
+ /*! Retrieve minimum number of packets lost on local side */
+ AST_RTP_INSTANCE_STAT_LOCAL_MINRXPLOSS,
+ /*! Retrieve average number of packets lost on local side */
+ AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVRXPLOSS,
+ /*! Retrieve standard deviation of packets lost on local side */
+ AST_RTP_INSTANCE_STAT_LOCAL_STDEVRXPLOSS,
+ /*! Retrieve ALL statistics relating to jitter */
+ AST_RTP_INSTANCE_STAT_COMBINED_JITTER,
+ /*! Retrieve jitter on transmitted packets */
+ AST_RTP_INSTANCE_STAT_TXJITTER,
+ /*! Retrieve jitter on received packets */
+ AST_RTP_INSTANCE_STAT_RXJITTER,
+ /*! Retrieve maximum jitter on remote side */
+ AST_RTP_INSTANCE_STAT_REMOTE_MAXJITTER,
+ /*! Retrieve minimum jitter on remote side */
+ AST_RTP_INSTANCE_STAT_REMOTE_MINJITTER,
+ /*! Retrieve average jitter on remote side */
+ AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVJITTER,
+ /*! Retrieve standard deviation jitter on remote side */
+ AST_RTP_INSTANCE_STAT_REMOTE_STDEVJITTER,
+ /*! Retrieve maximum jitter on local side */
+ AST_RTP_INSTANCE_STAT_LOCAL_MAXJITTER,
+ /*! Retrieve minimum jitter on local side */
+ AST_RTP_INSTANCE_STAT_LOCAL_MINJITTER,
+ /*! Retrieve average jitter on local side */
+ AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVJITTER,
+ /*! Retrieve standard deviation jitter on local side */
+ AST_RTP_INSTANCE_STAT_LOCAL_STDEVJITTER,
+ /*! Retrieve ALL statistics relating to round trip time */
+ AST_RTP_INSTANCE_STAT_COMBINED_RTT,
+ /*! Retrieve round trip time */
+ AST_RTP_INSTANCE_STAT_RTT,
+ /*! Retrieve maximum round trip time */
+ AST_RTP_INSTANCE_STAT_MAX_RTT,
+ /*! Retrieve minimum round trip time */
+ AST_RTP_INSTANCE_STAT_MIN_RTT,
+ /*! Retrieve average round trip time */
+ AST_RTP_INSTANCE_STAT_NORMDEVRTT,
+ /*! Retrieve standard deviation round trip time */
+ AST_RTP_INSTANCE_STAT_STDEVRTT,
+ /*! Retrieve local SSRC */
+ AST_RTP_INSTANCE_STAT_LOCAL_SSRC,
+ /*! Retrieve remote SSRC */
+ AST_RTP_INSTANCE_STAT_REMOTE_SSRC,
+};
+
+/* Codes for RTP-specific data - not defined by our AST_FORMAT codes */
+/*! DTMF (RFC2833) */
+#define AST_RTP_DTMF (1 << 0)
+/*! 'Comfort Noise' (RFC3389) */
+#define AST_RTP_CN (1 << 1)
+/*! DTMF (Cisco Proprietary) */
+#define AST_RTP_CISCO_DTMF (1 << 2)
+/*! Maximum RTP-specific code */
+#define AST_RTP_MAX AST_RTP_CISCO_DTMF
+
+/*! Structure that represents a payload */
+struct ast_rtp_payload_type {
+ /*! Is this an Asterisk value */
+ int asterisk_format;
+ /*! Actual internal value of the payload */
+ int code;
+};
+
+/*! Structure that represents statistics from an RTP instance */
+struct ast_rtp_instance_stats {
+ /*! Number of packets transmitted */
+ unsigned int txcount;
+ /*! Number of packets received */
+ unsigned int rxcount;
+ /*! Jitter on transmitted packets */
+ unsigned int txjitter;
+ /*! Jitter on received packets */
+ unsigned int rxjitter;
+ /*! Maximum jitter on remote side */
+ double remote_maxjitter;
+ /*! Minimum jitter on remote side */
+ double remote_minjitter;
+ /*! Average jitter on remote side */
+ double remote_normdevjitter;
+ /*! Standard deviation jitter on remote side */
+ double remote_stdevjitter;
+ /*! Maximum jitter on local side */
+ double local_maxjitter;
+ /*! Minimum jitter on local side */
+ double local_minjitter;
+ /*! Average jitter on local side */
+ double local_normdevjitter;
+ /*! Standard deviation jitter on local side */
+ double local_stdevjitter;
+ /*! Number of transmitted packets lost */
+ unsigned int txploss;
+ /*! Number of received packets lost */
+ unsigned int rxploss;
+ /*! Maximum number of packets lost on remote side */
+ double remote_maxrxploss;
+ /*! Minimum number of packets lost on remote side */
+ double remote_minrxploss;
+ /*! Average number of packets lost on remote side */
+ double remote_normdevrxploss;
+ /*! Standard deviation packets lost on remote side */
+ double remote_stdevrxploss;
+ /*! Maximum number of packets lost on local side */
+ double local_maxrxploss;
+ /*! Minimum number of packets lost on local side */
+ double local_minrxploss;
+ /*! Average number of packets lost on local side */
+ double local_normdevrxploss;
+ /*! Standard deviation packets lost on local side */
+ double local_stdevrxploss;
+ /*! Total round trip time */
+ unsigned int rtt;
+ /*! Maximum round trip time */
+ double maxrtt;
+ /*! Minimum round trip time */
+ double minrtt;
+ /*! Average round trip time */
+ double normdevrtt;
+ /*! Standard deviation round trip time */
+ double stdevrtt;
+ /*! Our SSRC */
+ unsigned int local_ssrc;
+ /*! Their SSRC */
+ unsigned int remote_ssrc;
+};
+
+#define AST_RTP_STAT_SET(current_stat, combined, placement, value) \
+if (stat == current_stat || stat == AST_RTP_INSTANCE_STAT_ALL || (combined >= 0 && combined == current_stat)) { \
+placement = value; \
+if (stat == current_stat) { \
+return 0; \
+} \
+}
+
+#define AST_RTP_STAT_TERMINATOR(combined) \
+if (stat == combined) { \
+return 0; \
+}
+
+/*! Structure that represents an RTP stack (engine) */
+struct ast_rtp_engine {
+ /*! Name of the RTP engine, used when explicitly requested */
+ const char *name;
+ /*! Module this RTP engine came from, used for reference counting */
+ struct ast_module *mod;
+ /*! Callback for setting up a new RTP instance */
+ int (*new)(struct ast_rtp_instance *instance, struct sched_context *sched, struct sockaddr_in *sin, void *data);
+ /*! Callback for destroying an RTP instance */
+ int (*destroy)(struct ast_rtp_instance *instance);
+ /*! Callback for writing out a frame */
+ int (*write)(struct ast_rtp_instance *instance, struct ast_frame *frame);
+ /*! Callback for stopping the RTP instance */
+ void (*stop)(struct ast_rtp_instance *instance);
+ /*! Callback for starting RFC2833 DTMF transmission */
+ int (*dtmf_begin)(struct ast_rtp_instance *instance, char digit);
+ /*! Callback for stopping RFC2833 DTMF transmission */
+ int (*dtmf_end)(struct ast_rtp_instance *instance, char digit);
+ /*! Callback to indicate that a new source of media has come in */
+ void (*new_source)(struct ast_rtp_instance *instance);
+ /*! Callback for setting an extended RTP property */
+ int (*extended_prop_set)(struct ast_rtp_instance *instance, int property, void *value);
+ /*! Callback for getting an extended RTP property */
+ void *(*extended_prop_get)(struct ast_rtp_instance *instance, int property);
+ /*! Callback for setting an RTP property */
+ void (*prop_set)(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value);
+ /*! Callback for setting a payload */
+ void (*payload_set)(struct ast_rtp_instance *instance, int payload, int astformat, int format);
+ /*! Callback for setting packetization preferences */
+ void (*packetization_set)(struct ast_rtp_instance *instance, struct ast_codec_pref *pref);
+ /*! Callback for setting the remote address that RTP is to be sent to */
+ void (*remote_address_set)(struct ast_rtp_instance *instance, struct sockaddr_in *sin);
+ /*! Callback for changing DTMF mode */
+ int (*dtmf_mode_set)(struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode);
+ /*! Callback for retrieving statistics */
+ int (*get_stat)(struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat);
+ /*! Callback for setting QoS values */
+ int (*qos)(struct ast_rtp_instance *instance, int tos, int cos, const char *desc);
+ /*! Callback for retrieving a file descriptor to poll on, not always required */
+ int (*fd)(struct ast_rtp_instance *instance, int rtcp);
+ /*! Callback for initializing RED support */
+ int (*red_init)(struct ast_rtp_instance *instance, int buffer_time, int *payloads, int generations);
+ /*! Callback for buffering a frame using RED */
+ int (*red_buffer)(struct ast_rtp_instance *instance, struct ast_frame *frame);
+ /*! Callback for reading a frame from the RTP engine */
+ struct ast_frame *(*read)(struct ast_rtp_instance *instance, int rtcp);
+ /*! Callback to locally bridge two RTP instances */
+ int (*local_bridge)(struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1);
+ /*! Callback to set the read format */
+ int (*set_read_format)(struct ast_rtp_instance *instance, int format);
+ /*! Callback to set the write format */
+ int (*set_write_format)(struct ast_rtp_instance *instance, int format);
+ /*! Callback to make two instances compatible */
+ int (*make_compatible)(struct ast_channel *chan0, struct ast_rtp_instance *instance0, struct ast_channel *chan1, struct ast_rtp_instance *instance1);
+ /*! Callback to see if two instances are compatible with DTMF */
+ int (*dtmf_compatible)(struct ast_channel *chan0, struct ast_rtp_instance *instance0, struct ast_channel *chan1, struct ast_rtp_instance *instance1);
+ /*! Callback to indicate that packets will now flow */
+ int (*activate)(struct ast_rtp_instance *instance);
+ /*! Callback to request that the RTP engine send a STUN BIND request */
+ void (*stun_request)(struct ast_rtp_instance *instance, struct sockaddr_in *suggestion, const char *username);
+ /*! Linked list information */
+ AST_RWLIST_ENTRY(ast_rtp_engine) entry;
+};
+
+/*! Structure that represents codec and packetization information */
+struct ast_rtp_codecs {
+ /*! Codec packetization preferences */
+ struct ast_codec_pref pref;
+ /*! Payloads present */
+ struct ast_rtp_payload_type payloads[AST_RTP_MAX_PT];
+};
+
+/*! Structure that represents the glue that binds an RTP instance to a channel */
+struct ast_rtp_glue {
+ /*! Name of the channel driver that this glue is responsible for */
+ const char *type;
+ /*! Module that the RTP glue came from */
+ struct ast_module *mod;
+ /*!
+ * \brief Callback for retrieving the RTP instance carrying audio
+ * \note This function increases the reference count on the returned RTP instance.
+ */
+ enum ast_rtp_glue_result (*get_rtp_info)(struct ast_channel *chan, struct ast_rtp_instance **instance);
+ /*!
+ * \brief Callback for retrieving the RTP instance carrying video
+ * \note This function increases the reference count on the returned RTP instance.
+ */
+ enum ast_rtp_glue_result (*get_vrtp_info)(struct ast_channel *chan, struct ast_rtp_instance **instance);
+ /*!
+ * \brief Callback for retrieving the RTP instance carrying text
+ * \note This function increases the reference count on the returned RTP instance.
+ */
+ enum ast_rtp_glue_result (*get_trtp_info)(struct ast_channel *chan, struct ast_rtp_instance **instance);
+ /*! Callback for updating the destination that the remote side should send RTP to */
+ int (*update_peer)(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, int codecs, int nat_active);
+ /*! Callback for retrieving codecs that the channel can do */
+ int (*get_codec)(struct ast_channel *chan);
+ /*! Linked list information */
+ AST_RWLIST_ENTRY(ast_rtp_glue) entry;
+};
+
+#define ast_rtp_engine_register(engine) ast_rtp_engine_register2(engine, ast_module_info->self)
+
+/*!
+ * \brief Register an RTP engine
+ *
+ * \param engine Structure of the RTP engine to register
+ * \param module Module that the RTP engine is part of
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_engine_register2(&example_rtp_engine, NULL);
+ * \endcode
+ *
+ * This registers the RTP engine declared as example_rtp_engine with the RTP engine core, but does not
+ * associate a module with it.
+ *
+ * \note It is recommended that you use the ast_rtp_engine_register macro so that the module is
+ * associated with the RTP engine and use counting is performed.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_engine_register2(struct ast_rtp_engine *engine, struct ast_module *module);
+
+/*!
+ * \brief Unregister an RTP engine
+ *
+ * \param engine Structure of the RTP engine to unregister
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_engine_unregister(&example_rtp_engine);
+ * \endcode
+ *
+ * This unregisters the RTP engine declared as example_rtp_engine from the RTP engine core. If a module
+ * reference was provided when it was registered then this will only be called once the RTP engine is no longer in use.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_engine_unregister(struct ast_rtp_engine *engine);
+
+#define ast_rtp_glue_register(glue) ast_rtp_glue_register2(glue, ast_module_info->self)
+
+/*!
+ * \brief Register RTP glue
+ *
+ * \param glue The glue to register
+ * \param module Module that the RTP glue is part of
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_glue_register2(&example_rtp_glue, NULL);
+ * \endcode
+ *
+ * This registers the RTP glue declared as example_rtp_glue with the RTP engine core, but does not
+ * associate a module with it.
+ *
+ * \note It is recommended that you use the ast_rtp_glue_register macro so that the module is
+ * associated with the RTP glue and use counting is performed.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_glue_register2(struct ast_rtp_glue *glue, struct ast_module *module);
+
+/*!
+ * \brief Unregister RTP glue
+ *
+ * \param glue The glue to unregister
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_glue_unregister(&example_rtp_glue);
+ * \endcode
+ *
+ * This unregisters the RTP glue declared as example_rtp_gkue from the RTP engine core. If a module
+ * reference was provided when it was registered then this will only be called once the RTP engine is no longer in use.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_glue_unregister(struct ast_rtp_glue *glue);
+
+/*!
+ * \brief Create a new RTP instance
+ *
+ * \param engine_name Name of the engine to use for the RTP instance
+ * \param sched Scheduler context that the RTP engine may want to use
+ * \param sin Address we want to bind to
+ * \param data Unique data for the engine
+ *
+ * \retval non-NULL success
+ * \retval NULL failure
+ *
+ * Example usage:
+ *
+ * \code
+ * struct ast_rtp_instance *instance = NULL;
+ * instance = ast_rtp_instance_new(NULL, sched, &sin, NULL);
+ * \endcode
+ *
+ * This creates a new RTP instance using the default engine and asks the RTP engine to bind to the address given
+ * in the sin structure.
+ *
+ * \note The RTP engine does not have to use the address provided when creating an RTP instance. It may choose to use
+ * another depending on it's own configuration.
+ *
+ * \since 1.6.3
+ */
+struct ast_rtp_instance *ast_rtp_instance_new(const char *engine_name, struct sched_context *sched, struct sockaddr_in *sin, void *data);
+
+/*!
+ * \brief Destroy an RTP instance
+ *
+ * \param instance The RTP instance to destroy
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_destroy(instance);
+ * \endcode
+ *
+ * This destroys the RTP instance pointed to by instance. Once this function returns instance no longer points to valid
+ * memory and may not be used again.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_destroy(struct ast_rtp_instance *instance);
+
+/*!
+ * \brief Set the data portion of an RTP instance
+ *
+ * \param instance The RTP instance to manipulate
+ * \param data Pointer to data
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_set_data(instance, blob);
+ * \endcode
+ *
+ * This sets the data pointer on the RTP instance pointed to by 'instance' to
+ * blob.
+ *
+ * \since 1.6.3
+ */
+void ast_rtp_instance_set_data(struct ast_rtp_instance *instance, void *data);
+
+/*!
+ * \brief Get the data portion of an RTP instance
+ *
+ * \param instance The RTP instance we want the data portion from
+ *
+ * Example usage:
+ *
+ * \code
+ * struct *blob = ast_rtp_instance_get_data(instance);
+ ( \endcode
+ *
+ * This gets the data pointer on the RTP instance pointed to by 'instance'.
+ *
+ * \since 1.6.3
+ */
+void *ast_rtp_instance_get_data(struct ast_rtp_instance *instance);
+
+/*!
+ * \brief Send a frame out over RTP
+ *
+ * \param instance The RTP instance to send frame out on
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_write(instance, frame);
+ * \endcode
+ *
+ * This gives the frame pointed to by frame to the RTP engine being used for the instance
+ * and asks that it be transmitted to the current remote address set on the RTP instance.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_write(struct ast_rtp_instance *instance, struct ast_frame *frame);
+
+/*!
+ * \brief Receive a frame over RTP
+ *
+ * \param instance The RTP instance to receive frame on
+ * \param rtcp Whether to read in RTCP or not
+ *
+ * \retval non-NULL success
+ * \retval NULL failure
+ *
+ * Example usage:
+ *
+ * \code
+ * struct ast_frame *frame;
+ * frame = ast_rtp_instance_read(instance, 0);
+ * \endcode
+ *
+ * This asks the RTP engine to read in RTP from the instance and return it as an Asterisk frame.
+ *
+ * \since 1.6.3
+ */
+struct ast_frame *ast_rtp_instance_read(struct ast_rtp_instance *instance, int rtcp);
+
+/*!
+ * \brief Set the address of the remote endpoint that we are sending RTP to
+ *
+ * \param instance The RTP instance to change the address on
+ * \param address Address to set it to
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_set_remote_address(instance, &sin);
+ * \endcode
+ *
+ * This changes the remote address that RTP will be sent to on instance to the address given in the sin
+ * structure.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_set_remote_address(struct ast_rtp_instance *instance, struct sockaddr_in *address);
+
+/*!
+ * \brief Set the address that we are expecting to receive RTP on
+ *
+ * \param instance The RTP instance to change the address on
+ * \param address Address to set it to
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_set_local_address(instance, &sin);
+ * \endcode
+ *
+ * This changes the local address that RTP is expected on to the address given in the sin
+ * structure.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_set_local_address(struct ast_rtp_instance *instance, struct sockaddr_in *address);
+
+/*!
+ * \brief Get the local address that we are expecting RTP on
+ *
+ * \param instance The RTP instance to get the address from
+ * \param address The variable to store the address in
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * struct sockaddr_in sin;
+ * ast_rtp_instance_get_local_address(instance, &sin);
+ * \endcode
+ *
+ * This gets the local address that we are expecting RTP on and stores it in the 'sin' structure.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_get_local_address(struct ast_rtp_instance *instance, struct sockaddr_in *address);
+
+/*!
+ * \brief Get the address of the remote endpoint that we are sending RTP to
+ *
+ * \param instance The instance that we want to get the remote address for
+ * \param address A structure to put the address into
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * struct sockaddr_in sin;
+ * ast_rtp_instance_get_remote_address(instance, &sin);
+ * \endcode
+ *
+ * This retrieves the current remote address set on the instance pointed to by instance and puts the value
+ * into the sin structure.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_get_remote_address(struct ast_rtp_instance *instance, struct sockaddr_in *address);
+
+/*!
+ * \brief Set the value of an RTP instance extended property
+ *
+ * \param instance The RTP instance to set the extended property on
+ * \param property The extended property to set
+ * \param value The value to set the extended property to
+ *
+ * \since 1.6.3
+ */
+void ast_rtp_instance_set_extended_prop(struct ast_rtp_instance *instance, int property, void *value);
+
+/*!
+ * \brief Get the value of an RTP instance extended property
+ *
+ * \param instance The RTP instance to get the extended property on
+ * \param property The extended property to get
+ *
+ * \since 1.6.3
+ */
+void *ast_rtp_instance_get_extended_prop(struct ast_rtp_instance *instance, int property);
+
+/*!
+ * \brief Set the value of an RTP instance property
+ *
+ * \param instance The RTP instance to set the property on
+ * \param property The property to modify
+ * \param value The value to set the property to
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_set_prop(instance, AST_RTP_PROPERTY_NAT, 1);
+ * \endcode
+ *
+ * This enables the AST_RTP_PROPERTY_NAT property on the instance pointed to by instance.
+ *
+ * \since 1.6.3
+ */
+void ast_rtp_instance_set_prop(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value);
+
+/*!
+ * \brief Get the value of an RTP instance property
+ *
+ * \param instance The RTP instance to get the property from
+ * \param property The property to get
+ *
+ * \retval Current value of the property
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT);
+ * \endcode
+ *
+ * This returns the current value of the NAT property on the instance pointed to by instance.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_get_prop(struct ast_rtp_instance *instance, enum ast_rtp_property property);
+
+/*!
+ * \brief Get the codecs structure of an RTP instance
+ *
+ * \param instance The RTP instance to get the codecs structure from
+ *
+ * Example usage:
+ *
+ * \code
+ * struct ast_rtp_codecs *codecs = ast_rtp_instance_get_codecs(instance);
+ * \endcode
+ *
+ * This gets the codecs structure on the RTP instance pointed to by 'instance'.
+ *
+ * \since 1.6.3
+ */
+struct ast_rtp_codecs *ast_rtp_instance_get_codecs(struct ast_rtp_instance *instance);
+
+/*!
+ * \brief Clear payload information from an RTP instance
+ *
+ * \param codecs The codecs structure that payloads will be cleared from
+ * \param instance Optionally the instance that the codecs structure belongs to
+ *
+ * Example usage:
+ *
+ * \code
+ * struct ast_rtp_codecs codecs;
+ * ast_rtp_codecs_payloads_clear(&codecs, NULL);
+ * \endcode
+ *
+ * This clears the codecs structure and puts it into a pristine state.
+ *
+ * \since 1.6.3
+ */
+void ast_rtp_codecs_payloads_clear(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance);
+
+/*!
+ * \brief Set payload information on an RTP instance to the default
+ *
+ * \param codecs The codecs structure to set defaults on
+ * \param instance Optionally the instance that the codecs structure belongs to
+ *
+ * Example usage:
+ *
+ * \code
+ * struct ast_rtp_codecs codecs;
+ * ast_rtp_codecs_payloads_default(&codecs, NULL);
+ * \endcode
+ *
+ * This sets the default payloads on the codecs structure.
+ *
+ * \since 1.6.3
+ */
+void ast_rtp_codecs_payloads_default(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance);
+
+/*!
+ * \brief Copy payload information from one RTP instance to another
+ *
+ * \param src The source codecs structure
+ * \param dst The destination codecs structure that the values from src will be copied to
+ * \param instance Optionally the instance that the dst codecs structure belongs to
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_codecs_payloads_copy(&codecs0, &codecs1, NULL);
+ * \endcode
+ *
+ * This copies the payloads from the codecs0 structure to the codecs1 structure, overwriting any current values.
+ *
+ * \since 1.6.3
+ */
+void ast_rtp_codecs_payloads_copy(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance);
+
+/*!
+ * \brief Record payload information that was seen in an m= SDP line
+ *
+ * \param codecs The codecs structure to muck with
+ * \param instance Optionally the instance that the codecs structure belongs to
+ * \param payload Numerical payload that was seen in the m= SDP line
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_codecs_payloads_set_m_type(&codecs, NULL, 0);
+ * \endcode
+ *
+ * This records that the numerical payload '0' was seen in the codecs structure.
+ *
+ * \since 1.6.3
+ */
+void ast_rtp_codecs_payloads_set_m_type(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload);
+
+/*!
+ * \brief Record payload information that was seen in an a=rtpmap: SDP line
+ *
+ * \param codecs The codecs structure to muck with
+ * \param instance Optionally the instance that the codecs structure belongs to
+ * \param payload Numerical payload that was seen in the a=rtpmap: SDP line
+ * \param mimetype The string mime type that was seen
+ * \param mimesubtype The strin mime sub type that was seen
+ * \param options Optional options that may change the behavior of this specific payload
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_codecs_payloads_set_rtpmap_type(&codecs, NULL, 0, "audio", "PCMU", 0);
+ * \endcode
+ *
+ * This records that the numerical payload '0' was seen with mime type 'audio' and sub mime type 'PCMU' in the codecs structure.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_codecs_payloads_set_rtpmap_type(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload, char *mimetype, char *mimesubtype, enum ast_rtp_options options);
+
+/*!
+ * \brief Set payload type to a known MIME media type for a codec with a specific sample rate
+ *
+ * \param rtp RTP structure to modify
+ * \param instance Optionally the instance that the codecs structure belongs to
+ * \param pt Payload type entry to modify
+ * \param mimetype top-level MIME type of media stream (typically "audio", "video", "text", etc.)
+ * \param mimesubtype MIME subtype of media stream (typically a codec name)
+ * \param options Zero or more flags from the ast_rtp_options enum
+ * \param sample_rate The sample rate of the media stream
+ *
+ * This function 'fills in' an entry in the list of possible formats for
+ * a media stream associated with an RTP structure.
+ *
+ * \retval 0 on success
+ * \retval -1 if the payload type is out of range
+ * \retval -2 if the mimeType/mimeSubtype combination was not found
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int pt,
+ char *mimetype, char *mimesubtype,
+ enum ast_rtp_options options,
+ unsigned int sample_rate);
+
+/*!
+ * \brief Remove payload information
+ *
+ * \param codecs The codecs structure to muck with
+ * \param instance Optionally the instance that the codecs structure belongs to
+ * \param payload Numerical payload to unset
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_codecs_payloads_unset(&codecs, NULL, 0);
+ * \endcode
+ *
+ * This clears the payload '0' from the codecs structure. It will be as if it was never set.
+ *
+ * \since 1.6.3
+ */
+void ast_rtp_codecs_payloads_unset(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload);
+
+/*!
+ * \brief Retrieve payload information by payload
+ *
+ * \param codecs Codecs structure to look in
+ * \param payload Numerical payload to look up
+ *
+ * \retval Payload information
+ *
+ * Example usage:
+ *
+ * \code
+ * struct ast_rtp_payload_type payload_type;
+ * payload_type = ast_rtp_codecs_payload_lookup(&codecs, 0);
+ * \endcode
+ *
+ * This looks up the information for payload '0' from the codecs structure.
+ *
+ * \since 1.6.3
+ */
+struct ast_rtp_payload_type ast_rtp_codecs_payload_lookup(struct ast_rtp_codecs *codecs, int payload);
+
+/*!
+ * \brief Get the sample rate associated with known RTP payload types
+ *
+ * \param asterisk_format True if the value in the 'code' parameter is an AST_FORMAT value
+ * \param code Format code, either from AST_FORMAT list or from AST_RTP list
+ *
+ * \return the sample rate if the format was found, zero if it was not found
+ *
+ * \since 1.6.3
+ */
+unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, int code);
+
+/*!
+ * \brief Retrieve all formats that were found
+ *
+ * \param codecs Codecs structure to look in
+ * \param astFormats An integer to put the Asterisk formats in
+ * \param nonastformats An integer to put the non-Asterisk formats in
+ *
+ * Example usage:
+ *
+ * \code
+ * int astformats, nonastformats;
+ * ast_rtp_codecs_payload_Formats(&codecs, &astformats, &nonastformats);
+ * \endcode
+ *
+ * This retrieves all the formats known about in the codecs structure and puts the Asterisk ones in the integer
+ * pointed to by astformats and the non-Asterisk ones in the integer pointed to by nonastformats.
+ *
+ * \since 1.6.3
+ */
+void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, int *astformats, int *nonastformats);
+
+/*!
+ * \brief Retrieve a payload based on whether it is an Asterisk format and the code
+ *
+ * \param codecs Codecs structure to look in
+ * \param asterisk_format Non-zero if the given code is an Asterisk format value
+ * \param code The format to look for
+ *
+ * \retval Numerical payload
+ *
+ * Example usage:
+ *
+ * \code
+ * int payload = ast_rtp_codecs_payload_code(&codecs, 1, AST_FORMAT_ULAW);
+ * \endcode
+ *
+ * This looks for the numerical payload for ULAW in the codecs structure.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, const int asterisk_format, const int code);
+
+/*!
+ * \brief Retrieve mime subtype information on a payload
+ *
+ * \param asterisk_format Non-zero if the given code is an Asterisk format value
+ * \param code Format to look up
+ * \param options Additional options that may change the result
+ *
+ * \retval Mime subtype success
+ * \retval NULL failure
+ *
+ * Example usage:
+ *
+ * \code
+ * const char *subtype = ast_rtp_lookup_mime_subtype2(1, AST_FORMAT_ULAW, 0);
+ * \endcode
+ *
+ * This looks up the mime subtype for the ULAW format.
+ *
+ * \since 1.6.3
+ */
+const char *ast_rtp_lookup_mime_subtype2(const int asterisk_format, const int code, enum ast_rtp_options options);
+
+/*!
+ * \brief Convert formats into a string and put them into a buffer
+ *
+ * \param buf Buffer to put the mime output into
+ * \param capability Formats that we are looking up
+ * \param asterisk_format Non-zero if the given capability are Asterisk format capabilities
+ * \param options Additional options that may change the result
+ *
+ * \retval non-NULL success
+ * \retval NULL failure
+ *
+ * Example usage:
+ *
+ * \code
+ * char buf[256] = "";
+ * char *mime = ast_rtp_lookup_mime_multiple2(&buf, sizeof(buf), AST_FORMAT_ULAW | AST_FORMAT_ALAW, 1, 0);
+ * \endcode
+ *
+ * This returns the mime values for ULAW and ALAW in the buffer pointed to by buf.
+ *
+ * \since 1.6.3
+ */
+char *ast_rtp_lookup_mime_multiple2(struct ast_str *buf, const int capability, const int asterisk_format, enum ast_rtp_options options);
+
+/*!
+ * \brief Set codec packetization preferences
+ *
+ * \param codecs Codecs structure to muck with
+ * \param instance Optionally the instance that the codecs structure belongs to
+ * \param prefs Codec packetization preferences
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_codecs_packetization_set(&codecs, NULL, &prefs);
+ * \endcode
+ *
+ * This sets the packetization preferences pointed to by prefs on the codecs structure pointed to by codecs.
+ *
+ * \since 1.6.3
+ */
+void ast_rtp_codecs_packetization_set(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, struct ast_codec_pref *prefs);
+
+/*!
+ * \brief Begin sending a DTMF digit
+ *
+ * \param instance The RTP instance to send the DTMF on
+ * \param digit What DTMF digit to send
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_dtmf_begin(instance, '1');
+ * \endcode
+ *
+ * This starts sending the DTMF '1' on the RTP instance pointed to by instance. It will
+ * continue being sent until it is ended.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_dtmf_begin(struct ast_rtp_instance *instance, char digit);
+
+/*!
+ * \brief Stop sending a DTMF digit
+ *
+ * \param instance The RTP instance to stop the DTMF on
+ * \param digit What DTMF digit to stop
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_dtmf_end(instance, '1');
+ * \endcode
+ *
+ * This stops sending the DTMF '1' on the RTP instance pointed to by instance.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_dtmf_end(struct ast_rtp_instance *instance, char digit);
+
+/*!
+ * \brief Set the DTMF mode that should be used
+ *
+ * \param instance the RTP instance to set DTMF mode on
+ * \param dtmf_mode The DTMF mode that is in use
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_dtmf_mode_set(instance, AST_RTP_DTMF_MODE_RFC2833);
+ * \endcode
+ *
+ * This sets the RTP instance to use RFC2833 for DTMF transmission and receiving.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode);
+
+/*!
+ * \brief Get the DTMF mode of an RTP instance
+ *
+ * \param instance The RTP instance to get the DTMF mode of
+ *
+ * \retval DTMF mode
+ *
+ * Example usage:
+ *
+ * \code
+ * enum ast_rtp_dtmf_mode dtmf_mode = ast_rtp_instance_dtmf_mode_get(instance);
+ * \endcode
+ *
+ * This gets the DTMF mode set on the RTP instance pointed to by 'instance'.
+ *
+ * \since 1.6.3
+ */
+enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *instance);
+
+/*!
+ * \brief Indicate a new source of audio has dropped in
+ *
+ * \param instance Instance that the new media source is feeding into
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_new_source(instance);
+ * \endcode
+ *
+ * This indicates that a new source of media is feeding the instance pointed to by
+ * instance.
+ *
+ * \since 1.6.3
+ */
+void ast_rtp_instance_new_source(struct ast_rtp_instance *instance);
+
+/*!
+ * \brief Set QoS parameters on an RTP session
+ *
+ * \param instance Instance to set the QoS parameters on
+ * \param tos Terms of service value
+ * \param cos Class of service value
+ * \param desc What is setting the QoS values
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_set_qos(instance, 0, 0, "Example");
+ * \endcode
+ *
+ * This sets the TOS and COS values to 0 on the instance pointed to by instance.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_set_qos(struct ast_rtp_instance *instance, int tos, int cos, const char *desc);
+
+/*!
+ * \brief Stop an RTP instance
+ *
+ * \param instance Instance that media is no longer going to at this time
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_stop(instance);
+ * \endcode
+ *
+ * This tells the RTP engine being used for the instance pointed to by instance
+ * that media is no longer going to it at this time, but may in the future.
+ *
+ * \since 1.6.3
+ */
+void ast_rtp_instance_stop(struct ast_rtp_instance *instance);
+
+/*!
+ * \brief Get the file descriptor for an RTP session (or RTCP)
+ *
+ * \param instance Instance to get the file descriptor for
+ * \param rtcp Whether to retrieve the file descriptor for RTCP or not
+ *
+ * \retval fd success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * int rtp_fd = ast_rtp_instance_fd(instance, 0);
+ * \endcode
+ *
+ * This retrieves the file descriptor for the socket carrying media on the instance
+ * pointed to by instance.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_fd(struct ast_rtp_instance *instance, int rtcp);
+
+/*!
+ * \brief Get the RTP glue that binds a channel to the RTP engine
+ *
+ * \param type Name of the glue we want
+ *
+ * \retval non-NULL success
+ * \retval NULL failure
+ *
+ * Example usage:
+ *
+ * \code
+ * struct ast_rtp_glue *glue = ast_rtp_instance_get_glue("Example");
+ * \endcode
+ *
+ * This retrieves the RTP glue that has the name 'Example'.
+ *
+ * \since 1.6.3
+ */
+struct ast_rtp_glue *ast_rtp_instance_get_glue(const char *type);
+
+/*!
+ * \brief Bridge two channels that use RTP instances
+ *
+ * \param c0 First channel part of the bridge
+ * \param c1 Second channel part of the bridge
+ * \param flags Bridging flags
+ * \param fo If a frame needs to be passed up it is stored here
+ * \param rc Channel that passed the above frame up
+ * \param timeoutms How long the channels should be bridged for
+ *
+ * \retval Bridge result
+ *
+ * \note This should only be used by channel drivers in their technology declaration.
+ *
+ * \since 1.6.3
+ */
+enum ast_bridge_result ast_rtp_instance_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms);
+
+/*!
+ * \brief Get the other RTP instance that an instance is bridged to
+ *
+ * \param instance The RTP instance that we want
+ *
+ * \retval non-NULL success
+ * \retval NULL failure
+ *
+ * Example usage:
+ *
+ * \code
+ * struct ast_rtp_instance *bridged = ast_rtp_instance_get_bridged(instance0);
+ * \endcode
+ *
+ * This gets the RTP instance that instance0 is bridged to.
+ *
+ * \since 1.6.3
+ */
+struct ast_rtp_instance *ast_rtp_instance_get_bridged(struct ast_rtp_instance *instance);
+
+/*!
+ * \brief Make two channels compatible for early bridging
+ *
+ * \param c0 First channel part of the bridge
+ * \param c1 Second channel part of the bridge
+ *
+ * \since 1.6.3
+ */
+void ast_rtp_instance_early_bridge_make_compatible(struct ast_channel *c0, struct ast_channel *c1);
+
+/*!
+ * \brief Early bridge two channels that use RTP instances
+ *
+ * \param c0 First channel part of the bridge
+ * \param c1 Second channel part of the bridge
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * \note This should only be used by channel drivers in their technology declaration.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_early_bridge(struct ast_channel *c0, struct ast_channel *c1);
+
+/*!
+ * \brief Initialize RED support on an RTP instance
+ *
+ * \param instance The instance to initialize RED support on
+ * \param buffer_time How long to buffer before sending
+ * \param payloads Payload values
+ * \param generations Number of generations
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_red_init(struct ast_rtp_instance *instance, int buffer_time, int *payloads, int generations);
+
+/*!
+ * \brief Buffer a frame in an RTP instance for RED
+ *
+ * \param instance The instance to buffer the frame on
+ * \param frame Frame that we want to buffer
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_red_buffer(struct ast_rtp_instance *instance, struct ast_frame *frame);
+
+/*!
+ * \brief Retrieve statistics about an RTP instance
+ *
+ * \param instance Instance to get statistics on
+ * \param stats Structure to put results into
+ * \param stat What statistic(s) to retrieve
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * struct ast_rtp_instance_stats stats;
+ * ast_rtp_instance_get_stats(instance, &stats, AST_RTP_INSTANCE_STAT_ALL);
+ * \endcode
+ *
+ * This retrieves all statistics the underlying RTP engine supports and puts the values into the
+ * stats structure.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_get_stats(struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat);
+
+/*!
+ * \brief Set standard statistics from an RTP instance on a channel
+ *
+ * \param chan Channel to set the statistics on
+ * \param instance The RTP instance that statistics will be retrieved from
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_set_stats_vars(chan, rtp);
+ * \endcode
+ *
+ * This retrieves standard statistics from the RTP instance rtp and sets it on the channel pointed to
+ * by chan.
+ *
+ * \since 1.6.3
+ */
+void ast_rtp_instance_set_stats_vars(struct ast_channel *chan, struct ast_rtp_instance *instance);
+
+/*!
+ * \brief Retrieve quality statistics about an RTP instance
+ *
+ * \param instance Instance to get statistics on
+ * \param field What quality statistic to retrieve
+ * \param buf What buffer to put the result into
+ * \param size Size of the above buffer
+ *
+ * \retval non-NULL success
+ * \retval NULL failure
+ *
+ * Example usage:
+ *
+ * \code
+ * char quality[AST_MAX_USER_FIELD];
+ * ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, &buf, sizeof(buf));
+ * \endcode
+ *
+ * This retrieves general quality statistics and places a text representation into the buf pointed to by buf.
+ *
+ * \since 1.6.3
+ */
+char *ast_rtp_instance_get_quality(struct ast_rtp_instance *instance, enum ast_rtp_instance_stat_field field, char *buf, size_t size);
+
+/*!
+ * \brief Request that the underlying RTP engine provide audio frames in a specific format
+ *
+ * \param instance The RTP instance to change read format on
+ * \param format Format that frames are wanted in
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_set_read_format(instance, AST_FORMAT_ULAW);
+ * \endcode
+ *
+ * This requests that the RTP engine provide audio frames in the ULAW format.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_set_read_format(struct ast_rtp_instance *instance, int format);
+
+/*!
+ * \brief Tell underlying RTP engine that audio frames will be provided in a specific format
+ *
+ * \param instance The RTP instance to change write format on
+ * \param format Format that frames will be provided in
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_set_write_format(instance, AST_FORMAT_ULAW);
+ * \endcode
+ *
+ * This tells the underlying RTP engine that audio frames will be provided to it in ULAW format.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_set_write_format(struct ast_rtp_instance *instance, int format);
+
+/*!
+ * \brief Request that the underlying RTP engine make two RTP instances compatible with eachother
+ *
+ * \param chan Our own Asterisk channel
+ * \param instance The first RTP instance
+ * \param peer The peer Asterisk channel
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_make_compatible(instance, peer);
+ * \endcode
+ *
+ * This makes the RTP instance for 'peer' compatible with 'instance' and vice versa.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_make_compatible(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_channel *peer);
+
+/*!
+ * \brief Indicate to the RTP engine that packets are now expected to be sent/received on the RTP instance
+ *
+ * \param instance The RTP instance
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_activate(instance);
+ * \endcode
+ *
+ * This tells the underlying RTP engine of instance that packets will now flow.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_activate(struct ast_rtp_instance *instance);
+
+/*!
+ * \brief Request that the underlying RTP engine send a STUN BIND request
+ *
+ * \param instance The RTP instance
+ * \param suggestion The suggested destination
+ * \param username Optionally a username for the request
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_stun_request(instance, NULL, NULL);
+ * \endcode
+ *
+ * This requests that the RTP engine send a STUN BIND request on the session pointed to by
+ * 'instance'.
+ *
+ * \since 1.6.3
+ */
+void ast_rtp_instance_stun_request(struct ast_rtp_instance *instance, struct sockaddr_in *suggestion, const char *username);
+
+/*!
+ * \brief Set the RTP timeout value
+ *
+ * \param instance The RTP instance
+ * \param timeout Value to set the timeout to
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_set_timeout(instance, 5000);
+ * \endcode
+ *
+ * This sets the RTP timeout value on 'instance' to be 5000.
+ *
+ * \since 1.6.3
+ */
+void ast_rtp_instance_set_timeout(struct ast_rtp_instance *instance, int timeout);
+
+/*!
+ * \brief Set the RTP timeout value for when the instance is on hold
+ *
+ * \param instance The RTP instance
+ * \param timeout Value to set the timeout to
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_set_hold_timeout(instance, 5000);
+ * \endcode
+ *
+ * This sets the RTP hold timeout value on 'instance' to be 5000.
+ *
+ * \since 1.6.3
+ */
+void ast_rtp_instance_set_hold_timeout(struct ast_rtp_instance *instance, int timeout);
+
+/*!
+ * \brief Get the RTP timeout value
+ *
+ * \param instance The RTP instance
+ *
+ * \retval timeout value
+ *
+ * Example usage:
+ *
+ * \code
+ * int timeout = ast_rtp_instance_get_timeout(instance);
+ * \endcode
+ *
+ * This gets the RTP timeout value for the RTP instance pointed to by 'instance'.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_get_timeout(struct ast_rtp_instance *instance);
+
+/*!
+ * \brief Get the RTP timeout value for when an RTP instance is on hold
+ *
+ * \param instance The RTP instance
+ *
+ * \retval timeout value
+ *
+ * Example usage:
+ *
+ * \code
+ * int timeout = ast_rtp_instance_get_hold_timeout(instance);
+ * \endcode
+ *
+ * This gets the RTP hold timeout value for the RTP instance pointed to by 'instance'.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_get_hold_timeout(struct ast_rtp_instance *instance);
+
+#if defined(__cplusplus) || defined(c_plusplus)
+}
+#endif
+
+#endif /* _ASTERISK_RTP_ENGINE_H */
diff --git a/include/asterisk/stun.h b/include/asterisk/stun.h
new file mode 100644
index 0000000000..11921f8148
--- /dev/null
+++ b/include/asterisk/stun.h
@@ -0,0 +1,71 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2008, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*!
+ * \file stun.h
+ * \brief STUN support.
+ *
+ * STUN is defined in RFC 3489.
+ */
+
+#ifndef _ASTERISK_STUN_H
+#define _ASTERISK_STUN_H
+
+#include "asterisk/network.h"
+
+#if defined(__cplusplus) || defined(c_plusplus)
+extern "C" {
+#endif
+
+enum ast_stun_result {
+ AST_STUN_IGNORE = 0,
+ AST_STUN_ACCEPT,
+};
+
+struct stun_attr;
+
+/*! \brief Generic STUN request
+ * send a generic stun request to the server specified.
+ * \param s the socket used to send the request
+ * \param dst the address of the STUN server
+ * \param username if non null, add the username in the request
+ * \param answer if non null, the function waits for a response and
+ * puts here the externally visible address.
+ * \return 0 on success, other values on error.
+ * The interface it may change in the future.
+ */
+int ast_stun_request(int s, struct sockaddr_in *dst, const char *username, struct sockaddr_in *answer);
+
+/*! \brief callback type to be invoked on stun responses. */
+typedef int (stun_cb_f)(struct stun_attr *attr, void *arg);
+
+/*! \brief handle an incoming STUN message.
+ *
+ * Do some basic sanity checks on packet size and content,
+ * try to extract a bit of information, and possibly reply.
+ * At the moment this only processes BIND requests, and returns
+ * the externally visible address of the request.
+ * If a callback is specified, invoke it with the attribute.
+ */
+int ast_stun_handle_packet(int s, struct sockaddr_in *src, unsigned char *data, size_t len, stun_cb_f *stun_cb, void *arg);
+
+#if defined(__cplusplus) || defined(c_plusplus)
+}
+#endif
+
+#endif /* _ASTERISK_STUN_H */
diff --git a/main/Makefile b/main/Makefile
index 3e6179229c..6817197999 100644
--- a/main/Makefile
+++ b/main/Makefile
@@ -20,7 +20,7 @@ include $(ASTTOPDIR)/Makefile.moddir_rules
OBJS= tcptls.o io.o sched.o logger.o frame.o loader.o config.o channel.o \
translate.o file.o pbx.o cli.o md5.o term.o heap.o \
ulaw.o alaw.o callerid.o fskmodem.o image.o app.o \
- cdr.o tdd.o acl.o rtp.o udptl.o manager.o asterisk.o \
+ cdr.o tdd.o acl.o udptl.o manager.o asterisk.o \
dsp.o chanvars.o indications.o autoservice.o db.o privacy.o \
astmm.o enum.o srv.o dns.o aescrypt.o aestab.o aeskey.o \
utils.o plc.o jitterbuf.o dnsmgr.o devicestate.o \
@@ -29,7 +29,7 @@ OBJS= tcptls.o io.o sched.o logger.o frame.o loader.o config.o channel.o \
strcompat.o threadstorage.o dial.o event.o adsistub.o audiohook.o \
astobj2.o hashtab.o global_datastores.o version.o \
features.o taskprocessor.o timing.o datastore.o xml.o xmldoc.o \
- strings.o bridging.o poll.o
+ strings.o bridging.o poll.o rtp_engine.o stun.o
# we need to link in the objects statically, not as a library, because
# otherwise modules will not have them available if none of the static
diff --git a/main/asterisk.c b/main/asterisk.c
index fdef5e156e..20c85b47fe 100644
--- a/main/asterisk.c
+++ b/main/asterisk.c
@@ -120,7 +120,6 @@ int daemon(int, int); /* defined in libresolv of all places */
#include "asterisk/cdr.h"
#include "asterisk/pbx.h"
#include "asterisk/enum.h"
-#include "asterisk/rtp.h"
#include "asterisk/http.h"
#include "asterisk/udptl.h"
#include "asterisk/app.h"
@@ -3579,7 +3578,6 @@ int main(int argc, char *argv[])
exit(1);
}
- ast_rtp_init();
ast_dsp_init();
ast_udptl_init();
diff --git a/main/loader.c b/main/loader.c
index 5f2fe86786..4e07e843b5 100644
--- a/main/loader.c
+++ b/main/loader.c
@@ -43,7 +43,6 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/manager.h"
#include "asterisk/cdr.h"
#include "asterisk/enum.h"
-#include "asterisk/rtp.h"
#include "asterisk/http.h"
#include "asterisk/lock.h"
#include "asterisk/features.h"
@@ -243,7 +242,6 @@ static struct reload_classes {
{ "extconfig", read_config_maps },
{ "enum", ast_enum_reload },
{ "manager", reload_manager },
- { "rtp", ast_rtp_reload },
{ "http", ast_http_reload },
{ "logger", logger_reload },
{ "features", ast_features_reload },
diff --git a/main/rtp.c b/main/rtp.c
deleted file mode 100644
index 38ff9ad3ab..0000000000
--- a/main/rtp.c
+++ /dev/null
@@ -1,4865 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 1999 - 2006, Digium, Inc.
- *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*!
- * \file
- *
- * \brief Supports RTP and RTCP with Symmetric RTP support for NAT traversal.
- *
- * \author Mark Spencer <markster@digium.com>
- *
- * \note RTP is defined in RFC 3550.
- */
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include <sys/time.h>
-#include <signal.h>
-#include <fcntl.h>
-#include <math.h>
-
-#include "asterisk/rtp.h"
-#include "asterisk/pbx.h"
-#include "asterisk/frame.h"
-#include "asterisk/channel.h"
-#include "asterisk/acl.h"
-#include "asterisk/config.h"
-#include "asterisk/lock.h"
-#include "asterisk/utils.h"
-#include "asterisk/netsock.h"
-#include "asterisk/cli.h"
-#include "asterisk/manager.h"
-#include "asterisk/unaligned.h"
-
-#define MAX_TIMESTAMP_SKEW 640
-
-#define RTP_SEQ_MOD (1<<16) /*!< A sequence number can't be more than 16 bits */
-#define RTCP_DEFAULT_INTERVALMS 5000 /*!< Default milli-seconds between RTCP reports we send */
-#define RTCP_MIN_INTERVALMS 500 /*!< Min milli-seconds between RTCP reports we send */
-#define RTCP_MAX_INTERVALMS 60000 /*!< Max milli-seconds between RTCP reports we send */
-
-#define RTCP_PT_FUR 192
-#define RTCP_PT_SR 200
-#define RTCP_PT_RR 201
-#define RTCP_PT_SDES 202
-#define RTCP_PT_BYE 203
-#define RTCP_PT_APP 204
-
-#define RTP_MTU 1200
-
-#define DEFAULT_DTMF_TIMEOUT 3000 /*!< samples */
-
-static int dtmftimeout = DEFAULT_DTMF_TIMEOUT;
-
-static int rtpstart = 5000; /*!< First port for RTP sessions (set in rtp.conf) */
-static int rtpend = 31000; /*!< Last port for RTP sessions (set in rtp.conf) */
-static int rtpdebug; /*!< Are we debugging? */
-static int rtcpdebug; /*!< Are we debugging RTCP? */
-static int rtcpstats; /*!< Are we debugging RTCP? */
-static int rtcpinterval = RTCP_DEFAULT_INTERVALMS; /*!< Time between rtcp reports in millisecs */
-static int stundebug; /*!< Are we debugging stun? */
-static struct sockaddr_in rtpdebugaddr; /*!< Debug packets to/from this host */
-static struct sockaddr_in rtcpdebugaddr; /*!< Debug RTCP packets to/from this host */
-#ifdef SO_NO_CHECK
-static int nochecksums;
-#endif
-static int strictrtp;
-
-enum strict_rtp_state {
- STRICT_RTP_OPEN = 0, /*! No RTP packets should be dropped, all sources accepted */
- STRICT_RTP_LEARN, /*! Accept next packet as source */
- STRICT_RTP_CLOSED, /*! Drop all RTP packets not coming from source that was learned */
-};
-
-/* Uncomment this to enable more intense native bridging, but note: this is currently buggy */
-/* #define P2P_INTENSE */
-
-/*!
- * \brief Structure representing a RTP session.
- *
- * RTP session is defined on page 9 of RFC 3550: "An association among a set of participants communicating with RTP. A participant may be involved in multiple RTP sessions at the same time [...]"
- *
- */
-
-/*! \brief RTP session description */
-struct ast_rtp {
- int s;
- struct ast_frame f;
- unsigned char rawdata[8192 + AST_FRIENDLY_OFFSET];
- unsigned int ssrc; /*!< Synchronization source, RFC 3550, page 10. */
- unsigned int themssrc; /*!< Their SSRC */
- unsigned int rxssrc;
- unsigned int lastts;
- unsigned int lastrxts;
- unsigned int lastividtimestamp;
- unsigned int lastovidtimestamp;
- unsigned int lastitexttimestamp;
- unsigned int lastotexttimestamp;
- unsigned int lasteventseqn;
- int lastrxseqno; /*!< Last received sequence number */
- unsigned short seedrxseqno; /*!< What sequence number did they start with?*/
- unsigned int seedrxts; /*!< What RTP timestamp did they start with? */
- unsigned int rxcount; /*!< How many packets have we received? */
- unsigned int rxoctetcount; /*!< How many octets have we received? should be rxcount *160*/
- unsigned int txcount; /*!< How many packets have we sent? */
- unsigned int txoctetcount; /*!< How many octets have we sent? (txcount*160)*/
- unsigned int cycles; /*!< Shifted count of sequence number cycles */
- double rxjitter; /*!< Interarrival jitter at the moment */
- double rxtransit; /*!< Relative transit time for previous packet */
- int lasttxformat;
- int lastrxformat;
-
- int rtptimeout; /*!< RTP timeout time (negative or zero means disabled, negative value means temporarily disabled) */
- int rtpholdtimeout; /*!< RTP timeout when on hold (negative or zero means disabled, negative value means temporarily disabled). */
- int rtpkeepalive; /*!< Send RTP comfort noice packets for keepalive */
-
- /* DTMF Reception Variables */
- char resp;
- unsigned int lastevent;
- int dtmfcount;
- unsigned int dtmfsamples;
- /* DTMF Transmission Variables */
- unsigned int lastdigitts;
- char sending_digit; /*!< boolean - are we sending digits */
- char send_digit; /*!< digit we are sending */
- int send_payload;
- int send_duration;
- int nat;
- unsigned int flags;
- struct sockaddr_in us; /*!< Socket representation of the local endpoint. */
- struct sockaddr_in them; /*!< Socket representation of the remote endpoint. */
- struct timeval rxcore;
- struct timeval txcore;
- double drxcore; /*!< The double representation of the first received packet */
- struct timeval lastrx; /*!< timeval when we last received a packet */
- struct timeval dtmfmute;
- struct ast_smoother *smoother;
- int *ioid;
- unsigned short seqno; /*!< Sequence number, RFC 3550, page 13. */
- unsigned short rxseqno;
- struct sched_context *sched;
- struct io_context *io;
- void *data;
- ast_rtp_callback callback;
-#ifdef P2P_INTENSE
- ast_mutex_t bridge_lock;
-#endif
- struct rtpPayloadType current_RTP_PT[MAX_RTP_PT];
- int rtp_lookup_code_cache_isAstFormat; /*!< a cache for the result of rtp_lookup_code(): */
- int rtp_lookup_code_cache_code;
- int rtp_lookup_code_cache_result;
- struct ast_rtcp *rtcp;
- struct ast_codec_pref pref;
- struct ast_rtp *bridged; /*!< Who we are Packet bridged to */
-
- enum strict_rtp_state strict_rtp_state; /*!< Current state that strict RTP protection is in */
- struct sockaddr_in strict_rtp_address; /*!< Remote address information for strict RTP purposes */
-
- int set_marker_bit:1; /*!< Whether to set the marker bit or not */
- struct rtp_red *red;
-};
-
-static struct ast_frame *red_t140_to_red(struct rtp_red *red);
-static int red_write(const void *data);
-
-struct rtp_red {
- struct ast_frame t140; /*!< Primary data */
- struct ast_frame t140red; /*!< Redundant t140*/
- unsigned char pt[RED_MAX_GENERATION]; /*!< Payload types for redundancy data */
- unsigned char ts[RED_MAX_GENERATION]; /*!< Time stamps */
- unsigned char len[RED_MAX_GENERATION]; /*!< length of each generation */
- int num_gen; /*!< Number of generations */
- int schedid; /*!< Timer id */
- int ti; /*!< How long to buffer data before send */
- unsigned char t140red_data[64000];
- unsigned char buf_data[64000]; /*!< buffered primary data */
- int hdrlen;
- long int prev_ts;
-};
-
-/* Forward declarations */
-static int ast_rtcp_write(const void *data);
-static void timeval2ntp(struct timeval tv, unsigned int *msw, unsigned int *lsw);
-static int ast_rtcp_write_sr(const void *data);
-static int ast_rtcp_write_rr(const void *data);
-static unsigned int ast_rtcp_calc_interval(struct ast_rtp *rtp);
-static int ast_rtp_senddigit_continuation(struct ast_rtp *rtp);
-int ast_rtp_senddigit_end(struct ast_rtp *rtp, char digit);
-
-#define FLAG_3389_WARNING (1 << 0)
-#define FLAG_NAT_ACTIVE (3 << 1)
-#define FLAG_NAT_INACTIVE (0 << 1)
-#define FLAG_NAT_INACTIVE_NOWARN (1 << 1)
-#define FLAG_HAS_DTMF (1 << 3)
-#define FLAG_P2P_SENT_MARK (1 << 4)
-#define FLAG_P2P_NEED_DTMF (1 << 5)
-#define FLAG_CALLBACK_MODE (1 << 6)
-#define FLAG_DTMF_COMPENSATE (1 << 7)
-#define FLAG_HAS_STUN (1 << 8)
-
-/*!
- * \brief Structure defining an RTCP session.
- *
- * The concept "RTCP session" is not defined in RFC 3550, but since
- * this structure is analogous to ast_rtp, which tracks a RTP session,
- * it is logical to think of this as a RTCP session.
- *
- * RTCP packet is defined on page 9 of RFC 3550.
- *
- */
-struct ast_rtcp {
- int rtcp_info;
- int s; /*!< Socket */
- struct sockaddr_in us; /*!< Socket representation of the local endpoint. */
- struct sockaddr_in them; /*!< Socket representation of the remote endpoint. */
- unsigned int soc; /*!< What they told us */
- unsigned int spc; /*!< What they told us */
- unsigned int themrxlsr; /*!< The middle 32 bits of the NTP timestamp in the last received SR*/
- struct timeval rxlsr; /*!< Time when we got their last SR */
- struct timeval txlsr; /*!< Time when we sent or last SR*/
- unsigned int expected_prior; /*!< no. packets in previous interval */
- unsigned int received_prior; /*!< no. packets received in previous interval */
- int schedid; /*!< Schedid returned from ast_sched_add() to schedule RTCP-transmissions*/
- unsigned int rr_count; /*!< number of RRs we've sent, not including report blocks in SR's */
- unsigned int sr_count; /*!< number of SRs we've sent */
- unsigned int lastsrtxcount; /*!< Transmit packet count when last SR sent */
- double accumulated_transit; /*!< accumulated a-dlsr-lsr */
- double rtt; /*!< Last reported rtt */
- unsigned int reported_jitter; /*!< The contents of their last jitter entry in the RR */
- unsigned int reported_lost; /*!< Reported lost packets in their RR */
- char quality[AST_MAX_USER_FIELD];
- char quality_jitter[AST_MAX_USER_FIELD];
- char quality_loss[AST_MAX_USER_FIELD];
- char quality_rtt[AST_MAX_USER_FIELD];
-
- double reported_maxjitter;
- double reported_minjitter;
- double reported_normdev_jitter;
- double reported_stdev_jitter;
- unsigned int reported_jitter_count;
-
- double reported_maxlost;
- double reported_minlost;
- double reported_normdev_lost;
- double reported_stdev_lost;
-
- double rxlost;
- double maxrxlost;
- double minrxlost;
- double normdev_rxlost;
- double stdev_rxlost;
- unsigned int rxlost_count;
-
- double maxrxjitter;
- double minrxjitter;
- double normdev_rxjitter;
- double stdev_rxjitter;
- unsigned int rxjitter_count;
- double maxrtt;
- double minrtt;
- double normdevrtt;
- double stdevrtt;
- unsigned int rtt_count;
- int sendfur;
-};
-
-/*!
- * \brief STUN support code
- *
- * This code provides some support for doing STUN transactions.
- * Eventually it should be moved elsewhere as other protocols
- * than RTP can benefit from it - e.g. SIP.
- * STUN is described in RFC3489 and it is based on the exchange
- * of UDP packets between a client and one or more servers to
- * determine the externally visible address (and port) of the client
- * once it has gone through the NAT boxes that connect it to the
- * outside.
- * The simplest request packet is just the header defined in
- * struct stun_header, and from the response we may just look at
- * one attribute, STUN_MAPPED_ADDRESS, that we find in the response.
- * By doing more transactions with different server addresses we
- * may determine more about the behaviour of the NAT boxes, of
- * course - the details are in the RFC.
- *
- * All STUN packets start with a simple header made of a type,
- * length (excluding the header) and a 16-byte random transaction id.
- * Following the header we may have zero or more attributes, each
- * structured as a type, length and a value (whose format depends
- * on the type, but often contains addresses).
- * Of course all fields are in network format.
- */
-
-typedef struct { unsigned int id[4]; } __attribute__((packed)) stun_trans_id;
-
-struct stun_header {
- unsigned short msgtype;
- unsigned short msglen;
- stun_trans_id id;
- unsigned char ies[0];
-} __attribute__((packed));
-
-struct stun_attr {
- unsigned short attr;
- unsigned short len;
- unsigned char value[0];
-} __attribute__((packed));
-
-/*
- * The format normally used for addresses carried by STUN messages.
- */
-struct stun_addr {
- unsigned char unused;
- unsigned char family;
- unsigned short port;
- unsigned int addr;
-} __attribute__((packed));
-
-#define STUN_IGNORE (0)
-#define STUN_ACCEPT (1)
-
-/*! \brief STUN message types
- * 'BIND' refers to transactions used to determine the externally
- * visible addresses. 'SEC' refers to transactions used to establish
- * a session key for subsequent requests.
- * 'SEC' functionality is not supported here.
- */
-
-#define STUN_BINDREQ 0x0001
-#define STUN_BINDRESP 0x0101
-#define STUN_BINDERR 0x0111
-#define STUN_SECREQ 0x0002
-#define STUN_SECRESP 0x0102
-#define STUN_SECERR 0x0112
-
-/*! \brief Basic attribute types in stun messages.
- * Messages can also contain custom attributes (codes above 0x7fff)
- */
-#define STUN_MAPPED_ADDRESS 0x0001
-#define STUN_RESPONSE_ADDRESS 0x0002
-#define STUN_CHANGE_REQUEST 0x0003
-#define STUN_SOURCE_ADDRESS 0x0004
-#define STUN_CHANGED_ADDRESS 0x0005
-#define STUN_USERNAME 0x0006
-#define STUN_PASSWORD 0x0007
-#define STUN_MESSAGE_INTEGRITY 0x0008
-#define STUN_ERROR_CODE 0x0009
-#define STUN_UNKNOWN_ATTRIBUTES 0x000a
-#define STUN_REFLECTED_FROM 0x000b
-
-/*! \brief helper function to print message names */
-static const char *stun_msg2str(int msg)
-{
- switch (msg) {
- case STUN_BINDREQ:
- return "Binding Request";
- case STUN_BINDRESP:
- return "Binding Response";
- case STUN_BINDERR:
- return "Binding Error Response";
- case STUN_SECREQ:
- return "Shared Secret Request";
- case STUN_SECRESP:
- return "Shared Secret Response";
- case STUN_SECERR:
- return "Shared Secret Error Response";
- }
- return "Non-RFC3489 Message";
-}
-
-/*! \brief helper function to print attribute names */
-static const char *stun_attr2str(int msg)
-{
- switch (msg) {
- case STUN_MAPPED_ADDRESS:
- return "Mapped Address";
- case STUN_RESPONSE_ADDRESS:
- return "Response Address";
- case STUN_CHANGE_REQUEST:
- return "Change Request";
- case STUN_SOURCE_ADDRESS:
- return "Source Address";
- case STUN_CHANGED_ADDRESS:
- return "Changed Address";
- case STUN_USERNAME:
- return "Username";
- case STUN_PASSWORD:
- return "Password";
- case STUN_MESSAGE_INTEGRITY:
- return "Message Integrity";
- case STUN_ERROR_CODE:
- return "Error Code";
- case STUN_UNKNOWN_ATTRIBUTES:
- return "Unknown Attributes";
- case STUN_REFLECTED_FROM:
- return "Reflected From";
- }
- return "Non-RFC3489 Attribute";
-}
-
-/*! \brief here we store credentials extracted from a message */
-struct stun_state {
- const char *username;
- const char *password;
-};
-
-static int stun_process_attr(struct stun_state *state, struct stun_attr *attr)
-{
- if (stundebug)
- ast_verbose("Found STUN Attribute %s (%04x), length %d\n",
- stun_attr2str(ntohs(attr->attr)), ntohs(attr->attr), ntohs(attr->len));
- switch (ntohs(attr->attr)) {
- case STUN_USERNAME:
- state->username = (const char *) (attr->value);
- break;
- case STUN_PASSWORD:
- state->password = (const char *) (attr->value);
- break;
- default:
- if (stundebug)
- ast_verbose("Ignoring STUN attribute %s (%04x), length %d\n",
- stun_attr2str(ntohs(attr->attr)), ntohs(attr->attr), ntohs(attr->len));
- }
- return 0;
-}
-
-/*! \brief append a string to an STUN message */
-static void append_attr_string(struct stun_attr **attr, int attrval, const char *s, int *len, int *left)
-{
- int size = sizeof(**attr) + strlen(s);
- if (*left > size) {
- (*attr)->attr = htons(attrval);
- (*attr)->len = htons(strlen(s));
- memcpy((*attr)->value, s, strlen(s));
- (*attr) = (struct stun_attr *)((*attr)->value + strlen(s));
- *len += size;
- *left -= size;
- }
-}
-
-/*! \brief append an address to an STUN message */
-static void append_attr_address(struct stun_attr **attr, int attrval, struct sockaddr_in *sock_in, int *len, int *left)
-{
- int size = sizeof(**attr) + 8;
- struct stun_addr *addr;
- if (*left > size) {
- (*attr)->attr = htons(attrval);
- (*attr)->len = htons(8);
- addr = (struct stun_addr *)((*attr)->value);
- addr->unused = 0;
- addr->family = 0x01;
- addr->port = sock_in->sin_port;
- addr->addr = sock_in->sin_addr.s_addr;
- (*attr) = (struct stun_attr *)((*attr)->value + 8);
- *len += size;
- *left -= size;
- }
-}
-
-/*! \brief wrapper to send an STUN message */
-static int stun_send(int s, struct sockaddr_in *dst, struct stun_header *resp)
-{
- return sendto(s, resp, ntohs(resp->msglen) + sizeof(*resp), 0,
- (struct sockaddr *)dst, sizeof(*dst));
-}
-
-/*! \brief helper function to generate a random request id */
-static void stun_req_id(struct stun_header *req)
-{
- int x;
- for (x = 0; x < 4; x++)
- req->id.id[x] = ast_random();
-}
-
-size_t ast_rtp_alloc_size(void)
-{
- return sizeof(struct ast_rtp);
-}
-
-/*! \brief callback type to be invoked on stun responses. */
-typedef int (stun_cb_f)(struct stun_attr *attr, void *arg);
-
-/*! \brief handle an incoming STUN message.
- *
- * Do some basic sanity checks on packet size and content,
- * try to extract a bit of information, and possibly reply.
- * At the moment this only processes BIND requests, and returns
- * the externally visible address of the request.
- * If a callback is specified, invoke it with the attribute.
- */
-static int stun_handle_packet(int s, struct sockaddr_in *src,
- unsigned char *data, size_t len, stun_cb_f *stun_cb, void *arg)
-{
- struct stun_header *hdr = (struct stun_header *)data;
- struct stun_attr *attr;
- struct stun_state st;
- int ret = STUN_IGNORE;
- int x;
-
- /* On entry, 'len' is the length of the udp payload. After the
- * initial checks it becomes the size of unprocessed options,
- * while 'data' is advanced accordingly.
- */
- if (len < sizeof(struct stun_header)) {
- ast_debug(1, "Runt STUN packet (only %d, wanting at least %d)\n", (int) len, (int) sizeof(struct stun_header));
- return -1;
- }
- len -= sizeof(struct stun_header);
- data += sizeof(struct stun_header);
- x = ntohs(hdr->msglen); /* len as advertised in the message */
- if (stundebug)
- ast_verbose("STUN Packet, msg %s (%04x), length: %d\n", stun_msg2str(ntohs(hdr->msgtype)), ntohs(hdr->msgtype), x);
- if (x > len) {
- ast_debug(1, "Scrambled STUN packet length (got %d, expecting %d)\n", x, (int)len);
- } else
- len = x;
- memset(&st, 0, sizeof(st));
- while (len) {
- if (len < sizeof(struct stun_attr)) {
- ast_debug(1, "Runt Attribute (got %d, expecting %d)\n", (int)len, (int) sizeof(struct stun_attr));
- break;
- }
- attr = (struct stun_attr *)data;
- /* compute total attribute length */
- x = ntohs(attr->len) + sizeof(struct stun_attr);
- if (x > len) {
- ast_debug(1, "Inconsistent Attribute (length %d exceeds remaining msg len %d)\n", x, (int)len);
- break;
- }
- if (stun_cb)
- stun_cb(attr, arg);
- if (stun_process_attr(&st, attr)) {
- ast_debug(1, "Failed to handle attribute %s (%04x)\n", stun_attr2str(ntohs(attr->attr)), ntohs(attr->attr));
- break;
- }
- /* Clear attribute id: in case previous entry was a string,
- * this will act as the terminator for the string.
- */
- attr->attr = 0;
- data += x;
- len -= x;
- }
- /* Null terminate any string.
- * XXX NOTE, we write past the size of the buffer passed by the
- * caller, so this is potentially dangerous. The only thing that
- * saves us is that usually we read the incoming message in a
- * much larger buffer in the struct ast_rtp
- */
- *data = '\0';
-
- /* Now prepare to generate a reply, which at the moment is done
- * only for properly formed (len == 0) STUN_BINDREQ messages.
- */
- if (len == 0) {
- unsigned char respdata[1024];
- struct stun_header *resp = (struct stun_header *)respdata;
- int resplen = 0; /* len excluding header */
- int respleft = sizeof(respdata) - sizeof(struct stun_header);
-
- resp->id = hdr->id;
- resp->msgtype = 0;
- resp->msglen = 0;
- attr = (struct stun_attr *)resp->ies;
- switch (ntohs(hdr->msgtype)) {
- case STUN_BINDREQ:
- if (stundebug)
- ast_verbose("STUN Bind Request, username: %s\n",
- st.username ? st.username : "<none>");
- if (st.username)
- append_attr_string(&attr, STUN_USERNAME, st.username, &resplen, &respleft);
- append_attr_address(&attr, STUN_MAPPED_ADDRESS, src, &resplen, &respleft);
- resp->msglen = htons(resplen);
- resp->msgtype = htons(STUN_BINDRESP);
- stun_send(s, src, resp);
- ret = STUN_ACCEPT;
- break;
- default:
- if (stundebug)
- ast_verbose("Dunno what to do with STUN message %04x (%s)\n", ntohs(hdr->msgtype), stun_msg2str(ntohs(hdr->msgtype)));
- }
- }
- return ret;
-}
-
-/*! \brief Extract the STUN_MAPPED_ADDRESS from the stun response.
- * This is used as a callback for stun_handle_response
- * when called from ast_stun_request.
- */
-static int stun_get_mapped(struct stun_attr *attr, void *arg)
-{
- struct stun_addr *addr = (struct stun_addr *)(attr + 1);
- struct sockaddr_in *sa = (struct sockaddr_in *)arg;
-
- if (ntohs(attr->attr) != STUN_MAPPED_ADDRESS || ntohs(attr->len) != 8)
- return 1; /* not us. */
- sa->sin_port = addr->port;
- sa->sin_addr.s_addr = addr->addr;
- return 0;
-}
-
-/*! \brief Generic STUN request
- * Send a generic stun request to the server specified,
- * possibly waiting for a reply and filling the 'reply' field with
- * the externally visible address. Note that in this case the request
- * will be blocking.
- * (Note, the interface may change slightly in the future).
- *
- * \param s the socket used to send the request
- * \param dst the address of the STUN server
- * \param username if non null, add the username in the request
- * \param answer if non null, the function waits for a response and
- * puts here the externally visible address.
- * \return 0 on success, other values on error.
- */
-int ast_stun_request(int s, struct sockaddr_in *dst,
- const char *username, struct sockaddr_in *answer)
-{
- struct stun_header *req;
- unsigned char reqdata[1024];
- int reqlen, reqleft;
- struct stun_attr *attr;
- int res = 0;
- int retry;
-
- req = (struct stun_header *)reqdata;
- stun_req_id(req);
- reqlen = 0;
- reqleft = sizeof(reqdata) - sizeof(struct stun_header);
- req->msgtype = 0;
- req->msglen = 0;
- attr = (struct stun_attr *)req->ies;
- if (username)
- append_attr_string(&attr, STUN_USERNAME, username, &reqlen, &reqleft);
- req->msglen = htons(reqlen);
- req->msgtype = htons(STUN_BINDREQ);
- for (retry = 0; retry < 3; retry++) { /* XXX make retries configurable */
- /* send request, possibly wait for reply */
- unsigned char reply_buf[1024];
- fd_set rfds;
- struct timeval to = { 3, 0 }; /* timeout, make it configurable */
- struct sockaddr_in src;
- socklen_t srclen;
-
- res = stun_send(s, dst, req);
- if (res < 0) {
- ast_log(LOG_WARNING, "ast_stun_request send #%d failed error %d, retry\n",
- retry, res);
- continue;
- }
- if (answer == NULL)
- break;
- FD_ZERO(&rfds);
- FD_SET(s, &rfds);
- res = ast_select(s + 1, &rfds, NULL, NULL, &to);
- if (res <= 0) /* timeout or error */
- continue;
- memset(&src, '\0', sizeof(src));
- srclen = sizeof(src);
- /* XXX pass -1 in the size, because stun_handle_packet might
- * write past the end of the buffer.
- */
- res = recvfrom(s, reply_buf, sizeof(reply_buf) - 1,
- 0, (struct sockaddr *)&src, &srclen);
- if (res < 0) {
- ast_log(LOG_WARNING, "ast_stun_request recvfrom #%d failed error %d, retry\n",
- retry, res);
- continue;
- }
- memset(answer, '\0', sizeof(struct sockaddr_in));
- stun_handle_packet(s, &src, reply_buf, res,
- stun_get_mapped, answer);
- res = 0; /* signal regular exit */
- break;
- }
- return res;
-}
-
-/*! \brief send a STUN BIND request to the given destination.
- * Optionally, add a username if specified.
- */
-void ast_rtp_stun_request(struct ast_rtp *rtp, struct sockaddr_in *suggestion, const char *username)
-{
- ast_stun_request(rtp->s, suggestion, username, NULL);
-}
-
-/*! \brief List of current sessions */
-static AST_RWLIST_HEAD_STATIC(protos, ast_rtp_protocol);
-
-static void timeval2ntp(struct timeval when, unsigned int *msw, unsigned int *lsw)
-{
- unsigned int sec, usec, frac;
- sec = when.tv_sec + 2208988800u; /* Sec between 1900 and 1970 */
- usec = when.tv_usec;
- frac = (usec << 12) + (usec << 8) - ((usec * 3650) >> 6);
- *msw = sec;
- *lsw = frac;
-}
-
-int ast_rtp_fd(struct ast_rtp *rtp)
-{
- return rtp->s;
-}
-
-int ast_rtcp_fd(struct ast_rtp *rtp)
-{
- if (rtp->rtcp)
- return rtp->rtcp->s;
- return -1;
-}
-
-unsigned int ast_rtcp_calc_interval(struct ast_rtp *rtp)
-{
- unsigned int interval;
- /*! \todo XXX Do a more reasonable calculation on this one
- * Look in RFC 3550 Section A.7 for an example*/
- interval = rtcpinterval;
- return interval;
-}
-
-/* \brief Put RTP timeout timers on hold during another transaction, like T.38 */
-void ast_rtp_set_rtptimers_onhold(struct ast_rtp *rtp)
-{
- rtp->rtptimeout = (-1) * rtp->rtptimeout;
- rtp->rtpholdtimeout = (-1) * rtp->rtpholdtimeout;
-}
-
-/*! \brief Set rtp timeout */
-void ast_rtp_set_rtptimeout(struct ast_rtp *rtp, int timeout)
-{
- rtp->rtptimeout = timeout;
-}
-
-/*! \brief Set rtp hold timeout */
-void ast_rtp_set_rtpholdtimeout(struct ast_rtp *rtp, int timeout)
-{
- rtp->rtpholdtimeout = timeout;
-}
-
-/*! \brief set RTP keepalive interval */
-void ast_rtp_set_rtpkeepalive(struct ast_rtp *rtp, int period)
-{
- rtp->rtpkeepalive = period;
-}
-
-/*! \brief Get rtp timeout */
-int ast_rtp_get_rtptimeout(struct ast_rtp *rtp)
-{
- if (rtp->rtptimeout < 0) /* We're not checking, but remembering the setting (during T.38 transmission) */
- return 0;
- return rtp->rtptimeout;
-}
-
-/*! \brief Get rtp hold timeout */
-int ast_rtp_get_rtpholdtimeout(struct ast_rtp *rtp)
-{
- if (rtp->rtptimeout < 0) /* We're not checking, but remembering the setting (during T.38 transmission) */
- return 0;
- return rtp->rtpholdtimeout;
-}
-
-/*! \brief Get RTP keepalive interval */
-int ast_rtp_get_rtpkeepalive(struct ast_rtp *rtp)
-{
- return rtp->rtpkeepalive;
-}
-
-void ast_rtp_set_data(struct ast_rtp *rtp, void *data)
-{
- rtp->data = data;
-}
-
-void ast_rtp_set_callback(struct ast_rtp *rtp, ast_rtp_callback callback)
-{
- rtp->callback = callback;
-}
-
-void ast_rtp_setnat(struct ast_rtp *rtp, int nat)
-{
- rtp->nat = nat;
-}
-
-int ast_rtp_getnat(struct ast_rtp *rtp)
-{
- return ast_test_flag(rtp, FLAG_NAT_ACTIVE);
-}
-
-void ast_rtp_setdtmf(struct ast_rtp *rtp, int dtmf)
-{
- ast_set2_flag(rtp, dtmf ? 1 : 0, FLAG_HAS_DTMF);
-}
-
-void ast_rtp_setdtmfcompensate(struct ast_rtp *rtp, int compensate)
-{
- ast_set2_flag(rtp, compensate ? 1 : 0, FLAG_DTMF_COMPENSATE);
-}
-
-void ast_rtp_setstun(struct ast_rtp *rtp, int stun_enable)
-{
- ast_set2_flag(rtp, stun_enable ? 1 : 0, FLAG_HAS_STUN);
-}
-
-static void rtp_bridge_lock(struct ast_rtp *rtp)
-{
-#ifdef P2P_INTENSE
- ast_mutex_lock(&rtp->bridge_lock);
-#endif
- return;
-}
-
-static void rtp_bridge_unlock(struct ast_rtp *rtp)
-{
-#ifdef P2P_INTENSE
- ast_mutex_unlock(&rtp->bridge_lock);
-#endif
- return;
-}
-
-/*! \brief Calculate normal deviation */
-static double normdev_compute(double normdev, double sample, unsigned int sample_count)
-{
- normdev = normdev * sample_count + sample;
- sample_count++;
-
- return normdev / sample_count;
-}
-
-static double stddev_compute(double stddev, double sample, double normdev, double normdev_curent, unsigned int sample_count)
-{
-/*
- for the formula check http://www.cs.umd.edu/~austinjp/constSD.pdf
- return sqrt( (sample_count*pow(stddev,2) + sample_count*pow((sample-normdev)/(sample_count+1),2) + pow(sample-normdev_curent,2)) / (sample_count+1));
- we can compute the sigma^2 and that way we would have to do the sqrt only 1 time at the end and would save another pow 2 compute
- optimized formula
-*/
-#define SQUARE(x) ((x) * (x))
-
- stddev = sample_count * stddev;
- sample_count++;
-
- return stddev +
- ( sample_count * SQUARE( (sample - normdev) / sample_count ) ) +
- ( SQUARE(sample - normdev_curent) / sample_count );
-
-#undef SQUARE
-}
-
-static struct ast_frame *send_dtmf(struct ast_rtp *rtp, enum ast_frame_type type)
-{
- if (((ast_test_flag(rtp, FLAG_DTMF_COMPENSATE) && type == AST_FRAME_DTMF_END) ||
- (type == AST_FRAME_DTMF_BEGIN)) && ast_tvcmp(ast_tvnow(), rtp->dtmfmute) < 0) {
- ast_debug(1, "Ignore potential DTMF echo from '%s'\n", ast_inet_ntoa(rtp->them.sin_addr));
- rtp->resp = 0;
- rtp->dtmfsamples = 0;
- return &ast_null_frame;
- }
- ast_debug(1, "Sending dtmf: %d (%c), at %s\n", rtp->resp, rtp->resp, ast_inet_ntoa(rtp->them.sin_addr));
- if (rtp->resp == 'X') {
- rtp->f.frametype = AST_FRAME_CONTROL;
- rtp->f.subclass = AST_CONTROL_FLASH;
- } else {
- rtp->f.frametype = type;
- rtp->f.subclass = rtp->resp;
- }
- rtp->f.datalen = 0;
- rtp->f.samples = 0;
- rtp->f.mallocd = 0;
- rtp->f.src = "RTP";
- return &rtp->f;
-
-}
-
-static inline int rtp_debug_test_addr(struct sockaddr_in *addr)
-{
- if (rtpdebug == 0)
- return 0;
- if (rtpdebugaddr.sin_addr.s_addr) {
- if (((ntohs(rtpdebugaddr.sin_port) != 0)
- && (rtpdebugaddr.sin_port != addr->sin_port))
- || (rtpdebugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
- return 0;
- }
- return 1;
-}
-
-static inline int rtcp_debug_test_addr(struct sockaddr_in *addr)
-{
- if (rtcpdebug == 0)
- return 0;
- if (rtcpdebugaddr.sin_addr.s_addr) {
- if (((ntohs(rtcpdebugaddr.sin_port) != 0)
- && (rtcpdebugaddr.sin_port != addr->sin_port))
- || (rtcpdebugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
- return 0;
- }
- return 1;
-}
-
-
-static struct ast_frame *process_cisco_dtmf(struct ast_rtp *rtp, unsigned char *data, int len)
-{
- unsigned int event;
- char resp = 0;
- struct ast_frame *f = NULL;
- unsigned char seq;
- unsigned int flags;
- unsigned int power;
-
- /* We should have at least 4 bytes in RTP data */
- if (len < 4)
- return f;
-
- /* The format of Cisco RTP DTMF packet looks like next:
- +0 - sequence number of DTMF RTP packet (begins from 1,
- wrapped to 0)
- +1 - set of flags
- +1 (bit 0) - flaps by different DTMF digits delimited by audio
- or repeated digit without audio???
- +2 (+4,+6,...) - power level? (rises from 0 to 32 at begin of tone
- then falls to 0 at its end)
- +3 (+5,+7,...) - detected DTMF digit (0..9,*,#,A-D,...)
- Repeated DTMF information (bytes 4/5, 6/7) is history shifted right
- by each new packet and thus provides some redudancy.
-
- Sample of Cisco RTP DTMF packet is (all data in hex):
- 19 07 00 02 12 02 20 02
- showing end of DTMF digit '2'.
-
- The packets
- 27 07 00 02 0A 02 20 02
- 28 06 20 02 00 02 0A 02
- shows begin of new digit '2' with very short pause (20 ms) after
- previous digit '2'. Bit +1.0 flips at begin of new digit.
-
- Cisco RTP DTMF packets comes as replacement of audio RTP packets
- so its uses the same sequencing and timestamping rules as replaced
- audio packets. Repeat interval of DTMF packets is 20 ms and not rely
- on audio framing parameters. Marker bit isn't used within stream of
- DTMFs nor audio stream coming immediately after DTMF stream. Timestamps
- are not sequential at borders between DTMF and audio streams,
- */
-
- seq = data[0];
- flags = data[1];
- power = data[2];
- event = data[3] & 0x1f;
-
- if (option_debug > 2 || rtpdebug)
- ast_debug(0, "Cisco DTMF Digit: %02x (len=%d, seq=%d, flags=%02x, power=%d, history count=%d)\n", event, len, seq, flags, power, (len - 4) / 2);
- if (event < 10) {
- resp = '0' + event;
- } else if (event < 11) {
- resp = '*';
- } else if (event < 12) {
- resp = '#';
- } else if (event < 16) {
- resp = 'A' + (event - 12);
- } else if (event < 17) {
- resp = 'X';
- }
- if ((!rtp->resp && power) || (rtp->resp && (rtp->resp != resp))) {
- rtp->resp = resp;
- /* Why we should care on DTMF compensation at reception? */
- if (!ast_test_flag(rtp, FLAG_DTMF_COMPENSATE)) {
- f = send_dtmf(rtp, AST_FRAME_DTMF_BEGIN);
- rtp->dtmfsamples = 0;
- }
- } else if ((rtp->resp == resp) && !power) {
- f = send_dtmf(rtp, AST_FRAME_DTMF_END);
- f->samples = rtp->dtmfsamples * 8;
- rtp->resp = 0;
- } else if (rtp->resp == resp)
- rtp->dtmfsamples += 20 * 8;
- rtp->dtmfcount = dtmftimeout;
- return f;
-}
-
-/*!
- * \brief Process RTP DTMF and events according to RFC 2833.
- *
- * RFC 2833 is "RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals".
- *
- * \param rtp
- * \param data
- * \param len
- * \param seqno
- * \param timestamp
- * \returns
- */
-static struct ast_frame *process_rfc2833(struct ast_rtp *rtp, unsigned char *data, int len, unsigned int seqno, unsigned int timestamp)
-{
- unsigned int event;
- unsigned int event_end;
- unsigned int samples;
- char resp = 0;
- struct ast_frame *f = NULL;
-
- /* Figure out event, event end, and samples */
- event = ntohl(*((unsigned int *)(data)));
- event >>= 24;
- event_end = ntohl(*((unsigned int *)(data)));
- event_end <<= 8;
- event_end >>= 24;
- samples = ntohl(*((unsigned int *)(data)));
- samples &= 0xFFFF;
-
- /* Print out debug if turned on */
- if (rtpdebug || option_debug > 2)
- ast_debug(0, "- RTP 2833 Event: %08x (len = %d)\n", event, len);
-
- /* Figure out what digit was pressed */
- if (event < 10) {
- resp = '0' + event;
- } else if (event < 11) {
- resp = '*';
- } else if (event < 12) {
- resp = '#';
- } else if (event < 16) {
- resp = 'A' + (event - 12);
- } else if (event < 17) { /* Event 16: Hook flash */
- resp = 'X';
- } else {
- /* Not a supported event */
- ast_log(LOG_DEBUG, "Ignoring RTP 2833 Event: %08x. Not a DTMF Digit.\n", event);
- return &ast_null_frame;
- }
-
- if (ast_test_flag(rtp, FLAG_DTMF_COMPENSATE)) {
- if ((rtp->lastevent != timestamp) || (rtp->resp && rtp->resp != resp)) {
- rtp->resp = resp;
- rtp->dtmfcount = 0;
- f = send_dtmf(rtp, AST_FRAME_DTMF_END);
- f->len = 0;
- rtp->lastevent = timestamp;
- }
- } else {
- if ((!(rtp->resp) && (!(event_end & 0x80))) || (rtp->resp && rtp->resp != resp)) {
- rtp->resp = resp;
- f = send_dtmf(rtp, AST_FRAME_DTMF_BEGIN);
- rtp->dtmfcount = dtmftimeout;
- } else if ((event_end & 0x80) && (rtp->lastevent != seqno) && rtp->resp) {
- f = send_dtmf(rtp, AST_FRAME_DTMF_END);
- f->len = ast_tvdiff_ms(ast_samp2tv(samples, 8000), ast_tv(0, 0)); /* XXX hard coded 8kHz */
- rtp->resp = 0;
- rtp->dtmfcount = 0;
- rtp->lastevent = seqno;
- }
- }
-
- rtp->dtmfsamples = samples;
-
- return f;
-}
-
-/*!
- * \brief Process Comfort Noise RTP.
- *
- * This is incomplete at the moment.
- *
-*/
-static struct ast_frame *process_rfc3389(struct ast_rtp *rtp, unsigned char *data, int len)
-{
- struct ast_frame *f = NULL;
- /* Convert comfort noise into audio with various codecs. Unfortunately this doesn't
- totally help us out becuase we don't have an engine to keep it going and we are not
- guaranteed to have it every 20ms or anything */
- if (rtpdebug)
- ast_debug(0, "- RTP 3389 Comfort noise event: Level %d (len = %d)\n", rtp->lastrxformat, len);
-
- if (!(ast_test_flag(rtp, FLAG_3389_WARNING))) {
- ast_log(LOG_NOTICE, "Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: %s\n",
- ast_inet_ntoa(rtp->them.sin_addr));
- ast_set_flag(rtp, FLAG_3389_WARNING);
- }
-
- /* Must have at least one byte */
- if (!len)
- return NULL;
- if (len < 24) {
- rtp->f.data.ptr = rtp->rawdata + AST_FRIENDLY_OFFSET;
- rtp->f.datalen = len - 1;
- rtp->f.offset = AST_FRIENDLY_OFFSET;
- memcpy(rtp->f.data.ptr, data + 1, len - 1);
- } else {
- rtp->f.data.ptr = NULL;
- rtp->f.offset = 0;
- rtp->f.datalen = 0;
- }
- rtp->f.frametype = AST_FRAME_CNG;
- rtp->f.subclass = data[0] & 0x7f;
- rtp->f.datalen = len - 1;
- rtp->f.samples = 0;
- rtp->f.delivery.tv_usec = rtp->f.delivery.tv_sec = 0;
- f = &rtp->f;
- return f;
-}
-
-static int rtpread(int *id, int fd, short events, void *cbdata)
-{
- struct ast_rtp *rtp = cbdata;
- struct ast_frame *f;
- f = ast_rtp_read(rtp);
- if (f) {
- if (rtp->callback)
- rtp->callback(rtp, f, rtp->data);
- }
- return 1;
-}
-
-struct ast_frame *ast_rtcp_read(struct ast_rtp *rtp)
-{
- socklen_t len;
- int position, i, packetwords;
- int res;
- struct sockaddr_in sock_in;
- unsigned int rtcpdata[8192 + AST_FRIENDLY_OFFSET];
- unsigned int *rtcpheader;
- int pt;
- struct timeval now;
- unsigned int length;
- int rc;
- double rttsec;
- uint64_t rtt = 0;
- unsigned int dlsr;
- unsigned int lsr;
- unsigned int msw;
- unsigned int lsw;
- unsigned int comp;
- struct ast_frame *f = &ast_null_frame;
-
- double reported_jitter;
- double reported_normdev_jitter_current;
- double normdevrtt_current;
- double reported_lost;
- double reported_normdev_lost_current;
-
- if (!rtp || !rtp->rtcp)
- return &ast_null_frame;
-
- len = sizeof(sock_in);
-
- res = recvfrom(rtp->rtcp->s, rtcpdata + AST_FRIENDLY_OFFSET, sizeof(rtcpdata) - sizeof(unsigned int) * AST_FRIENDLY_OFFSET,
- 0, (struct sockaddr *)&sock_in, &len);
- rtcpheader = (unsigned int *)(rtcpdata + AST_FRIENDLY_OFFSET);
-
- if (res < 0) {
- ast_assert(errno != EBADF);
- if (errno != EAGAIN) {
- ast_log(LOG_WARNING, "RTCP Read error: %s. Hanging up.\n", strerror(errno));
- return NULL;
- }
- return &ast_null_frame;
- }
-
- packetwords = res / 4;
-
- if (rtp->nat) {
- /* Send to whoever sent to us */
- if ((rtp->rtcp->them.sin_addr.s_addr != sock_in.sin_addr.s_addr) ||
- (rtp->rtcp->them.sin_port != sock_in.sin_port)) {
- memcpy(&rtp->rtcp->them, &sock_in, sizeof(rtp->rtcp->them));
- if (option_debug || rtpdebug)
- ast_debug(0, "RTCP NAT: Got RTCP from other end. Now sending to address %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
- }
- }
-
- ast_debug(1, "Got RTCP report of %d bytes\n", res);
-
- /* Process a compound packet */
- position = 0;
- while (position < packetwords) {
- i = position;
- length = ntohl(rtcpheader[i]);
- pt = (length & 0xff0000) >> 16;
- rc = (length & 0x1f000000) >> 24;
- length &= 0xffff;
-
- if ((i + length) > packetwords) {
- if (option_debug || rtpdebug)
- ast_log(LOG_DEBUG, "RTCP Read too short\n");
- return &ast_null_frame;
- }
-
- if (rtcp_debug_test_addr(&sock_in)) {
- ast_verbose("\n\nGot RTCP from %s:%d\n", ast_inet_ntoa(sock_in.sin_addr), ntohs(sock_in.sin_port));
- ast_verbose("PT: %d(%s)\n", pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown");
- ast_verbose("Reception reports: %d\n", rc);
- ast_verbose("SSRC of sender: %u\n", rtcpheader[i + 1]);
- }
-
- i += 2; /* Advance past header and ssrc */
-
- switch (pt) {
- case RTCP_PT_SR:
- gettimeofday(&rtp->rtcp->rxlsr,NULL); /* To be able to populate the dlsr */
- rtp->rtcp->spc = ntohl(rtcpheader[i+3]);
- rtp->rtcp->soc = ntohl(rtcpheader[i + 4]);
- rtp->rtcp->themrxlsr = ((ntohl(rtcpheader[i]) & 0x0000ffff) << 16) | ((ntohl(rtcpheader[i + 1]) & 0xffff0000) >> 16); /* Going to LSR in RR*/
-
- if (rtcp_debug_test_addr(&sock_in)) {
- ast_verbose("NTP timestamp: %lu.%010lu\n", (unsigned long) ntohl(rtcpheader[i]), (unsigned long) ntohl(rtcpheader[i + 1]) * 4096);
- ast_verbose("RTP timestamp: %lu\n", (unsigned long) ntohl(rtcpheader[i + 2]));
- ast_verbose("SPC: %lu\tSOC: %lu\n", (unsigned long) ntohl(rtcpheader[i + 3]), (unsigned long) ntohl(rtcpheader[i + 4]));
- }
- i += 5;
- if (rc < 1)
- break;
- /* Intentional fall through */
- case RTCP_PT_RR:
- /* Don't handle multiple reception reports (rc > 1) yet */
- /* Calculate RTT per RFC */
- gettimeofday(&now, NULL);
- timeval2ntp(now, &msw, &lsw);
- if (ntohl(rtcpheader[i + 4]) && ntohl(rtcpheader[i + 5])) { /* We must have the LSR && DLSR */
- comp = ((msw & 0xffff) << 16) | ((lsw & 0xffff0000) >> 16);
- lsr = ntohl(rtcpheader[i + 4]);
- dlsr = ntohl(rtcpheader[i + 5]);
- rtt = comp - lsr - dlsr;
-
- /* Convert end to end delay to usec (keeping the calculation in 64bit space)
- sess->ee_delay = (eedelay * 1000) / 65536; */
- if (rtt < 4294) {
- rtt = (rtt * 1000000) >> 16;
- } else {
- rtt = (rtt * 1000) >> 16;
- rtt *= 1000;
- }
- rtt = rtt / 1000.;
- rttsec = rtt / 1000.;
- rtp->rtcp->rtt = rttsec;
-
- if (comp - dlsr >= lsr) {
- rtp->rtcp->accumulated_transit += rttsec;
-
- if (rtp->rtcp->rtt_count == 0)
- rtp->rtcp->minrtt = rttsec;
-
- if (rtp->rtcp->maxrtt<rttsec)
- rtp->rtcp->maxrtt = rttsec;
-
- if (rtp->rtcp->minrtt>rttsec)
- rtp->rtcp->minrtt = rttsec;
-
- normdevrtt_current = normdev_compute(rtp->rtcp->normdevrtt, rttsec, rtp->rtcp->rtt_count);
-
- rtp->rtcp->stdevrtt = stddev_compute(rtp->rtcp->stdevrtt, rttsec, rtp->rtcp->normdevrtt, normdevrtt_current, rtp->rtcp->rtt_count);
-
- rtp->rtcp->normdevrtt = normdevrtt_current;
-
- rtp->rtcp->rtt_count++;
- } else if (rtcp_debug_test_addr(&sock_in)) {
- ast_verbose("Internal RTCP NTP clock skew detected: "
- "lsr=%u, now=%u, dlsr=%u (%d:%03dms), "
- "diff=%d\n",
- lsr, comp, dlsr, dlsr / 65536,
- (dlsr % 65536) * 1000 / 65536,
- dlsr - (comp - lsr));
- }
- }
-
- rtp->rtcp->reported_jitter = ntohl(rtcpheader[i + 3]);
- reported_jitter = (double) rtp->rtcp->reported_jitter;
-
- if (rtp->rtcp->reported_jitter_count == 0)
- rtp->rtcp->reported_minjitter = reported_jitter;
-
- if (reported_jitter < rtp->rtcp->reported_minjitter)
- rtp->rtcp->reported_minjitter = reported_jitter;
-
- if (reported_jitter > rtp->rtcp->reported_maxjitter)
- rtp->rtcp->reported_maxjitter = reported_jitter;
-
- reported_normdev_jitter_current = normdev_compute(rtp->rtcp->reported_normdev_jitter, reported_jitter, rtp->rtcp->reported_jitter_count);
-
- rtp->rtcp->reported_stdev_jitter = stddev_compute(rtp->rtcp->reported_stdev_jitter, reported_jitter, rtp->rtcp->reported_normdev_jitter, reported_normdev_jitter_current, rtp->rtcp->reported_jitter_count);
-
- rtp->rtcp->reported_normdev_jitter = reported_normdev_jitter_current;
-
- rtp->rtcp->reported_lost = ntohl(rtcpheader[i + 1]) & 0xffffff;
-
- reported_lost = (double) rtp->rtcp->reported_lost;
-
- /* using same counter as for jitter */
- if (rtp->rtcp->reported_jitter_count == 0)
- rtp->rtcp->reported_minlost = reported_lost;
-
- if (reported_lost < rtp->rtcp->reported_minlost)
- rtp->rtcp->reported_minlost = reported_lost;
-
- if (reported_lost > rtp->rtcp->reported_maxlost)
- rtp->rtcp->reported_maxlost = reported_lost;
-
- reported_normdev_lost_current = normdev_compute(rtp->rtcp->reported_normdev_lost, reported_lost, rtp->rtcp->reported_jitter_count);
-
- rtp->rtcp->reported_stdev_lost = stddev_compute(rtp->rtcp->reported_stdev_lost, reported_lost, rtp->rtcp->reported_normdev_lost, reported_normdev_lost_current, rtp->rtcp->reported_jitter_count);
-
- rtp->rtcp->reported_normdev_lost = reported_normdev_lost_current;
-
- rtp->rtcp->reported_jitter_count++;
-
- if (rtcp_debug_test_addr(&sock_in)) {
- ast_verbose(" Fraction lost: %ld\n", (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24));
- ast_verbose(" Packets lost so far: %d\n", rtp->rtcp->reported_lost);
- ast_verbose(" Highest sequence number: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff));
- ast_verbose(" Sequence number cycles: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16);
- ast_verbose(" Interarrival jitter: %u\n", rtp->rtcp->reported_jitter);
- ast_verbose(" Last SR(our NTP): %lu.%010lu\n",(unsigned long) ntohl(rtcpheader[i + 4]) >> 16,((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096);
- ast_verbose(" DLSR: %4.4f (sec)\n",ntohl(rtcpheader[i + 5])/65536.0);
- if (rtt)
- ast_verbose(" RTT: %lu(sec)\n", (unsigned long) rtt);
- }
-
- if (rtt) {
- manager_event(EVENT_FLAG_REPORTING, "RTCPReceived", "From: %s:%d\r\n"
- "PT: %d(%s)\r\n"
- "ReceptionReports: %d\r\n"
- "SenderSSRC: %u\r\n"
- "FractionLost: %ld\r\n"
- "PacketsLost: %d\r\n"
- "HighestSequence: %ld\r\n"
- "SequenceNumberCycles: %ld\r\n"
- "IAJitter: %u\r\n"
- "LastSR: %lu.%010lu\r\n"
- "DLSR: %4.4f(sec)\r\n"
- "RTT: %llu(sec)\r\n",
- ast_inet_ntoa(sock_in.sin_addr), ntohs(sock_in.sin_port),
- pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown",
- rc,
- rtcpheader[i + 1],
- (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24),
- rtp->rtcp->reported_lost,
- (long) (ntohl(rtcpheader[i + 2]) & 0xffff),
- (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16,
- rtp->rtcp->reported_jitter,
- (unsigned long) ntohl(rtcpheader[i + 4]) >> 16, ((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096,
- ntohl(rtcpheader[i + 5])/65536.0,
- (unsigned long long)rtt);
- } else {
- manager_event(EVENT_FLAG_REPORTING, "RTCPReceived", "From: %s:%d\r\n"
- "PT: %d(%s)\r\n"
- "ReceptionReports: %d\r\n"
- "SenderSSRC: %u\r\n"
- "FractionLost: %ld\r\n"
- "PacketsLost: %d\r\n"
- "HighestSequence: %ld\r\n"
- "SequenceNumberCycles: %ld\r\n"
- "IAJitter: %u\r\n"
- "LastSR: %lu.%010lu\r\n"
- "DLSR: %4.4f(sec)\r\n",
- ast_inet_ntoa(sock_in.sin_addr), ntohs(sock_in.sin_port),
- pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown",
- rc,
- rtcpheader[i + 1],
- (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24),
- rtp->rtcp->reported_lost,
- (long) (ntohl(rtcpheader[i + 2]) & 0xffff),
- (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16,
- rtp->rtcp->reported_jitter,
- (unsigned long) ntohl(rtcpheader[i + 4]) >> 16,
- ((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096,
- ntohl(rtcpheader[i + 5])/65536.0);
- }
- break;
- case RTCP_PT_FUR:
- if (rtcp_debug_test_addr(&sock_in))
- ast_verbose("Received an RTCP Fast Update Request\n");
- rtp->f.frametype = AST_FRAME_CONTROL;
- rtp->f.subclass = AST_CONTROL_VIDUPDATE;
- rtp->f.datalen = 0;
- rtp->f.samples = 0;
- rtp->f.mallocd = 0;
- rtp->f.src = "RTP";
- f = &rtp->f;
- break;
- case RTCP_PT_SDES:
- if (rtcp_debug_test_addr(&sock_in))
- ast_verbose("Received an SDES from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
- break;
- case RTCP_PT_BYE:
- if (rtcp_debug_test_addr(&sock_in))
- ast_verbose("Received a BYE from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
- break;
- default:
- ast_debug(1, "Unknown RTCP packet (pt=%d) received from %s:%d\n", pt, ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
- break;
- }
- position += (length + 1);
- }
- rtp->rtcp->rtcp_info = 1;
- return f;
-}
-
-static void calc_rxstamp(struct timeval *when, struct ast_rtp *rtp, unsigned int timestamp, int mark)
-{
- struct timeval now;
- double transit;
- double current_time;
- double d;
- double dtv;
- double prog;
-
- double normdev_rxjitter_current;
- if ((!rtp->rxcore.tv_sec && !rtp->rxcore.tv_usec) || mark) {
- gettimeofday(&rtp->rxcore, NULL);
- rtp->drxcore = (double) rtp->rxcore.tv_sec + (double) rtp->rxcore.tv_usec / 1000000;
- /* map timestamp to a real time */
- rtp->seedrxts = timestamp; /* Their RTP timestamp started with this */
- rtp->rxcore.tv_sec -= timestamp / 8000;
- rtp->rxcore.tv_usec -= (timestamp % 8000) * 125;
- /* Round to 0.1ms for nice, pretty timestamps */
- rtp->rxcore.tv_usec -= rtp->rxcore.tv_usec % 100;
- if (rtp->rxcore.tv_usec < 0) {
- /* Adjust appropriately if necessary */
- rtp->rxcore.tv_usec += 1000000;
- rtp->rxcore.tv_sec -= 1;
- }
- }
-
- gettimeofday(&now,NULL);
- /* rxcore is the mapping between the RTP timestamp and _our_ real time from gettimeofday() */
- when->tv_sec = rtp->rxcore.tv_sec + timestamp / 8000;
- when->tv_usec = rtp->rxcore.tv_usec + (timestamp % 8000) * 125;
- if (when->tv_usec >= 1000000) {
- when->tv_usec -= 1000000;
- when->tv_sec += 1;
- }
- prog = (double)((timestamp-rtp->seedrxts)/8000.);
- dtv = (double)rtp->drxcore + (double)(prog);
- current_time = (double)now.tv_sec + (double)now.tv_usec/1000000;
- transit = current_time - dtv;
- d = transit - rtp->rxtransit;
- rtp->rxtransit = transit;
- if (d<0)
- d=-d;
- rtp->rxjitter += (1./16.) * (d - rtp->rxjitter);
- if (rtp->rtcp && rtp->rxjitter > rtp->rtcp->maxrxjitter)
- rtp->rtcp->maxrxjitter = rtp->rxjitter;
- if (rtp->rtcp->rxjitter_count == 1)
- rtp->rtcp->minrxjitter = rtp->rxjitter;
- if (rtp->rtcp && rtp->rxjitter < rtp->rtcp->minrxjitter)
- rtp->rtcp->minrxjitter = rtp->rxjitter;
-
- normdev_rxjitter_current = normdev_compute(rtp->rtcp->normdev_rxjitter,rtp->rxjitter,rtp->rtcp->rxjitter_count);
- rtp->rtcp->stdev_rxjitter = stddev_compute(rtp->rtcp->stdev_rxjitter,rtp->rxjitter,rtp->rtcp->normdev_rxjitter,normdev_rxjitter_current,rtp->rtcp->rxjitter_count);
-
- rtp->rtcp->normdev_rxjitter = normdev_rxjitter_current;
- rtp->rtcp->rxjitter_count++;
-}
-
-/*! \brief Perform a Packet2Packet RTP write */
-static int bridge_p2p_rtp_write(struct ast_rtp *rtp, struct ast_rtp *bridged, unsigned int *rtpheader, int len, int hdrlen)
-{
- int res = 0, payload = 0, bridged_payload = 0, mark;
- struct rtpPayloadType rtpPT;
- int reconstruct = ntohl(rtpheader[0]);
-
- /* Get fields from packet */
- payload = (reconstruct & 0x7f0000) >> 16;
- mark = (((reconstruct & 0x800000) >> 23) != 0);
-
- /* Check what the payload value should be */
- rtpPT = ast_rtp_lookup_pt(rtp, payload);
-
- /* If the payload is DTMF, and we are listening for DTMF - then feed it into the core */
- if (ast_test_flag(rtp, FLAG_P2P_NEED_DTMF) && !rtpPT.isAstFormat && rtpPT.code == AST_RTP_DTMF)
- return -1;
-
- /* Otherwise adjust bridged payload to match */
- bridged_payload = ast_rtp_lookup_code(bridged, rtpPT.isAstFormat, rtpPT.code);
-
- /* If the payload coming in is not one of the negotiated ones then send it to the core, this will cause formats to change and the bridge to break */
- if (!bridged->current_RTP_PT[bridged_payload].code)
- return -1;
-
-
- /* If the mark bit has not been sent yet... do it now */
- if (!ast_test_flag(rtp, FLAG_P2P_SENT_MARK)) {
- mark = 1;
- ast_set_flag(rtp, FLAG_P2P_SENT_MARK);
- }
-
- /* Reconstruct part of the packet */
- reconstruct &= 0xFF80FFFF;
- reconstruct |= (bridged_payload << 16);
- reconstruct |= (mark << 23);
- rtpheader[0] = htonl(reconstruct);
-
- /* Send the packet back out */
- res = sendto(bridged->s, (void *)rtpheader, len, 0, (struct sockaddr *)&bridged->them, sizeof(bridged->them));
- if (res < 0) {
- if (!bridged->nat || (bridged->nat && (ast_test_flag(bridged, FLAG_NAT_ACTIVE) == FLAG_NAT_ACTIVE))) {
- ast_debug(1, "RTP Transmission error of packet to %s:%d: %s\n", ast_inet_ntoa(bridged->them.sin_addr), ntohs(bridged->them.sin_port), strerror(errno));
- } else if (((ast_test_flag(bridged, FLAG_NAT_ACTIVE) == FLAG_NAT_INACTIVE) || rtpdebug) && !ast_test_flag(bridged, FLAG_NAT_INACTIVE_NOWARN)) {
- if (option_debug || rtpdebug)
- ast_debug(0, "RTP NAT: Can't write RTP to private address %s:%d, waiting for other end to send audio...\n", ast_inet_ntoa(bridged->them.sin_addr), ntohs(bridged->them.sin_port));
- ast_set_flag(bridged, FLAG_NAT_INACTIVE_NOWARN);
- }
- return 0;
- } else if (rtp_debug_test_addr(&bridged->them))
- ast_verbose("Sent RTP P2P packet to %s:%u (type %-2.2d, len %-6.6u)\n", ast_inet_ntoa(bridged->them.sin_addr), ntohs(bridged->them.sin_port), bridged_payload, len - hdrlen);
-
- return 0;
-}
-
-struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
-{
- int res;
- struct sockaddr_in sock_in;
- socklen_t len;
- unsigned int seqno;
- int version;
- int payloadtype;
- int hdrlen = 12;
- int padding;
- int mark;
- int ext;
- int cc;
- unsigned int ssrc;
- unsigned int timestamp;
- unsigned int *rtpheader;
- struct rtpPayloadType rtpPT;
- struct ast_rtp *bridged = NULL;
- int prev_seqno;
-
- /* If time is up, kill it */
- if (rtp->sending_digit)
- ast_rtp_senddigit_continuation(rtp);
-
- len = sizeof(sock_in);
-
- /* Cache where the header will go */
- res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET,
- 0, (struct sockaddr *)&sock_in, &len);
-
- /* If strict RTP protection is enabled see if we need to learn this address or if the packet should be dropped */
- if (rtp->strict_rtp_state == STRICT_RTP_LEARN) {
- /* Copy over address that this packet was received on */
- memcpy(&rtp->strict_rtp_address, &sock_in, sizeof(rtp->strict_rtp_address));
- /* Now move over to actually protecting the RTP port */
- rtp->strict_rtp_state = STRICT_RTP_CLOSED;
- ast_debug(1, "Learned remote address is %s:%d for strict RTP purposes, now protecting the port.\n", ast_inet_ntoa(rtp->strict_rtp_address.sin_addr), ntohs(rtp->strict_rtp_address.sin_port));
- } else if (rtp->strict_rtp_state == STRICT_RTP_CLOSED) {
- /* If the address we previously learned doesn't match the address this packet came in on simply drop it */
- if ((rtp->strict_rtp_address.sin_addr.s_addr != sock_in.sin_addr.s_addr) || (rtp->strict_rtp_address.sin_port != sock_in.sin_port)) {
- ast_debug(1, "Received RTP packet from %s:%d, dropping due to strict RTP protection. Expected it to be from %s:%d\n", ast_inet_ntoa(sock_in.sin_addr), ntohs(sock_in.sin_port), ast_inet_ntoa(rtp->strict_rtp_address.sin_addr), ntohs(rtp->strict_rtp_address.sin_port));
- return &ast_null_frame;
- }
- }
-
- rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET);
- if (res < 0) {
- ast_assert(errno != EBADF);
- if (errno != EAGAIN) {
- ast_log(LOG_WARNING, "RTP Read error: %s. Hanging up.\n", strerror(errno));
- return NULL;
- }
- return &ast_null_frame;
- }
-
- if (res < hdrlen) {
- ast_log(LOG_WARNING, "RTP Read too short\n");
- return &ast_null_frame;
- }
-
- /* Get fields */
- seqno = ntohl(rtpheader[0]);
-
- /* Check RTP version */
- version = (seqno & 0xC0000000) >> 30;
- if (!version) {
- /* If the two high bits are 0, this might be a
- * STUN message, so process it. stun_handle_packet()
- * answers to requests, and it returns STUN_ACCEPT
- * if the request is valid.
- */
- if ((stun_handle_packet(rtp->s, &sock_in, rtp->rawdata + AST_FRIENDLY_OFFSET, res, NULL, NULL) == STUN_ACCEPT) &&
- (!rtp->them.sin_port && !rtp->them.sin_addr.s_addr)) {
- memcpy(&rtp->them, &sock_in, sizeof(rtp->them));
- }
- return &ast_null_frame;
- }
-
-#if 0 /* Allow to receive RTP stream with closed transmission path */
- /* If we don't have the other side's address, then ignore this */
- if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
- return &ast_null_frame;
-#endif
-
- /* Send to whoever send to us if NAT is turned on */
- if (rtp->nat) {
- if ((rtp->them.sin_addr.s_addr != sock_in.sin_addr.s_addr) ||
- (rtp->them.sin_port != sock_in.sin_port)) {
- rtp->them = sock_in;
- if (rtp->rtcp) {
- int h = 0;
- memcpy(&rtp->rtcp->them, &sock_in, sizeof(rtp->rtcp->them));
- h = ntohs(rtp->them.sin_port);
- rtp->rtcp->them.sin_port = htons(h + 1);
- }
- rtp->rxseqno = 0;
- ast_set_flag(rtp, FLAG_NAT_ACTIVE);
- if (option_debug || rtpdebug)
- ast_debug(0, "RTP NAT: Got audio from other end. Now sending to address %s:%d\n", ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
- }
- }
-
- /* If we are bridged to another RTP stream, send direct */
- if ((bridged = ast_rtp_get_bridged(rtp)) && !bridge_p2p_rtp_write(rtp, bridged, rtpheader, res, hdrlen))
- return &ast_null_frame;
-
- if (version != 2)
- return &ast_null_frame;
-
- payloadtype = (seqno & 0x7f0000) >> 16;
- padding = seqno & (1 << 29);
- mark = seqno & (1 << 23);
- ext = seqno & (1 << 28);
- cc = (seqno & 0xF000000) >> 24;
- seqno &= 0xffff;
- timestamp = ntohl(rtpheader[1]);
- ssrc = ntohl(rtpheader[2]);
-
- if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) {
- if (option_debug || rtpdebug)
- ast_debug(0, "Forcing Marker bit, because SSRC has changed\n");
- mark = 1;
- }
-
- rtp->rxssrc = ssrc;
-
- if (padding) {
- /* Remove padding bytes */
- res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1];
- }
-
- if (cc) {
- /* CSRC fields present */
- hdrlen += cc*4;
- }
-
- if (ext) {
- /* RTP Extension present */
- hdrlen += (ntohl(rtpheader[hdrlen/4]) & 0xffff) << 2;
- hdrlen += 4;
- if (option_debug) {
- int profile;
- profile = (ntohl(rtpheader[3]) & 0xffff0000) >> 16;
- if (profile == 0x505a)
- ast_debug(1, "Found Zfone extension in RTP stream - zrtp - not supported.\n");
- else
- ast_debug(1, "Found unknown RTP Extensions %x\n", profile);
- }
- }
-
- if (res < hdrlen) {
- ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen);
- return &ast_null_frame;
- }
-
- rtp->rxcount++; /* Only count reasonably valid packets, this'll make the rtcp stats more accurate */
-
- if (rtp->rxcount==1) {
- /* This is the first RTP packet successfully received from source */
- rtp->seedrxseqno = seqno;
- }
-
- /* Do not schedule RR if RTCP isn't run */
- if (rtp->rtcp && rtp->rtcp->them.sin_addr.s_addr && rtp->rtcp->schedid < 1) {
- /* Schedule transmission of Receiver Report */
- rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp);
- }
- if ((int)rtp->lastrxseqno - (int)seqno > 100) /* if so it would indicate that the sender cycled; allow for misordering */
- rtp->cycles += RTP_SEQ_MOD;
-
- prev_seqno = rtp->lastrxseqno;
-
- rtp->lastrxseqno = seqno;
-
- if (!rtp->themssrc)
- rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */
-
- if (rtp_debug_test_addr(&sock_in))
- ast_verbose("Got RTP packet from %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
- ast_inet_ntoa(sock_in.sin_addr), ntohs(sock_in.sin_port), payloadtype, seqno, timestamp,res - hdrlen);
-
- rtpPT = ast_rtp_lookup_pt(rtp, payloadtype);
- if (!rtpPT.isAstFormat) {
- struct ast_frame *f = NULL;
-
- /* This is special in-band data that's not one of our codecs */
- if (rtpPT.code == AST_RTP_DTMF) {
- /* It's special -- rfc2833 process it */
- if (rtp_debug_test_addr(&sock_in)) {
- unsigned char *data;
- unsigned int event;
- unsigned int event_end;
- unsigned int duration;
- data = rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen;
- event = ntohl(*((unsigned int *)(data)));
- event >>= 24;
- event_end = ntohl(*((unsigned int *)(data)));
- event_end <<= 8;
- event_end >>= 24;
- duration = ntohl(*((unsigned int *)(data)));
- duration &= 0xFFFF;
- ast_verbose("Got RTP RFC2833 from %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u, mark %d, event %08x, end %d, duration %-5.5d) \n", ast_inet_ntoa(sock_in.sin_addr), ntohs(sock_in.sin_port), payloadtype, seqno, timestamp, res - hdrlen, (mark?1:0), event, ((event_end & 0x80)?1:0), duration);
- }
- f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp);
- } else if (rtpPT.code == AST_RTP_CISCO_DTMF) {
- /* It's really special -- process it the Cisco way */
- if (rtp->lastevent <= seqno || (rtp->lastevent >= 65530 && seqno <= 6)) {
- f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
- rtp->lastevent = seqno;
- }
- } else if (rtpPT.code == AST_RTP_CN) {
- /* Comfort Noise */
- f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
- } else {
- ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(rtp->them.sin_addr));
- }
- return f ? f : &ast_null_frame;
- }
- rtp->lastrxformat = rtp->f.subclass = rtpPT.code;
- rtp->f.frametype = (rtp->f.subclass & AST_FORMAT_AUDIO_MASK) ? AST_FRAME_VOICE : (rtp->f.subclass & AST_FORMAT_VIDEO_MASK) ? AST_FRAME_VIDEO : AST_FRAME_TEXT;
-
- rtp->rxseqno = seqno;
-
- if (rtp->dtmfcount) {
- rtp->dtmfcount -= (timestamp - rtp->lastrxts);
-
- if (rtp->dtmfcount < 0) {
- rtp->dtmfcount = 0;
- }
-
- if (rtp->resp && !rtp->dtmfcount) {
- struct ast_frame *f;
- f = send_dtmf(rtp, AST_FRAME_DTMF_END);
- rtp->resp = 0;
- return f;
- }
- }
-
- /* Record received timestamp as last received now */
- rtp->lastrxts = timestamp;
-
- rtp->f.mallocd = 0;
- rtp->f.datalen = res - hdrlen;
- rtp->f.data.ptr = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET;
- rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
- rtp->f.seqno = seqno;
-
- if (rtp->f.subclass == AST_FORMAT_T140 && (int)seqno - (prev_seqno+1) > 0 && (int)seqno - (prev_seqno+1) < 10) {
- unsigned char *data = rtp->f.data.ptr;
-
- memmove(rtp->f.data.ptr+3, rtp->f.data.ptr, rtp->f.datalen);
- rtp->f.datalen +=3;
- *data++ = 0xEF;
- *data++ = 0xBF;
- *data = 0xBD;
- }
-
- if (rtp->f.subclass == AST_FORMAT_T140RED) {
- unsigned char *data = rtp->f.data.ptr;
- unsigned char *header_end;
- int num_generations;
- int header_length;
- int length;
- int diff =(int)seqno - (prev_seqno+1); /* if diff = 0, no drop*/
- int x;
-
- rtp->f.subclass = AST_FORMAT_T140;
- header_end = memchr(data, ((*data) & 0x7f), rtp->f.datalen);
- header_end++;
-
- header_length = header_end - data;
- num_generations = header_length / 4;
- length = header_length;
-
- if (!diff) {
- for (x = 0; x < num_generations; x++)
- length += data[x * 4 + 3];
-
- if (!(rtp->f.datalen - length))
- return &ast_null_frame;
-
- rtp->f.data.ptr += length;
- rtp->f.datalen -= length;
- } else if (diff > num_generations && diff < 10) {
- length -= 3;
- rtp->f.data.ptr += length;
- rtp->f.datalen -= length;
-
- data = rtp->f.data.ptr;
- *data++ = 0xEF;
- *data++ = 0xBF;
- *data = 0xBD;
- } else {
- for ( x = 0; x < num_generations - diff; x++)
- length += data[x * 4 + 3];
-
- rtp->f.data.ptr += length;
- rtp->f.datalen -= length;
- }
- }
-
- if (rtp->f.subclass & AST_FORMAT_AUDIO_MASK) {
- rtp->f.samples = ast_codec_get_samples(&rtp->f);
- if (rtp->f.subclass == AST_FORMAT_SLINEAR)
- ast_frame_byteswap_be(&rtp->f);
- calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
- /* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
- ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);
- rtp->f.ts = timestamp / 8;
- rtp->f.len = rtp->f.samples / ((ast_format_rate(rtp->f.subclass) / 1000));
- } else if (rtp->f.subclass & AST_FORMAT_VIDEO_MASK) {
- /* Video -- samples is # of samples vs. 90000 */
- if (!rtp->lastividtimestamp)
- rtp->lastividtimestamp = timestamp;
- rtp->f.samples = timestamp - rtp->lastividtimestamp;
- rtp->lastividtimestamp = timestamp;
- rtp->f.delivery.tv_sec = 0;
- rtp->f.delivery.tv_usec = 0;
- /* Pass the RTP marker bit as bit 0 in the subclass field.
- * This is ok because subclass is actually a bitmask, and
- * the low bits represent audio formats, that are not
- * involved here since we deal with video.
- */
- if (mark)
- rtp->f.subclass |= 0x1;
- } else {
- /* TEXT -- samples is # of samples vs. 1000 */
- if (!rtp->lastitexttimestamp)
- rtp->lastitexttimestamp = timestamp;
- rtp->f.samples = timestamp - rtp->lastitexttimestamp;
- rtp->lastitexttimestamp = timestamp;
- rtp->f.delivery.tv_sec = 0;
- rtp->f.delivery.tv_usec = 0;
- }
- rtp->f.src = "RTP";
- return &rtp->f;
-}
-
-/* The following array defines the MIME Media type (and subtype) for each
- of our codecs, or RTP-specific data type. */
-static const struct mimeType {
- struct rtpPayloadType payloadType;
- char *type;
- char *subtype;
- unsigned int sample_rate;
-} mimeTypes[] = {
- {{1, AST_FORMAT_G723_1}, "audio", "G723", 8000},
- {{1, AST_FORMAT_GSM}, "audio", "GSM", 8000},
- {{1, AST_FORMAT_ULAW}, "audio", "PCMU", 8000},
- {{1, AST_FORMAT_ULAW}, "audio", "G711U", 8000},
- {{1, AST_FORMAT_ALAW}, "audio", "PCMA", 8000},
- {{1, AST_FORMAT_ALAW}, "audio", "G711A", 8000},
- {{1, AST_FORMAT_G726}, "audio", "G726-32", 8000},
- {{1, AST_FORMAT_ADPCM}, "audio", "DVI4", 8000},
- {{1, AST_FORMAT_SLINEAR}, "audio", "L16", 8000},
- {{1, AST_FORMAT_LPC10}, "audio", "LPC", 8000},
- {{1, AST_FORMAT_G729A}, "audio", "G729", 8000},
- {{1, AST_FORMAT_G729A}, "audio", "G729A", 8000},
- {{1, AST_FORMAT_G729A}, "audio", "G.729", 8000},
- {{1, AST_FORMAT_SPEEX}, "audio", "speex", 8000},
- {{1, AST_FORMAT_ILBC}, "audio", "iLBC", 8000},
- /* this is the sample rate listed in the RTP profile for the G.722
- codec, *NOT* the actual sample rate of the media stream
- */
- {{1, AST_FORMAT_G722}, "audio", "G722", 8000},
- {{1, AST_FORMAT_G726_AAL2}, "audio", "AAL2-G726-32", 8000},
- {{0, AST_RTP_DTMF}, "audio", "telephone-event", 8000},
- {{0, AST_RTP_CISCO_DTMF}, "audio", "cisco-telephone-event", 8000},
- {{0, AST_RTP_CN}, "audio", "CN", 8000},
- {{1, AST_FORMAT_JPEG}, "video", "JPEG", 90000},
- {{1, AST_FORMAT_PNG}, "video", "PNG", 90000},
- {{1, AST_FORMAT_H261}, "video", "H261", 90000},
- {{1, AST_FORMAT_H263}, "video", "H263", 90000},
- {{1, AST_FORMAT_H263_PLUS}, "video", "h263-1998", 90000},
- {{1, AST_FORMAT_H264}, "video", "H264", 90000},
- {{1, AST_FORMAT_MP4_VIDEO}, "video", "MP4V-ES", 90000},
- {{1, AST_FORMAT_T140RED}, "text", "RED", 1000},
- {{1, AST_FORMAT_T140}, "text", "T140", 1000},
- {{1, AST_FORMAT_SIREN7}, "audio", "G7221", 16000},
- {{1, AST_FORMAT_SIREN14}, "audio", "G7221", 32000},
-};
-
-/*!
- * \brief Mapping between Asterisk codecs and rtp payload types
- *
- * Static (i.e., well-known) RTP payload types for our "AST_FORMAT..."s:
- * also, our own choices for dynamic payload types. This is our master
- * table for transmission
- *
- * See http://www.iana.org/assignments/rtp-parameters for a list of
- * assigned values
- */
-static const struct rtpPayloadType static_RTP_PT[MAX_RTP_PT] = {
- [0] = {1, AST_FORMAT_ULAW},
-#ifdef USE_DEPRECATED_G726
- [2] = {1, AST_FORMAT_G726}, /* Technically this is G.721, but if Cisco can do it, so can we... */
-#endif
- [3] = {1, AST_FORMAT_GSM},
- [4] = {1, AST_FORMAT_G723_1},
- [5] = {1, AST_FORMAT_ADPCM}, /* 8 kHz */
- [6] = {1, AST_FORMAT_ADPCM}, /* 16 kHz */
- [7] = {1, AST_FORMAT_LPC10},
- [8] = {1, AST_FORMAT_ALAW},
- [9] = {1, AST_FORMAT_G722},
- [10] = {1, AST_FORMAT_SLINEAR}, /* 2 channels */
- [11] = {1, AST_FORMAT_SLINEAR}, /* 1 channel */
- [13] = {0, AST_RTP_CN},
- [16] = {1, AST_FORMAT_ADPCM}, /* 11.025 kHz */
- [17] = {1, AST_FORMAT_ADPCM}, /* 22.050 kHz */
- [18] = {1, AST_FORMAT_G729A},
- [19] = {0, AST_RTP_CN}, /* Also used for CN */
- [26] = {1, AST_FORMAT_JPEG},
- [31] = {1, AST_FORMAT_H261},
- [34] = {1, AST_FORMAT_H263},
- [97] = {1, AST_FORMAT_ILBC},
- [98] = {1, AST_FORMAT_H263_PLUS},
- [99] = {1, AST_FORMAT_H264},
- [101] = {0, AST_RTP_DTMF},
- [102] = {1, AST_FORMAT_SIREN7},
- [103] = {1, AST_FORMAT_H263_PLUS},
- [104] = {1, AST_FORMAT_MP4_VIDEO},
- [105] = {1, AST_FORMAT_T140RED}, /* Real time text chat (with redundancy encoding) */
- [106] = {1, AST_FORMAT_T140}, /* Real time text chat */
- [110] = {1, AST_FORMAT_SPEEX},
- [111] = {1, AST_FORMAT_G726},
- [112] = {1, AST_FORMAT_G726_AAL2},
- [115] = {1, AST_FORMAT_SIREN14},
- [121] = {0, AST_RTP_CISCO_DTMF}, /* Must be type 121 */
-};
-
-void ast_rtp_pt_clear(struct ast_rtp* rtp)
-{
- int i;
-
- if (!rtp)
- return;
-
- rtp_bridge_lock(rtp);
-
- for (i = 0; i < MAX_RTP_PT; ++i) {
- rtp->current_RTP_PT[i].isAstFormat = 0;
- rtp->current_RTP_PT[i].code = 0;
- }
-
- rtp->rtp_lookup_code_cache_isAstFormat = 0;
- rtp->rtp_lookup_code_cache_code = 0;
- rtp->rtp_lookup_code_cache_result = 0;
-
- rtp_bridge_unlock(rtp);
-}
-
-void ast_rtp_pt_default(struct ast_rtp* rtp)
-{
- int i;
-
- rtp_bridge_lock(rtp);
-
- /* Initialize to default payload types */
- for (i = 0; i < MAX_RTP_PT; ++i) {
- rtp->current_RTP_PT[i].isAstFormat = static_RTP_PT[i].isAstFormat;
- rtp->current_RTP_PT[i].code = static_RTP_PT[i].code;
- }
-
- rtp->rtp_lookup_code_cache_isAstFormat = 0;
- rtp->rtp_lookup_code_cache_code = 0;
- rtp->rtp_lookup_code_cache_result = 0;
-
- rtp_bridge_unlock(rtp);
-}
-
-void ast_rtp_pt_copy(struct ast_rtp *dest, struct ast_rtp *src)
-{
- unsigned int i;
-
- rtp_bridge_lock(dest);
- rtp_bridge_lock(src);
-
- for (i = 0; i < MAX_RTP_PT; ++i) {
- dest->current_RTP_PT[i].isAstFormat =
- src->current_RTP_PT[i].isAstFormat;
- dest->current_RTP_PT[i].code =
- src->current_RTP_PT[i].code;
- }
- dest->rtp_lookup_code_cache_isAstFormat = 0;
- dest->rtp_lookup_code_cache_code = 0;
- dest->rtp_lookup_code_cache_result = 0;
-
- rtp_bridge_unlock(src);
- rtp_bridge_unlock(dest);
-}
-
-/*! \brief Get channel driver interface structure */
-static struct ast_rtp_protocol *get_proto(struct ast_channel *chan)
-{
- struct ast_rtp_protocol *cur = NULL;
-
- AST_RWLIST_RDLOCK(&protos);
- AST_RWLIST_TRAVERSE(&protos, cur, list) {
- if (cur->type == chan->tech->type)
- break;
- }
- AST_RWLIST_UNLOCK(&protos);
-
- return cur;
-}
-
-int ast_rtp_early_bridge(struct ast_channel *c0, struct ast_channel *c1)
-{
- struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */
- struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */
- struct ast_rtp *tdestp = NULL, *tsrcp = NULL; /* Text RTP channels */
- struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
- enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED, text_dest_res = AST_RTP_GET_FAILED;
- enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED, text_src_res = AST_RTP_GET_FAILED;
- int srccodec, destcodec, nat_active = 0;
-
- /* Lock channels */
- ast_channel_lock(c0);
- if (c1) {
- while (ast_channel_trylock(c1)) {
- ast_channel_unlock(c0);
- usleep(1);
- ast_channel_lock(c0);
- }
- }
-
- /* Find channel driver interfaces */
- destpr = get_proto(c0);
- if (c1)
- srcpr = get_proto(c1);
- if (!destpr) {
- ast_debug(1, "Channel '%s' has no RTP, not doing anything\n", c0->name);
- ast_channel_unlock(c0);
- if (c1)
- ast_channel_unlock(c1);
- return -1;
- }
- if (!srcpr) {
- ast_debug(1, "Channel '%s' has no RTP, not doing anything\n", c1 ? c1->name : "<unspecified>");
- ast_channel_unlock(c0);
- if (c1)
- ast_channel_unlock(c1);
- return -1;
- }
-
- /* Get audio, video and text interface (if native bridge is possible) */
- audio_dest_res = destpr->get_rtp_info(c0, &destp);
- video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(c0, &vdestp) : AST_RTP_GET_FAILED;
- text_dest_res = destpr->get_trtp_info ? destpr->get_trtp_info(c0, &tdestp) : AST_RTP_GET_FAILED;
- if (srcpr) {
- audio_src_res = srcpr->get_rtp_info(c1, &srcp);
- video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(c1, &vsrcp) : AST_RTP_GET_FAILED;
- text_src_res = srcpr->get_trtp_info ? srcpr->get_trtp_info(c1, &tsrcp) : AST_RTP_GET_FAILED;
- }
-
- /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
- if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE)) {
- /* Somebody doesn't want to play... */
- ast_channel_unlock(c0);
- if (c1)
- ast_channel_unlock(c1);
- return -1;
- }
- if (audio_src_res == AST_RTP_TRY_NATIVE && (video_src_res == AST_RTP_GET_FAILED || video_src_res == AST_RTP_TRY_NATIVE) && srcpr->get_codec)
- srccodec = srcpr->get_codec(c1);
- else
- srccodec = 0;
- if (audio_dest_res == AST_RTP_TRY_NATIVE && (video_dest_res == AST_RTP_GET_FAILED || video_dest_res == AST_RTP_TRY_NATIVE) && destpr->get_codec)
- destcodec = destpr->get_codec(c0);
- else
- destcodec = 0;
- /* Ensure we have at least one matching codec */
- if (srcp && !(srccodec & destcodec)) {
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
- return 0;
- }
- /* Consider empty media as non-existent */
- if (audio_src_res == AST_RTP_TRY_NATIVE && !srcp->them.sin_addr.s_addr)
- srcp = NULL;
- if (srcp && (srcp->nat || ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
- nat_active = 1;
- /* Bridge media early */
- if (destpr->set_rtp_peer(c0, srcp, vsrcp, tsrcp, srccodec, nat_active))
- ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", c0->name, c1 ? c1->name : "<unspecified>");
- ast_channel_unlock(c0);
- if (c1)
- ast_channel_unlock(c1);
- ast_debug(1, "Setting early bridge SDP of '%s' with that of '%s'\n", c0->name, c1 ? c1->name : "<unspecified>");
- return 0;
-}
-
-int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media)
-{
- struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */
- struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */
- struct ast_rtp *tdestp = NULL, *tsrcp = NULL; /* Text RTP channels */
- struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
- enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED, text_dest_res = AST_RTP_GET_FAILED;
- enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED, text_src_res = AST_RTP_GET_FAILED;
- int srccodec, destcodec;
-
- /* Lock channels */
- ast_channel_lock(dest);
- while (ast_channel_trylock(src)) {
- ast_channel_unlock(dest);
- usleep(1);
- ast_channel_lock(dest);
- }
-
- /* Find channel driver interfaces */
- if (!(destpr = get_proto(dest))) {
- ast_debug(1, "Channel '%s' has no RTP, not doing anything\n", dest->name);
- ast_channel_unlock(dest);
- ast_channel_unlock(src);
- return 0;
- }
- if (!(srcpr = get_proto(src))) {
- ast_debug(1, "Channel '%s' has no RTP, not doing anything\n", src->name);
- ast_channel_unlock(dest);
- ast_channel_unlock(src);
- return 0;
- }
-
- /* Get audio and video interface (if native bridge is possible) */
- audio_dest_res = destpr->get_rtp_info(dest, &destp);
- video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED;
- text_dest_res = destpr->get_trtp_info ? destpr->get_trtp_info(dest, &tdestp) : AST_RTP_GET_FAILED;
- audio_src_res = srcpr->get_rtp_info(src, &srcp);
- video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED;
- text_src_res = srcpr->get_trtp_info ? srcpr->get_trtp_info(src, &tsrcp) : AST_RTP_GET_FAILED;
-
- /* Ensure we have at least one matching codec */
- if (srcpr->get_codec)
- srccodec = srcpr->get_codec(src);
- else
- srccodec = 0;
- if (destpr->get_codec)
- destcodec = destpr->get_codec(dest);
- else
- destcodec = 0;
-
- /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
- if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE) || audio_src_res != AST_RTP_TRY_NATIVE || (video_src_res != AST_RTP_GET_FAILED && video_src_res != AST_RTP_TRY_NATIVE) || !(srccodec & destcodec)) {
- /* Somebody doesn't want to play... */
- ast_channel_unlock(dest);
- ast_channel_unlock(src);
- return 0;
- }
- ast_rtp_pt_copy(destp, srcp);
- if (vdestp && vsrcp)
- ast_rtp_pt_copy(vdestp, vsrcp);
- if (tdestp && tsrcp)
- ast_rtp_pt_copy(tdestp, tsrcp);
- if (media) {
- /* Bridge early */
- if (destpr->set_rtp_peer(dest, srcp, vsrcp, tsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
- ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src->name);
- }
- ast_channel_unlock(dest);
- ast_channel_unlock(src);
- ast_debug(1, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name);
- return 1;
-}
-
-/*! \brief Make a note of a RTP payload type that was seen in a SDP "m=" line.
- * By default, use the well-known value for this type (although it may
- * still be set to a different value by a subsequent "a=rtpmap:" line)
- */
-void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt)
-{
- if (pt < 0 || pt > MAX_RTP_PT || static_RTP_PT[pt].code == 0)
- return; /* bogus payload type */
-
- rtp_bridge_lock(rtp);
- rtp->current_RTP_PT[pt] = static_RTP_PT[pt];
- rtp_bridge_unlock(rtp);
-}
-
-/*! \brief remove setting from payload type list if the rtpmap header indicates
- an unknown media type */
-void ast_rtp_unset_m_type(struct ast_rtp* rtp, int pt)
-{
- if (pt < 0 || pt > MAX_RTP_PT)
- return; /* bogus payload type */
-
- rtp_bridge_lock(rtp);
- rtp->current_RTP_PT[pt].isAstFormat = 0;
- rtp->current_RTP_PT[pt].code = 0;
- rtp_bridge_unlock(rtp);
-}
-
-/*! \brief Make a note of a RTP payload type (with MIME type) that was seen in
- * an SDP "a=rtpmap:" line.
- * \return 0 if the MIME type was found and set, -1 if it wasn't found
- */
-int ast_rtp_set_rtpmap_type_rate(struct ast_rtp *rtp, int pt,
- char *mimeType, char *mimeSubtype,
- enum ast_rtp_options options,
- unsigned int sample_rate)
-{
- unsigned int i;
- int found = 0;
-
- if (pt < 0 || pt > MAX_RTP_PT)
- return -1; /* bogus payload type */
-
- rtp_bridge_lock(rtp);
-
- for (i = 0; i < ARRAY_LEN(mimeTypes); ++i) {
- const struct mimeType *t = &mimeTypes[i];
-
- if (strcasecmp(mimeSubtype, t->subtype)) {
- continue;
- }
-
- if (strcasecmp(mimeType, t->type)) {
- continue;
- }
-
- /* if both sample rates have been supplied, and they don't match,
- then this not a match; if one has not been supplied, then the
- rates are not compared */
- if (sample_rate && t->sample_rate &&
- (sample_rate != t->sample_rate)) {
- continue;
- }
-
- found = 1;
- rtp->current_RTP_PT[pt] = t->payloadType;
-
- if ((t->payloadType.code == AST_FORMAT_G726) &&
- t->payloadType.isAstFormat &&
- (options & AST_RTP_OPT_G726_NONSTANDARD)) {
- rtp->current_RTP_PT[pt].code = AST_FORMAT_G726_AAL2;
- }
-
- break;
- }
-
- rtp_bridge_unlock(rtp);
-
- return (found ? 0 : -2);
-}
-
-int ast_rtp_set_rtpmap_type(struct ast_rtp *rtp, int pt,
- char *mimeType, char *mimeSubtype,
- enum ast_rtp_options options)
-{
- return ast_rtp_set_rtpmap_type_rate(rtp, pt, mimeType, mimeSubtype, options, 0);
-}
-
-/*! \brief Return the union of all of the codecs that were set by rtp_set...() calls
- * They're returned as two distinct sets: AST_FORMATs, and AST_RTPs */
-void ast_rtp_get_current_formats(struct ast_rtp* rtp,
- int* astFormats, int* nonAstFormats)
-{
- int pt;
-
- rtp_bridge_lock(rtp);
-
- *astFormats = *nonAstFormats = 0;
- for (pt = 0; pt < MAX_RTP_PT; ++pt) {
- if (rtp->current_RTP_PT[pt].isAstFormat) {
- *astFormats |= rtp->current_RTP_PT[pt].code;
- } else {
- *nonAstFormats |= rtp->current_RTP_PT[pt].code;
- }
- }
-
- rtp_bridge_unlock(rtp);
-}
-
-struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt)
-{
- struct rtpPayloadType result;
-
- result.isAstFormat = result.code = 0;
-
- if (pt < 0 || pt > MAX_RTP_PT)
- return result; /* bogus payload type */
-
- /* Start with negotiated codecs */
- rtp_bridge_lock(rtp);
- result = rtp->current_RTP_PT[pt];
- rtp_bridge_unlock(rtp);
-
- /* If it doesn't exist, check our static RTP type list, just in case */
- if (!result.code)
- result = static_RTP_PT[pt];
-
- return result;
-}
-
-/*! \brief Looks up an RTP code out of our *static* outbound list */
-int ast_rtp_lookup_code(struct ast_rtp* rtp, const int isAstFormat, const int code)
-{
- int pt = 0;
-
- rtp_bridge_lock(rtp);
-
- if (isAstFormat == rtp->rtp_lookup_code_cache_isAstFormat &&
- code == rtp->rtp_lookup_code_cache_code) {
- /* Use our cached mapping, to avoid the overhead of the loop below */
- pt = rtp->rtp_lookup_code_cache_result;
- rtp_bridge_unlock(rtp);
- return pt;
- }
-
- /* Check the dynamic list first */
- for (pt = 0; pt < MAX_RTP_PT; ++pt) {
- if (rtp->current_RTP_PT[pt].code == code && rtp->current_RTP_PT[pt].isAstFormat == isAstFormat) {
- rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
- rtp->rtp_lookup_code_cache_code = code;
- rtp->rtp_lookup_code_cache_result = pt;
- rtp_bridge_unlock(rtp);
- return pt;
- }
- }
-
- /* Then the static list */
- for (pt = 0; pt < MAX_RTP_PT; ++pt) {
- if (static_RTP_PT[pt].code == code && static_RTP_PT[pt].isAstFormat == isAstFormat) {
- rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
- rtp->rtp_lookup_code_cache_code = code;
- rtp->rtp_lookup_code_cache_result = pt;
- rtp_bridge_unlock(rtp);
- return pt;
- }
- }
-
- rtp_bridge_unlock(rtp);
-
- return -1;
-}
-
-const char *ast_rtp_lookup_mime_subtype(const int isAstFormat, const int code,
- enum ast_rtp_options options)
-{
- unsigned int i;
-
- for (i = 0; i < ARRAY_LEN(mimeTypes); ++i) {
- if ((mimeTypes[i].payloadType.code == code) && (mimeTypes[i].payloadType.isAstFormat == isAstFormat)) {
- if (isAstFormat &&
- (code == AST_FORMAT_G726_AAL2) &&
- (options & AST_RTP_OPT_G726_NONSTANDARD))
- return "G726-32";
- else
- return mimeTypes[i].subtype;
- }
- }
-
- return "";
-}
-
-unsigned int ast_rtp_lookup_sample_rate(int isAstFormat, int code)
-{
- unsigned int i;
-
- for (i = 0; i < ARRAY_LEN(mimeTypes); ++i) {
- if ((mimeTypes[i].payloadType.code == code) && (mimeTypes[i].payloadType.isAstFormat == isAstFormat)) {
- return mimeTypes[i].sample_rate;
- }
- }
-
- return 0;
-}
-
-char *ast_rtp_lookup_mime_multiple(char *buf, size_t size, const int capability,
- const int isAstFormat, enum ast_rtp_options options)
-{
- int format;
- unsigned len;
- char *end = buf;
- char *start = buf;
-
- if (!buf || !size)
- return NULL;
-
- snprintf(end, size, "0x%x (", capability);
-
- len = strlen(end);
- end += len;
- size -= len;
- start = end;
-
- for (format = 1; format < AST_RTP_MAX; format <<= 1) {
- if (capability & format) {
- const char *name = ast_rtp_lookup_mime_subtype(isAstFormat, format, options);
-
- snprintf(end, size, "%s|", name);
- len = strlen(end);
- end += len;
- size -= len;
- }
- }
-
- if (start == end)
- ast_copy_string(start, "nothing)", size);
- else if (size > 1)
- *(end -1) = ')';
-
- return buf;
-}
-
-/*! \brief Open RTP or RTCP socket for a session.
- * Print a message on failure.
- */
-static int rtp_socket(const char *type)
-{
- int s = socket(AF_INET, SOCK_DGRAM, 0);
- if (s < 0) {
- if (type == NULL)
- type = "RTP/RTCP";
- ast_log(LOG_WARNING, "Unable to allocate %s socket: %s\n", type, strerror(errno));
- } else {
- long flags = fcntl(s, F_GETFL);
- fcntl(s, F_SETFL, flags | O_NONBLOCK);
-#ifdef SO_NO_CHECK
- if (nochecksums)
- setsockopt(s, SOL_SOCKET, SO_NO_CHECK, &nochecksums, sizeof(nochecksums));
-#endif
- }
- return s;
-}
-
-/*!
- * \brief Initialize a new RTCP session.
- *
- * \returns The newly initialized RTCP session.
- */
-static struct ast_rtcp *ast_rtcp_new(void)
-{
- struct ast_rtcp *rtcp;
-
- if (!(rtcp = ast_calloc(1, sizeof(*rtcp))))
- return NULL;
- rtcp->s = rtp_socket("RTCP");
- rtcp->us.sin_family = AF_INET;
- rtcp->them.sin_family = AF_INET;
- rtcp->schedid = -1;
-
- if (rtcp->s < 0) {
- ast_free(rtcp);
- return NULL;
- }
-
- return rtcp;
-}
-
-/*!
- * \brief Initialize a new RTP structure.
- *
- */
-void ast_rtp_new_init(struct ast_rtp *rtp)
-{
-#ifdef P2P_INTENSE
- ast_mutex_init(&rtp->bridge_lock);
-#endif
-
- rtp->them.sin_family = AF_INET;
- rtp->us.sin_family = AF_INET;
- rtp->ssrc = ast_random();
- rtp->seqno = ast_random() & 0xffff;
- ast_set_flag(rtp, FLAG_HAS_DTMF);
- rtp->strict_rtp_state = (strictrtp ? STRICT_RTP_LEARN : STRICT_RTP_OPEN);
-}
-
-struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode, struct in_addr addr)
-{
- struct ast_rtp *rtp;
- int x;
- int startplace;
-
- if (!(rtp = ast_calloc(1, sizeof(*rtp))))
- return NULL;
-
- ast_rtp_new_init(rtp);
-
- rtp->s = rtp_socket("RTP");
- if (rtp->s < 0)
- goto fail;
- if (sched && rtcpenable) {
- rtp->sched = sched;
- rtp->rtcp = ast_rtcp_new();
- }
-
- /*
- * Try to bind the RTP port, x, and possibly the RTCP port, x+1 as well.
- * Start from a random (even, by RTP spec) port number, and
- * iterate until success or no ports are available.
- * Note that the requirement of RTP port being even, or RTCP being the
- * next one, cannot be enforced in presence of a NAT box because the
- * mapping is not under our control.
- */
- x = (rtpend == rtpstart) ? rtpstart : (ast_random() % (rtpend - rtpstart)) + rtpstart;
- x = x & ~1; /* make it an even number */
- startplace = x; /* remember the starting point */
- /* this is constant across the loop */
- rtp->us.sin_addr = addr;
- if (rtp->rtcp)
- rtp->rtcp->us.sin_addr = addr;
- for (;;) {
- rtp->us.sin_port = htons(x);
- if (!bind(rtp->s, (struct sockaddr *)&rtp->us, sizeof(rtp->us))) {
- /* bind succeeded, if no rtcp then we are done */
- if (!rtp->rtcp)
- break;
- /* have rtcp, try to bind it */
- rtp->rtcp->us.sin_port = htons(x + 1);
- if (!bind(rtp->rtcp->s, (struct sockaddr *)&rtp->rtcp->us, sizeof(rtp->rtcp->us)))
- break; /* success again, we are really done */
- /*
- * RTCP bind failed, so close and recreate the
- * already bound RTP socket for the next round.
- */
- close(rtp->s);
- rtp->s = rtp_socket("RTP");
- if (rtp->s < 0)
- goto fail;
- }
- /*
- * If we get here, there was an error in one of the bind()
- * calls, so make sure it is nothing unexpected.
- */
- if (errno != EADDRINUSE) {
- /* We got an error that wasn't expected, abort! */
- ast_log(LOG_ERROR, "Unexpected bind error: %s\n", strerror(errno));
- goto fail;
- }
- /*
- * One of the ports is in use. For the next iteration,
- * increment by two and handle wraparound.
- * If we reach the starting point, then declare failure.
- */
- x += 2;
- if (x > rtpend)
- x = (rtpstart + 1) & ~1;
- if (x == startplace) {
- ast_log(LOG_ERROR, "No RTP ports remaining. Can't setup media stream for this call.\n");
- goto fail;
- }
- }
- rtp->sched = sched;
- rtp->io = io;
- if (callbackmode) {
- rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp);
- ast_set_flag(rtp, FLAG_CALLBACK_MODE);
- }
- ast_rtp_pt_default(rtp);
- return rtp;
-
-fail:
- if (rtp->s >= 0)
- close(rtp->s);
- if (rtp->rtcp) {
- close(rtp->rtcp->s);
- ast_free(rtp->rtcp);
- }
- ast_free(rtp);
- return NULL;
-}
-
-struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode)
-{
- struct in_addr ia;
-
- memset(&ia, 0, sizeof(ia));
- return ast_rtp_new_with_bindaddr(sched, io, rtcpenable, callbackmode, ia);
-}
-
-int ast_rtp_setqos(struct ast_rtp *rtp, int type_of_service, int class_of_service, char *desc)
-{
- return ast_netsock_set_qos(rtp->s, type_of_service, class_of_service, desc);
-}
-
-void ast_rtp_new_source(struct ast_rtp *rtp)
-{
- if (rtp) {
- rtp->set_marker_bit = 1;
- }
- return;
-}
-
-void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
-{
- rtp->them.sin_port = them->sin_port;
- rtp->them.sin_addr = them->sin_addr;
- if (rtp->rtcp) {
- int h = ntohs(them->sin_port);
- rtp->rtcp->them.sin_port = htons(h + 1);
- rtp->rtcp->them.sin_addr = them->sin_addr;
- }
- rtp->rxseqno = 0;
- /* If strict RTP protection is enabled switch back to the learn state so we don't drop packets from above */
- if (strictrtp)
- rtp->strict_rtp_state = STRICT_RTP_LEARN;
-}
-
-int ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
-{
- if ((them->sin_family != AF_INET) ||
- (them->sin_port != rtp->them.sin_port) ||
- (them->sin_addr.s_addr != rtp->them.sin_addr.s_addr)) {
- them->sin_family = AF_INET;
- them->sin_port = rtp->them.sin_port;
- them->sin_addr = rtp->them.sin_addr;
- return 1;
- }
- return 0;
-}
-
-void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us)
-{
- *us = rtp->us;
-}
-
-struct ast_rtp *ast_rtp_get_bridged(struct ast_rtp *rtp)
-{
- struct ast_rtp *bridged = NULL;
-
- rtp_bridge_lock(rtp);
- bridged = rtp->bridged;
- rtp_bridge_unlock(rtp);
-
- return bridged;
-}
-
-void ast_rtp_stop(struct ast_rtp *rtp)
-{
- if (rtp->rtcp) {
- AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
- }
- if (rtp->red) {
- AST_SCHED_DEL(rtp->sched, rtp->red->schedid);
- free(rtp->red);
- rtp->red = NULL;
- }
-
- memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr));
- memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port));
- if (rtp->rtcp) {
- memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->rtcp->them.sin_addr));
- memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->rtcp->them.sin_port));
- }
-
- ast_clear_flag(rtp, FLAG_P2P_SENT_MARK);
-}
-
-void ast_rtp_reset(struct ast_rtp *rtp)
-{
- memset(&rtp->rxcore, 0, sizeof(rtp->rxcore));
- memset(&rtp->txcore, 0, sizeof(rtp->txcore));
- memset(&rtp->dtmfmute, 0, sizeof(rtp->dtmfmute));
- rtp->lastts = 0;
- rtp->lastdigitts = 0;
- rtp->lastrxts = 0;
- rtp->lastividtimestamp = 0;
- rtp->lastovidtimestamp = 0;
- rtp->lastitexttimestamp = 0;
- rtp->lastotexttimestamp = 0;
- rtp->lasteventseqn = 0;
- rtp->lastevent = 0;
- rtp->lasttxformat = 0;
- rtp->lastrxformat = 0;
- rtp->dtmfcount = 0;
- rtp->dtmfsamples = 0;
- rtp->seqno = 0;
- rtp->rxseqno = 0;
-}
-
-/*! Get QoS values from RTP and RTCP data (used in "sip show channelstats") */
-unsigned int ast_rtp_get_qosvalue(struct ast_rtp *rtp, enum ast_rtp_qos_vars value)
-{
- if (rtp == NULL) {
- if (option_debug > 1)
- ast_log(LOG_DEBUG, "NO RTP Structure? Kidding me? \n");
- return 0;
- }
- if (option_debug > 1 && rtp->rtcp == NULL) {
- ast_log(LOG_DEBUG, "NO RTCP structure. Maybe in RTP p2p bridging mode? \n");
- }
-
- switch (value) {
- case AST_RTP_TXCOUNT:
- return (unsigned int) rtp->txcount;
- case AST_RTP_RXCOUNT:
- return (unsigned int) rtp->rxcount;
- case AST_RTP_TXJITTER:
- return (unsigned int) (rtp->rxjitter * 100.0);
- case AST_RTP_RXJITTER:
- return (unsigned int) (rtp->rtcp ? (rtp->rtcp->reported_jitter / (unsigned int) 65536.0) : 0);
- case AST_RTP_RXPLOSS:
- return rtp->rtcp ? (rtp->rtcp->expected_prior - rtp->rtcp->received_prior) : 0;
- case AST_RTP_TXPLOSS:
- return rtp->rtcp ? rtp->rtcp->reported_lost : 0;
- case AST_RTP_RTT:
- return (unsigned int) (rtp->rtcp ? (rtp->rtcp->rtt * 100) : 0);
- }
- return 0; /* To make the compiler happy */
-}
-
-static double __ast_rtp_get_qos(struct ast_rtp *rtp, const char *qos, int *found)
-{
- *found = 1;
-
- if (!strcasecmp(qos, "remote_maxjitter"))
- return rtp->rtcp->reported_maxjitter * 1000.0;
- if (!strcasecmp(qos, "remote_minjitter"))
- return rtp->rtcp->reported_minjitter * 1000.0;
- if (!strcasecmp(qos, "remote_normdevjitter"))
- return rtp->rtcp->reported_normdev_jitter * 1000.0;
- if (!strcasecmp(qos, "remote_stdevjitter"))
- return sqrt(rtp->rtcp->reported_stdev_jitter) * 1000.0;
-
- if (!strcasecmp(qos, "local_maxjitter"))
- return rtp->rtcp->maxrxjitter * 1000.0;
- if (!strcasecmp(qos, "local_minjitter"))
- return rtp->rtcp->minrxjitter * 1000.0;
- if (!strcasecmp(qos, "local_normdevjitter"))
- return rtp->rtcp->normdev_rxjitter * 1000.0;
- if (!strcasecmp(qos, "local_stdevjitter"))
- return sqrt(rtp->rtcp->stdev_rxjitter) * 1000.0;
-
- if (!strcasecmp(qos, "maxrtt"))
- return rtp->rtcp->maxrtt * 1000.0;
- if (!strcasecmp(qos, "minrtt"))
- return rtp->rtcp->minrtt * 1000.0;
- if (!strcasecmp(qos, "normdevrtt"))
- return rtp->rtcp->normdevrtt * 1000.0;
- if (!strcasecmp(qos, "stdevrtt"))
- return sqrt(rtp->rtcp->stdevrtt) * 1000.0;
-
- *found = 0;
-
- return 0.0;
-}
-
-int ast_rtp_get_qos(struct ast_rtp *rtp, const char *qos, char *buf, unsigned int buflen)
-{
- double value;
- int found;
-
- value = __ast_rtp_get_qos(rtp, qos, &found);
-
- if (!found)
- return -1;
-
- snprintf(buf, buflen, "%.0lf", value);
-
- return 0;
-}
-
-void ast_rtp_set_vars(struct ast_channel *chan, struct ast_rtp *rtp) {
- char *audioqos;
- char *audioqos_jitter;
- char *audioqos_loss;
- char *audioqos_rtt;
- struct ast_channel *bridge;
-
- if (!rtp || !chan)
- return;
-
- bridge = ast_bridged_channel(chan);
-
- audioqos = ast_rtp_get_quality(rtp, NULL, RTPQOS_SUMMARY);
- audioqos_jitter = ast_rtp_get_quality(rtp, NULL, RTPQOS_JITTER);
- audioqos_loss = ast_rtp_get_quality(rtp, NULL, RTPQOS_LOSS);
- audioqos_rtt = ast_rtp_get_quality(rtp, NULL, RTPQOS_RTT);
-
- pbx_builtin_setvar_helper(chan, "RTPAUDIOQOS", audioqos);
- pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSJITTER", audioqos_jitter);
- pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSLOSS", audioqos_loss);
- pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSRTT", audioqos_rtt);
-
- if (!bridge)
- return;
-
- pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSBRIDGED", audioqos);
- pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSJITTERBRIDGED", audioqos_jitter);
- pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSLOSSBRIDGED", audioqos_loss);
- pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSRTTBRIDGED", audioqos_rtt);
-}
-
-static char *__ast_rtp_get_quality_jitter(struct ast_rtp *rtp)
-{
- /*
- *ssrc our ssrc
- *themssrc their ssrc
- *lp lost packets
- *rxjitter our calculated jitter(rx)
- *rxcount no. received packets
- *txjitter reported jitter of the other end
- *txcount transmitted packets
- *rlp remote lost packets
- *rtt round trip time
- */
-#define RTCP_JITTER_FORMAT1 \
- "minrxjitter=%f;" \
- "maxrxjitter=%f;" \
- "avgrxjitter=%f;" \
- "stdevrxjitter=%f;" \
- "reported_minjitter=%f;" \
- "reported_maxjitter=%f;" \
- "reported_avgjitter=%f;" \
- "reported_stdevjitter=%f;"
-
-#define RTCP_JITTER_FORMAT2 \
- "rxjitter=%f;"
-
- if (rtp->rtcp && rtp->rtcp->rtcp_info) {
- snprintf(rtp->rtcp->quality_jitter, sizeof(rtp->rtcp->quality_jitter), RTCP_JITTER_FORMAT1,
- rtp->rtcp->minrxjitter,
- rtp->rtcp->maxrxjitter,
- rtp->rtcp->normdev_rxjitter,
- sqrt(rtp->rtcp->stdev_rxjitter),
- rtp->rtcp->reported_minjitter,
- rtp->rtcp->reported_maxjitter,
- rtp->rtcp->reported_normdev_jitter,
- sqrt(rtp->rtcp->reported_stdev_jitter)
- );
- } else {
- snprintf(rtp->rtcp->quality_jitter, sizeof(rtp->rtcp->quality_jitter), RTCP_JITTER_FORMAT2,
- rtp->rxjitter
- );
- }
-
- return rtp->rtcp->quality_jitter;
-
-#undef RTCP_JITTER_FORMAT1
-#undef RTCP_JITTER_FORMAT2
-}
-
-static char *__ast_rtp_get_quality_loss(struct ast_rtp *rtp)
-{
- unsigned int lost;
- unsigned int extended;
- unsigned int expected;
- int fraction;
-
-#define RTCP_LOSS_FORMAT1 \
- "minrxlost=%f;" \
- "maxrxlost=%f;" \
- "avgrxlostr=%f;" \
- "stdevrxlost=%f;" \
- "reported_minlost=%f;" \
- "reported_maxlost=%f;" \
- "reported_avglost=%f;" \
- "reported_stdevlost=%f;"
-
-#define RTCP_LOSS_FORMAT2 \
- "lost=%d;" \
- "expected=%d;"
-
- if (rtp->rtcp && rtp->rtcp->rtcp_info && rtp->rtcp->maxrxlost > 0) {
- snprintf(rtp->rtcp->quality_loss, sizeof(rtp->rtcp->quality_loss), RTCP_LOSS_FORMAT1,
- rtp->rtcp->minrxlost,
- rtp->rtcp->maxrxlost,
- rtp->rtcp->normdev_rxlost,
- sqrt(rtp->rtcp->stdev_rxlost),
- rtp->rtcp->reported_minlost,
- rtp->rtcp->reported_maxlost,
- rtp->rtcp->reported_normdev_lost,
- sqrt(rtp->rtcp->reported_stdev_lost)
- );
- } else {
- extended = rtp->cycles + rtp->lastrxseqno;
- expected = extended - rtp->seedrxseqno + 1;
- if (rtp->rxcount > expected)
- expected += rtp->rxcount - expected;
- lost = expected - rtp->rxcount;
-
- if (!expected || lost <= 0)
- fraction = 0;
- else
- fraction = (lost << 8) / expected;
-
- snprintf(rtp->rtcp->quality_loss, sizeof(rtp->rtcp->quality_loss), RTCP_LOSS_FORMAT2,
- lost,
- expected
- );
- }
-
- return rtp->rtcp->quality_loss;
-
-#undef RTCP_LOSS_FORMAT1
-#undef RTCP_LOSS_FORMAT2
-}
-
-static char *__ast_rtp_get_quality_rtt(struct ast_rtp *rtp)
-{
- if (rtp->rtcp && rtp->rtcp->rtcp_info) {
- snprintf(rtp->rtcp->quality_rtt, sizeof(rtp->rtcp->quality_rtt), "minrtt=%f;maxrtt=%f;avgrtt=%f;stdevrtt=%f;",
- rtp->rtcp->minrtt,
- rtp->rtcp->maxrtt,
- rtp->rtcp->normdevrtt,
- sqrt(rtp->rtcp->stdevrtt)
- );
- } else {
- snprintf(rtp->rtcp->quality_rtt, sizeof(rtp->rtcp->quality_rtt), "Not available");
- }
-
- return rtp->rtcp->quality_rtt;
-}
-
-static char *__ast_rtp_get_quality(struct ast_rtp *rtp)
-{
- /*
- *ssrc our ssrc
- *themssrc their ssrc
- *lp lost packets
- *rxjitter our calculated jitter(rx)
- *rxcount no. received packets
- *txjitter reported jitter of the other end
- *txcount transmitted packets
- *rlp remote lost packets
- *rtt round trip time
- */
-
- if (rtp->rtcp && rtp->rtcp->rtcp_info) {
- snprintf(rtp->rtcp->quality, sizeof(rtp->rtcp->quality),
- "ssrc=%u;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f",
- rtp->ssrc,
- rtp->themssrc,
- rtp->rtcp->expected_prior - rtp->rtcp->received_prior,
- rtp->rxjitter,
- rtp->rxcount,
- (double)rtp->rtcp->reported_jitter / 65536.0,
- rtp->txcount,
- rtp->rtcp->reported_lost,
- rtp->rtcp->rtt
- );
- } else {
- snprintf(rtp->rtcp->quality, sizeof(rtp->rtcp->quality), "ssrc=%u;themssrc=%u;rxjitter=%f;rxcount=%u;txcount=%u;",
- rtp->ssrc,
- rtp->themssrc,
- rtp->rxjitter,
- rtp->rxcount,
- rtp->txcount
- );
- }
-
- return rtp->rtcp->quality;
-}
-
-char *ast_rtp_get_quality(struct ast_rtp *rtp, struct ast_rtp_quality *qual, enum ast_rtp_quality_type qtype)
-{
- if (qual && rtp) {
- qual->local_ssrc = rtp->ssrc;
- qual->local_jitter = rtp->rxjitter;
- qual->local_count = rtp->rxcount;
- qual->remote_ssrc = rtp->themssrc;
- qual->remote_count = rtp->txcount;
-
- if (rtp->rtcp) {
- qual->local_lostpackets = rtp->rtcp->expected_prior - rtp->rtcp->received_prior;
- qual->remote_lostpackets = rtp->rtcp->reported_lost;
- qual->remote_jitter = rtp->rtcp->reported_jitter / 65536.0;
- qual->rtt = rtp->rtcp->rtt;
- }
- }
-
- switch (qtype) {
- case RTPQOS_SUMMARY:
- return __ast_rtp_get_quality(rtp);
- case RTPQOS_JITTER:
- return __ast_rtp_get_quality_jitter(rtp);
- case RTPQOS_LOSS:
- return __ast_rtp_get_quality_loss(rtp);
- case RTPQOS_RTT:
- return __ast_rtp_get_quality_rtt(rtp);
- }
-
- return NULL;
-}
-
-void ast_rtp_destroy(struct ast_rtp *rtp)
-{
- if (rtcp_debug_test_addr(&rtp->them) || rtcpstats) {
- /*Print some info on the call here */
- ast_verbose(" RTP-stats\n");
- ast_verbose("* Our Receiver:\n");
- ast_verbose(" SSRC: %u\n", rtp->themssrc);
- ast_verbose(" Received packets: %u\n", rtp->rxcount);
- ast_verbose(" Lost packets: %u\n", rtp->rtcp ? (rtp->rtcp->expected_prior - rtp->rtcp->received_prior) : 0);
- ast_verbose(" Jitter: %.4f\n", rtp->rxjitter);
- ast_verbose(" Transit: %.4f\n", rtp->rxtransit);
- ast_verbose(" RR-count: %u\n", rtp->rtcp ? rtp->rtcp->rr_count : 0);
- ast_verbose("* Our Sender:\n");
- ast_verbose(" SSRC: %u\n", rtp->ssrc);
- ast_verbose(" Sent packets: %u\n", rtp->txcount);
- ast_verbose(" Lost packets: %u\n", rtp->rtcp ? rtp->rtcp->reported_lost : 0);
- ast_verbose(" Jitter: %u\n", rtp->rtcp ? (rtp->rtcp->reported_jitter / (unsigned int)65536.0) : 0);
- ast_verbose(" SR-count: %u\n", rtp->rtcp ? rtp->rtcp->sr_count : 0);
- ast_verbose(" RTT: %f\n", rtp->rtcp ? rtp->rtcp->rtt : 0);
- }
-
- manager_event(EVENT_FLAG_REPORTING, "RTPReceiverStat", "SSRC: %u\r\n"
- "ReceivedPackets: %u\r\n"
- "LostPackets: %u\r\n"
- "Jitter: %.4f\r\n"
- "Transit: %.4f\r\n"
- "RRCount: %u\r\n",
- rtp->themssrc,
- rtp->rxcount,
- rtp->rtcp ? (rtp->rtcp->expected_prior - rtp->rtcp->received_prior) : 0,
- rtp->rxjitter,
- rtp->rxtransit,
- rtp->rtcp ? rtp->rtcp->rr_count : 0);
- manager_event(EVENT_FLAG_REPORTING, "RTPSenderStat", "SSRC: %u\r\n"
- "SentPackets: %u\r\n"
- "LostPackets: %u\r\n"
- "Jitter: %u\r\n"
- "SRCount: %u\r\n"
- "RTT: %f\r\n",
- rtp->ssrc,
- rtp->txcount,
- rtp->rtcp ? rtp->rtcp->reported_lost : 0,
- rtp->rtcp ? rtp->rtcp->reported_jitter : 0,
- rtp->rtcp ? rtp->rtcp->sr_count : 0,
- rtp->rtcp ? rtp->rtcp->rtt : 0);
- if (rtp->smoother)
- ast_smoother_free(rtp->smoother);
- if (rtp->ioid)
- ast_io_remove(rtp->io, rtp->ioid);
- if (rtp->s > -1)
- close(rtp->s);
- if (rtp->rtcp) {
- AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
- close(rtp->rtcp->s);
- ast_free(rtp->rtcp);
- rtp->rtcp=NULL;
- }
-#ifdef P2P_INTENSE
- ast_mutex_destroy(&rtp->bridge_lock);
-#endif
- ast_free(rtp);
-}
-
-static unsigned int calc_txstamp(struct ast_rtp *rtp, struct timeval *delivery)
-{
- struct timeval t;
- long ms;
- if (ast_tvzero(rtp->txcore)) {
- rtp->txcore = ast_tvnow();
- /* Round to 20ms for nice, pretty timestamps */
- rtp->txcore.tv_usec -= rtp->txcore.tv_usec % 20000;
- }
- /* Use previous txcore if available */
- t = (delivery && !ast_tvzero(*delivery)) ? *delivery : ast_tvnow();
- ms = ast_tvdiff_ms(t, rtp->txcore);
- if (ms < 0)
- ms = 0;
- /* Use what we just got for next time */
- rtp->txcore = t;
- return (unsigned int) ms;
-}
-
-/*! \brief Send begin frames for DTMF */
-int ast_rtp_senddigit_begin(struct ast_rtp *rtp, char digit)
-{
- unsigned int *rtpheader;
- int hdrlen = 12, res = 0, i = 0, payload = 0;
- char data[256];
-
- if ((digit <= '9') && (digit >= '0'))
- digit -= '0';
- else if (digit == '*')
- digit = 10;
- else if (digit == '#')
- digit = 11;
- else if ((digit >= 'A') && (digit <= 'D'))
- digit = digit - 'A' + 12;
- else if ((digit >= 'a') && (digit <= 'd'))
- digit = digit - 'a' + 12;
- else {
- ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
- return 0;
- }
-
- /* If we have no peer, return immediately */
- if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
- return 0;
-
- payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_DTMF);
-
- rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
- rtp->send_duration = 160;
- rtp->lastdigitts = rtp->lastts + rtp->send_duration;
-
- /* Get a pointer to the header */
- rtpheader = (unsigned int *)data;
- rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno));
- rtpheader[1] = htonl(rtp->lastdigitts);
- rtpheader[2] = htonl(rtp->ssrc);
-
- for (i = 0; i < 2; i++) {
- rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration));
- res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them));
- if (res < 0)
- ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n",
- ast_inet_ntoa(rtp->them.sin_addr),
- ntohs(rtp->them.sin_port), strerror(errno));
- if (rtp_debug_test_addr(&rtp->them))
- ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
- ast_inet_ntoa(rtp->them.sin_addr),
- ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
- /* Increment sequence number */
- rtp->seqno++;
- /* Increment duration */
- rtp->send_duration += 160;
- /* Clear marker bit and set seqno */
- rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno));
- }
-
- /* Since we received a begin, we can safely store the digit and disable any compensation */
- rtp->sending_digit = 1;
- rtp->send_digit = digit;
- rtp->send_payload = payload;
-
- return 0;
-}
-
-/*! \brief Send continuation frame for DTMF */
-static int ast_rtp_senddigit_continuation(struct ast_rtp *rtp)
-{
- unsigned int *rtpheader;
- int hdrlen = 12, res = 0;
- char data[256];
-
- if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
- return 0;
-
- /* Setup packet to send */
- rtpheader = (unsigned int *)data;
- rtpheader[0] = htonl((2 << 30) | (1 << 23) | (rtp->send_payload << 16) | (rtp->seqno));
- rtpheader[1] = htonl(rtp->lastdigitts);
- rtpheader[2] = htonl(rtp->ssrc);
- rtpheader[3] = htonl((rtp->send_digit << 24) | (0xa << 16) | (rtp->send_duration));
- rtpheader[0] = htonl((2 << 30) | (rtp->send_payload << 16) | (rtp->seqno));
-
- /* Transmit */
- res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them));
- if (res < 0)
- ast_log(LOG_ERROR, "RTP Transmission error to %s:%d: %s\n",
- ast_inet_ntoa(rtp->them.sin_addr),
- ntohs(rtp->them.sin_port), strerror(errno));
- if (rtp_debug_test_addr(&rtp->them))
- ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
- ast_inet_ntoa(rtp->them.sin_addr),
- ntohs(rtp->them.sin_port), rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
-
- /* Increment sequence number */
- rtp->seqno++;
- /* Increment duration */
- rtp->send_duration += 160;
-
- return 0;
-}
-
-/*! \brief Send end packets for DTMF */
-int ast_rtp_senddigit_end(struct ast_rtp *rtp, char digit)
-{
- unsigned int *rtpheader;
- int hdrlen = 12, res = 0, i = 0;
- char data[256];
-
- /* If no address, then bail out */
- if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
- return 0;
-
- if ((digit <= '9') && (digit >= '0'))
- digit -= '0';
- else if (digit == '*')
- digit = 10;
- else if (digit == '#')
- digit = 11;
- else if ((digit >= 'A') && (digit <= 'D'))
- digit = digit - 'A' + 12;
- else if ((digit >= 'a') && (digit <= 'd'))
- digit = digit - 'a' + 12;
- else {
- ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
- return 0;
- }
-
- rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
-
- rtpheader = (unsigned int *)data;
- rtpheader[1] = htonl(rtp->lastdigitts);
- rtpheader[2] = htonl(rtp->ssrc);
- rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration));
- /* Set end bit */
- rtpheader[3] |= htonl((1 << 23));
-
- /* Send 3 termination packets */
- for (i = 0; i < 3; i++) {
- rtpheader[0] = htonl((2 << 30) | (rtp->send_payload << 16) | (rtp->seqno));
- res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them));
- rtp->seqno++;
- if (res < 0)
- ast_log(LOG_ERROR, "RTP Transmission error to %s:%d: %s\n",
- ast_inet_ntoa(rtp->them.sin_addr),
- ntohs(rtp->them.sin_port), strerror(errno));
- if (rtp_debug_test_addr(&rtp->them))
- ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
- ast_inet_ntoa(rtp->them.sin_addr),
- ntohs(rtp->them.sin_port), rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
- }
- rtp->lastts += rtp->send_duration;
- rtp->sending_digit = 0;
- rtp->send_digit = 0;
-
- return res;
-}
-
-/*! \brief Public function: Send an H.261 fast update request, some devices need this rather than SIP XML */
-int ast_rtcp_send_h261fur(void *data)
-{
- struct ast_rtp *rtp = data;
- int res;
-
- rtp->rtcp->sendfur = 1;
- res = ast_rtcp_write(data);
-
- return res;
-}
-
-/*! \brief Send RTCP sender's report */
-static int ast_rtcp_write_sr(const void *data)
-{
- struct ast_rtp *rtp = (struct ast_rtp *)data;
- int res;
- int len = 0;
- struct timeval now;
- unsigned int now_lsw;
- unsigned int now_msw;
- unsigned int *rtcpheader;
- unsigned int lost;
- unsigned int extended;
- unsigned int expected;
- unsigned int expected_interval;
- unsigned int received_interval;
- int lost_interval;
- int fraction;
- struct timeval dlsr;
- char bdata[512];
-
- /* Commented condition is always not NULL if rtp->rtcp is not NULL */
- if (!rtp || !rtp->rtcp/* || (&rtp->rtcp->them.sin_addr == 0)*/)
- return 0;
-
- if (!rtp->rtcp->them.sin_addr.s_addr) { /* This'll stop rtcp for this rtp session */
- ast_verbose("RTCP SR transmission error, rtcp halted\n");
- AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
- return 0;
- }
-
- gettimeofday(&now, NULL);
- timeval2ntp(now, &now_msw, &now_lsw); /* fill thses ones in from utils.c*/
- rtcpheader = (unsigned int *)bdata;
- rtcpheader[1] = htonl(rtp->ssrc); /* Our SSRC */
- rtcpheader[2] = htonl(now_msw); /* now, MSW. gettimeofday() + SEC_BETWEEN_1900_AND_1970*/
- rtcpheader[3] = htonl(now_lsw); /* now, LSW */
- rtcpheader[4] = htonl(rtp->lastts); /* FIXME shouldn't be that, it should be now */
- rtcpheader[5] = htonl(rtp->txcount); /* No. packets sent */
- rtcpheader[6] = htonl(rtp->txoctetcount); /* No. bytes sent */
- len += 28;
-
- extended = rtp->cycles + rtp->lastrxseqno;
- expected = extended - rtp->seedrxseqno + 1;
- if (rtp->rxcount > expected)
- expected += rtp->rxcount - expected;
- lost = expected - rtp->rxcount;
- expected_interval = expected - rtp->rtcp->expected_prior;
- rtp->rtcp->expected_prior = expected;
- received_interval = rtp->rxcount - rtp->rtcp->received_prior;
- rtp->rtcp->received_prior = rtp->rxcount;
- lost_interval = expected_interval - received_interval;
- if (expected_interval == 0 || lost_interval <= 0)
- fraction = 0;
- else
- fraction = (lost_interval << 8) / expected_interval;
- timersub(&now, &rtp->rtcp->rxlsr, &dlsr);
- rtcpheader[7] = htonl(rtp->themssrc);
- rtcpheader[8] = htonl(((fraction & 0xff) << 24) | (lost & 0xffffff));
- rtcpheader[9] = htonl((rtp->cycles) | ((rtp->lastrxseqno & 0xffff)));
- rtcpheader[10] = htonl((unsigned int)(rtp->rxjitter * 65536.));
- rtcpheader[11] = htonl(rtp->rtcp->themrxlsr);
- rtcpheader[12] = htonl((((dlsr.tv_sec * 1000) + (dlsr.tv_usec / 1000)) * 65536) / 1000);
- len += 24;
-
- rtcpheader[0] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_SR << 16) | ((len/4)-1));
-
- if (rtp->rtcp->sendfur) {
- rtcpheader[13] = htonl((2 << 30) | (0 << 24) | (RTCP_PT_FUR << 16) | 1);
- rtcpheader[14] = htonl(rtp->ssrc); /* Our SSRC */
- len += 8;
- rtp->rtcp->sendfur = 0;
- }
-
- /* Insert SDES here. Probably should make SDES text equal to mimetypes[code].type (not subtype 'cos */
- /* it can change mid call, and SDES can't) */
- rtcpheader[len/4] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_SDES << 16) | 2);
- rtcpheader[(len/4)+1] = htonl(rtp->ssrc); /* Our SSRC */
- rtcpheader[(len/4)+2] = htonl(0x01 << 24); /* Empty for the moment */
- len += 12;
-
- res = sendto(rtp->rtcp->s, (unsigned int *)rtcpheader, len, 0, (struct sockaddr *)&rtp->rtcp->them, sizeof(rtp->rtcp->them));
- if (res < 0) {
- ast_log(LOG_ERROR, "RTCP SR transmission error to %s:%d, rtcp halted %s\n",ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port), strerror(errno));
- AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
- return 0;
- }
-
- /* FIXME Don't need to get a new one */
- gettimeofday(&rtp->rtcp->txlsr, NULL);
- rtp->rtcp->sr_count++;
-
- rtp->rtcp->lastsrtxcount = rtp->txcount;
-
- if (rtcp_debug_test_addr(&rtp->rtcp->them)) {
- ast_verbose("* Sent RTCP SR to %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
- ast_verbose(" Our SSRC: %u\n", rtp->ssrc);
- ast_verbose(" Sent(NTP): %u.%010u\n", (unsigned int)now.tv_sec, (unsigned int)now.tv_usec*4096);
- ast_verbose(" Sent(RTP): %u\n", rtp->lastts);
- ast_verbose(" Sent packets: %u\n", rtp->txcount);
- ast_verbose(" Sent octets: %u\n", rtp->txoctetcount);
- ast_verbose(" Report block:\n");
- ast_verbose(" Fraction lost: %u\n", fraction);
- ast_verbose(" Cumulative loss: %u\n", lost);
- ast_verbose(" IA jitter: %.4f\n", rtp->rxjitter);
- ast_verbose(" Their last SR: %u\n", rtp->rtcp->themrxlsr);
- ast_verbose(" DLSR: %4.4f (sec)\n\n", (double)(ntohl(rtcpheader[12])/65536.0));
- }
- manager_event(EVENT_FLAG_REPORTING, "RTCPSent", "To: %s:%d\r\n"
- "OurSSRC: %u\r\n"
- "SentNTP: %u.%010u\r\n"
- "SentRTP: %u\r\n"
- "SentPackets: %u\r\n"
- "SentOctets: %u\r\n"
- "ReportBlock:\r\n"
- "FractionLost: %u\r\n"
- "CumulativeLoss: %u\r\n"
- "IAJitter: %.4f\r\n"
- "TheirLastSR: %u\r\n"
- "DLSR: %4.4f (sec)\r\n",
- ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port),
- rtp->ssrc,
- (unsigned int)now.tv_sec, (unsigned int)now.tv_usec*4096,
- rtp->lastts,
- rtp->txcount,
- rtp->txoctetcount,
- fraction,
- lost,
- rtp->rxjitter,
- rtp->rtcp->themrxlsr,
- (double)(ntohl(rtcpheader[12])/65536.0));
- return res;
-}
-
-/*! \brief Send RTCP recipient's report */
-static int ast_rtcp_write_rr(const void *data)
-{
- struct ast_rtp *rtp = (struct ast_rtp *)data;
- int res;
- int len = 32;
- unsigned int lost;
- unsigned int extended;
- unsigned int expected;
- unsigned int expected_interval;
- unsigned int received_interval;
- int lost_interval;
- struct timeval now;
- unsigned int *rtcpheader;
- char bdata[1024];
- struct timeval dlsr;
- int fraction;
-
- double rxlost_current;
-
- if (!rtp || !rtp->rtcp || (&rtp->rtcp->them.sin_addr == 0))
- return 0;
-
- if (!rtp->rtcp->them.sin_addr.s_addr) {
- ast_log(LOG_ERROR, "RTCP RR transmission error, rtcp halted\n");
- AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
- return 0;
- }
-
- extended = rtp->cycles + rtp->lastrxseqno;
- expected = extended - rtp->seedrxseqno + 1;
- lost = expected - rtp->rxcount;
- expected_interval = expected - rtp->rtcp->expected_prior;
- rtp->rtcp->expected_prior = expected;
- received_interval = rtp->rxcount - rtp->rtcp->received_prior;
- rtp->rtcp->received_prior = rtp->rxcount;
- lost_interval = expected_interval - received_interval;
-
- if (lost_interval <= 0)
- rtp->rtcp->rxlost = 0;
- else rtp->rtcp->rxlost = rtp->rtcp->rxlost;
- if (rtp->rtcp->rxlost_count == 0)
- rtp->rtcp->minrxlost = rtp->rtcp->rxlost;
- if (lost_interval < rtp->rtcp->minrxlost)
- rtp->rtcp->minrxlost = rtp->rtcp->rxlost;
- if (lost_interval > rtp->rtcp->maxrxlost)
- rtp->rtcp->maxrxlost = rtp->rtcp->rxlost;
-
- rxlost_current = normdev_compute(rtp->rtcp->normdev_rxlost, rtp->rtcp->rxlost, rtp->rtcp->rxlost_count);
- rtp->rtcp->stdev_rxlost = stddev_compute(rtp->rtcp->stdev_rxlost, rtp->rtcp->rxlost, rtp->rtcp->normdev_rxlost, rxlost_current, rtp->rtcp->rxlost_count);
- rtp->rtcp->normdev_rxlost = rxlost_current;
- rtp->rtcp->rxlost_count++;
-
- if (expected_interval == 0 || lost_interval <= 0)
- fraction = 0;
- else
- fraction = (lost_interval << 8) / expected_interval;
- gettimeofday(&now, NULL);
- timersub(&now, &rtp->rtcp->rxlsr, &dlsr);
- rtcpheader = (unsigned int *)bdata;
- rtcpheader[0] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_RR << 16) | ((len/4)-1));
- rtcpheader[1] = htonl(rtp->ssrc);
- rtcpheader[2] = htonl(rtp->themssrc);
- rtcpheader[3] = htonl(((fraction & 0xff) << 24) | (lost & 0xffffff));
- rtcpheader[4] = htonl((rtp->cycles) | ((rtp->lastrxseqno & 0xffff)));
- rtcpheader[5] = htonl((unsigned int)(rtp->rxjitter * 65536.));
- rtcpheader[6] = htonl(rtp->rtcp->themrxlsr);
- rtcpheader[7] = htonl((((dlsr.tv_sec * 1000) + (dlsr.tv_usec / 1000)) * 65536) / 1000);
-
- if (rtp->rtcp->sendfur) {
- rtcpheader[8] = htonl((2 << 30) | (0 << 24) | (RTCP_PT_FUR << 16) | 1); /* Header from page 36 in RFC 3550 */
- rtcpheader[9] = htonl(rtp->ssrc); /* Our SSRC */
- len += 8;
- rtp->rtcp->sendfur = 0;
- }
-
- /*! \note Insert SDES here. Probably should make SDES text equal to mimetypes[code].type (not subtype 'cos
- it can change mid call, and SDES can't) */
- rtcpheader[len/4] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_SDES << 16) | 2);
- rtcpheader[(len/4)+1] = htonl(rtp->ssrc); /* Our SSRC */
- rtcpheader[(len/4)+2] = htonl(0x01 << 24); /* Empty for the moment */
- len += 12;
-
- res = sendto(rtp->rtcp->s, (unsigned int *)rtcpheader, len, 0, (struct sockaddr *)&rtp->rtcp->them, sizeof(rtp->rtcp->them));
-
- if (res < 0) {
- ast_log(LOG_ERROR, "RTCP RR transmission error, rtcp halted: %s\n",strerror(errno));
- /* Remove the scheduler */
- AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
- return 0;
- }
-
- rtp->rtcp->rr_count++;
-
- if (rtcp_debug_test_addr(&rtp->rtcp->them)) {
- ast_verbose("\n* Sending RTCP RR to %s:%d\n"
- " Our SSRC: %u\nTheir SSRC: %u\niFraction lost: %d\nCumulative loss: %u\n"
- " IA jitter: %.4f\n"
- " Their last SR: %u\n"
- " DLSR: %4.4f (sec)\n\n",
- ast_inet_ntoa(rtp->rtcp->them.sin_addr),
- ntohs(rtp->rtcp->them.sin_port),
- rtp->ssrc, rtp->themssrc, fraction, lost,
- rtp->rxjitter,
- rtp->rtcp->themrxlsr,
- (double)(ntohl(rtcpheader[7])/65536.0));
- }
-
- return res;
-}
-
-/*! \brief Write and RTCP packet to the far end
- * \note Decide if we are going to send an SR (with Reception Block) or RR
- * RR is sent if we have not sent any rtp packets in the previous interval */
-static int ast_rtcp_write(const void *data)
-{
- struct ast_rtp *rtp = (struct ast_rtp *)data;
- int res;
-
- if (!rtp || !rtp->rtcp)
- return 0;
-
- if (rtp->txcount > rtp->rtcp->lastsrtxcount)
- res = ast_rtcp_write_sr(data);
- else
- res = ast_rtcp_write_rr(data);
-
- return res;
-}
-
-/*! \brief generate comfort noice (CNG) */
-int ast_rtp_sendcng(struct ast_rtp *rtp, int level)
-{
- unsigned int *rtpheader;
- int hdrlen = 12;
- int res;
- int payload;
- char data[256];
- level = 127 - (level & 0x7f);
- payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_CN);
-
- /* If we have no peer, return immediately */
- if (!rtp->them.sin_addr.s_addr)
- return 0;
-
- rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
-
- /* Get a pointer to the header */
- rtpheader = (unsigned int *)data;
- rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno++));
- rtpheader[1] = htonl(rtp->lastts);
- rtpheader[2] = htonl(rtp->ssrc);
- data[12] = level;
- if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
- res = sendto(rtp->s, (void *)rtpheader, hdrlen + 1, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them));
- if (res <0)
- ast_log(LOG_ERROR, "RTP Comfort Noise Transmission error to %s:%d: %s\n", ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno));
- if (rtp_debug_test_addr(&rtp->them))
- ast_verbose("Sent Comfort Noise RTP packet to %s:%u (type %d, seq %u, ts %u, len %d)\n"
- , ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastts,res - hdrlen);
-
- }
- return 0;
-}
-
-/*! \brief Write RTP packet with audio or video media frames into UDP packet */
-static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec)
-{
- unsigned char *rtpheader;
- int hdrlen = 12;
- int res;
- unsigned int ms;
- int pred;
- int mark = 0;
-
- if (rtp->sending_digit) {
- return 0;
- }
-
- ms = calc_txstamp(rtp, &f->delivery);
- /* Default prediction */
- if (f->frametype == AST_FRAME_VOICE) {
- pred = rtp->lastts + f->samples;
-
- /* Re-calculate last TS */
- rtp->lastts = rtp->lastts + ms * 8;
- if (ast_tvzero(f->delivery)) {
- /* If this isn't an absolute delivery time, Check if it is close to our prediction,
- and if so, go with our prediction */
- if (abs(rtp->lastts - pred) < MAX_TIMESTAMP_SKEW)
- rtp->lastts = pred;
- else {
- ast_debug(3, "Difference is %d, ms is %d\n", abs(rtp->lastts - pred), ms);
- mark = 1;
- }
- }
- } else if (f->frametype == AST_FRAME_VIDEO) {
- mark = f->subclass & 0x1;
- pred = rtp->lastovidtimestamp + f->samples;
- /* Re-calculate last TS */
- rtp->lastts = rtp->lastts + ms * 90;
- /* If it's close to our prediction, go for it */
- if (ast_tvzero(f->delivery)) {
- if (abs(rtp->lastts - pred) < 7200) {
- rtp->lastts = pred;
- rtp->lastovidtimestamp += f->samples;
- } else {
- ast_debug(3, "Difference is %d, ms is %d (%d), pred/ts/samples %d/%d/%d\n", abs(rtp->lastts - pred), ms, ms * 90, rtp->lastts, pred, f->samples);
- rtp->lastovidtimestamp = rtp->lastts;
- }
- }
- } else {
- pred = rtp->lastotexttimestamp + f->samples;
- /* Re-calculate last TS */
- rtp->lastts = rtp->lastts + ms * 90;
- /* If it's close to our prediction, go for it */
- if (ast_tvzero(f->delivery)) {
- if (abs(rtp->lastts - pred) < 7200) {
- rtp->lastts = pred;
- rtp->lastotexttimestamp += f->samples;
- } else {
- ast_debug(3, "Difference is %d, ms is %d (%d), pred/ts/samples %d/%d/%d\n", abs(rtp->lastts - pred), ms, ms * 90, rtp->lastts, pred, f->samples);
- rtp->lastotexttimestamp = rtp->lastts;
- }
- }
- }
-
- /* If we have been explicitly told to set the marker bit do so */
- if (rtp->set_marker_bit) {
- mark = 1;
- rtp->set_marker_bit = 0;
- }
-
- /* If the timestamp for non-digit packets has moved beyond the timestamp
- for digits, update the digit timestamp.
- */
- if (rtp->lastts > rtp->lastdigitts)
- rtp->lastdigitts = rtp->lastts;
-
- if (ast_test_flag(f, AST_FRFLAG_HAS_TIMING_INFO))
- rtp->lastts = f->ts * 8;
-
- /* Get a pointer to the header */
- rtpheader = (unsigned char *)(f->data.ptr - hdrlen);
-
- put_unaligned_uint32(rtpheader, htonl((2 << 30) | (codec << 16) | (rtp->seqno) | (mark << 23)));
- put_unaligned_uint32(rtpheader + 4, htonl(rtp->lastts));
- put_unaligned_uint32(rtpheader + 8, htonl(rtp->ssrc));
-
- if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
- res = sendto(rtp->s, (void *)rtpheader, f->datalen + hdrlen, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them));
- if (res < 0) {
- if (!rtp->nat || (rtp->nat && (ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_ACTIVE))) {
- ast_debug(1, "RTP Transmission error of packet %d to %s:%d: %s\n", rtp->seqno, ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno));
- } else if (((ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_INACTIVE) || rtpdebug) && !ast_test_flag(rtp, FLAG_NAT_INACTIVE_NOWARN)) {
- /* Only give this error message once if we are not RTP debugging */
- if (option_debug || rtpdebug)
- ast_debug(0, "RTP NAT: Can't write RTP to private address %s:%d, waiting for other end to send audio...\n", ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
- ast_set_flag(rtp, FLAG_NAT_INACTIVE_NOWARN);
- }
- } else {
- rtp->txcount++;
- rtp->txoctetcount +=(res - hdrlen);
-
- /* Do not schedule RR if RTCP isn't run */
- if (rtp->rtcp && rtp->rtcp->them.sin_addr.s_addr && rtp->rtcp->schedid < 1) {
- rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp);
- }
- }
-
- if (rtp_debug_test_addr(&rtp->them))
- ast_verbose("Sent RTP packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
- ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), codec, rtp->seqno, rtp->lastts,res - hdrlen);
- }
-
- rtp->seqno++;
-
- return 0;
-}
-
-void ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs)
-{
- struct ast_format_list current_format_old, current_format_new;
-
- /* if no packets have been sent through this session yet, then
- * changing preferences does not require any extra work
- */
- if (rtp->lasttxformat == 0) {
- rtp->pref = *prefs;
- return;
- }
-
- current_format_old = ast_codec_pref_getsize(&rtp->pref, rtp->lasttxformat);
-
- rtp->pref = *prefs;
-
- current_format_new = ast_codec_pref_getsize(&rtp->pref, rtp->lasttxformat);
-
- /* if the framing desired for the current format has changed, we may have to create
- * or adjust the smoother for this session
- */
- if ((current_format_new.inc_ms != 0) &&
- (current_format_new.cur_ms != current_format_old.cur_ms)) {
- int new_size = (current_format_new.cur_ms * current_format_new.fr_len) / current_format_new.inc_ms;
-
- if (rtp->smoother) {
- ast_smoother_reconfigure(rtp->smoother, new_size);
- if (option_debug) {
- ast_log(LOG_DEBUG, "Adjusted smoother to %d ms and %d bytes\n", current_format_new.cur_ms, new_size);
- }
- } else {
- if (!(rtp->smoother = ast_smoother_new(new_size))) {
- ast_log(LOG_WARNING, "Unable to create smoother: format: %d ms: %d len: %d\n", rtp->lasttxformat, current_format_new.cur_ms, new_size);
- return;
- }
- if (current_format_new.flags) {
- ast_smoother_set_flags(rtp->smoother, current_format_new.flags);
- }
- if (option_debug) {
- ast_log(LOG_DEBUG, "Created smoother: format: %d ms: %d len: %d\n", rtp->lasttxformat, current_format_new.cur_ms, new_size);
- }
- }
- }
-
-}
-
-struct ast_codec_pref *ast_rtp_codec_getpref(struct ast_rtp *rtp)
-{
- return &rtp->pref;
-}
-
-int ast_rtp_codec_getformat(int pt)
-{
- if (pt < 0 || pt > MAX_RTP_PT)
- return 0; /* bogus payload type */
-
- if (static_RTP_PT[pt].isAstFormat)
- return static_RTP_PT[pt].code;
- else
- return 0;
-}
-
-int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
-{
- struct ast_frame *f;
- int codec;
- int hdrlen = 12;
- int subclass;
-
-
- /* If we have no peer, return immediately */
- if (!rtp->them.sin_addr.s_addr)
- return 0;
-
- /* If there is no data length, return immediately */
- if (!_f->datalen && !rtp->red)
- return 0;
-
- /* Make sure we have enough space for RTP header */
- if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO) && (_f->frametype != AST_FRAME_TEXT)) {
- ast_log(LOG_WARNING, "RTP can only send voice, video and text\n");
- return -1;
- }
-
- if (rtp->red) {
- /* return 0; */
- /* no primary data or generations to send */
- if ((_f = red_t140_to_red(rtp->red)) == NULL)
- return 0;
- }
-
- /* The bottom bit of a video subclass contains the marker bit */
- subclass = _f->subclass;
- if (_f->frametype == AST_FRAME_VIDEO)
- subclass &= ~0x1;
-
- codec = ast_rtp_lookup_code(rtp, 1, subclass);
- if (codec < 0) {
- ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(_f->subclass));
- return -1;
- }
-
- if (rtp->lasttxformat != subclass) {
- /* New format, reset the smoother */
- ast_debug(1, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass));
- rtp->lasttxformat = subclass;
- if (rtp->smoother)
- ast_smoother_free(rtp->smoother);
- rtp->smoother = NULL;
- }
-
- if (!rtp->smoother) {
- struct ast_format_list fmt = ast_codec_pref_getsize(&rtp->pref, subclass);
-
- switch (subclass) {
- case AST_FORMAT_SPEEX:
- case AST_FORMAT_G723_1:
- case AST_FORMAT_SIREN7:
- case AST_FORMAT_SIREN14:
- /* these are all frame-based codecs and cannot be safely run through
- a smoother */
- break;
- default:
- if (fmt.inc_ms) { /* if codec parameters is set / avoid division by zero */
- if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) {
- ast_log(LOG_WARNING, "Unable to create smoother: format: %d ms: %d len: %d\n", subclass, fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms));
- return -1;
- }
- if (fmt.flags)
- ast_smoother_set_flags(rtp->smoother, fmt.flags);
- ast_debug(1, "Created smoother: format: %d ms: %d len: %d\n", subclass, fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms));
- }
- }
- }
- if (rtp->smoother) {
- if (ast_smoother_test_flag(rtp->smoother, AST_SMOOTHER_FLAG_BE)) {
- ast_smoother_feed_be(rtp->smoother, _f);
- } else {
- ast_smoother_feed(rtp->smoother, _f);
- }
-
- while ((f = ast_smoother_read(rtp->smoother)) && (f->data.ptr)) {
- if (f->subclass == AST_FORMAT_G722) {
- /* G.722 is silllllllllllllly */
- f->samples /= 2;
- }
-
- ast_rtp_raw_write(rtp, f, codec);
- }
- } else {
- /* Don't buffer outgoing frames; send them one-per-packet: */
- if (_f->offset < hdrlen)
- f = ast_frdup(_f); /*! \bug XXX this might never be free'd. Why do we do this? */
- else
- f = _f;
- if (f->data.ptr)
- ast_rtp_raw_write(rtp, f, codec);
- if (f != _f)
- ast_frfree(f);
- }
-
- return 0;
-}
-
-/*! \brief Unregister interface to channel driver */
-void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto)
-{
- AST_RWLIST_WRLOCK(&protos);
- AST_RWLIST_REMOVE(&protos, proto, list);
- AST_RWLIST_UNLOCK(&protos);
-}
-
-/*! \brief Register interface to channel driver */
-int ast_rtp_proto_register(struct ast_rtp_protocol *proto)
-{
- struct ast_rtp_protocol *cur;
-
- AST_RWLIST_WRLOCK(&protos);
- AST_RWLIST_TRAVERSE(&protos, cur, list) {
- if (!strcmp(cur->type, proto->type)) {
- ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type);
- AST_RWLIST_UNLOCK(&protos);
- return -1;
- }
- }
- AST_RWLIST_INSERT_HEAD(&protos, proto, list);
- AST_RWLIST_UNLOCK(&protos);
-
- return 0;
-}
-
-/*! \brief Bridge loop for true native bridge (reinvite) */
-static enum ast_bridge_result bridge_native_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp *p0, struct ast_rtp *p1, struct ast_rtp *vp0, struct ast_rtp *vp1, struct ast_rtp *tp0, struct ast_rtp *tp1, struct ast_rtp_protocol *pr0, struct ast_rtp_protocol *pr1, int codec0, int codec1, int timeoutms, int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1)
-{
- struct ast_frame *fr = NULL;
- struct ast_channel *who = NULL, *other = NULL, *cs[3] = {NULL, };
- int oldcodec0 = codec0, oldcodec1 = codec1;
- struct sockaddr_in ac1 = {0,}, vac1 = {0,}, tac1 = {0,}, ac0 = {0,}, vac0 = {0,}, tac0 = {0,};
- struct sockaddr_in t1 = {0,}, vt1 = {0,}, tt1 = {0,}, t0 = {0,}, vt0 = {0,}, tt0 = {0,};
-
- /* Set it up so audio goes directly between the two endpoints */
-
- /* Test the first channel */
- if (!(pr0->set_rtp_peer(c0, p1, vp1, tp1, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE)))) {
- ast_rtp_get_peer(p1, &ac1);
- if (vp1)
- ast_rtp_get_peer(vp1, &vac1);
- if (tp1)
- ast_rtp_get_peer(tp1, &tac1);
- } else
- ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name);
-
- /* Test the second channel */
- if (!(pr1->set_rtp_peer(c1, p0, vp0, tp0, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE)))) {
- ast_rtp_get_peer(p0, &ac0);
- if (vp0)
- ast_rtp_get_peer(vp0, &vac0);
- if (tp0)
- ast_rtp_get_peer(tp0, &tac0);
- } else
- ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c1->name, c0->name);
-
- /* Now we can unlock and move into our loop */
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
-
- ast_poll_channel_add(c0, c1);
-
- /* Throw our channels into the structure and enter the loop */
- cs[0] = c0;
- cs[1] = c1;
- cs[2] = NULL;
- for (;;) {
- /* Check if anything changed */
- if ((c0->tech_pvt != pvt0) ||
- (c1->tech_pvt != pvt1) ||
- (c0->masq || c0->masqr || c1->masq || c1->masqr) ||
- (c0->monitor || c0->audiohooks || c1->monitor || c1->audiohooks)) {
- ast_debug(1, "Oooh, something is weird, backing out\n");
- if (c0->tech_pvt == pvt0)
- if (pr0->set_rtp_peer(c0, NULL, NULL, NULL, 0, 0))
- ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
- if (c1->tech_pvt == pvt1)
- if (pr1->set_rtp_peer(c1, NULL, NULL, NULL, 0, 0))
- ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
- ast_poll_channel_del(c0, c1);
- return AST_BRIDGE_RETRY;
- }
-
- /* Check if they have changed their address */
- ast_rtp_get_peer(p1, &t1);
- if (vp1)
- ast_rtp_get_peer(vp1, &vt1);
- if (tp1)
- ast_rtp_get_peer(tp1, &tt1);
- if (pr1->get_codec)
- codec1 = pr1->get_codec(c1);
- ast_rtp_get_peer(p0, &t0);
- if (vp0)
- ast_rtp_get_peer(vp0, &vt0);
- if (tp0)
- ast_rtp_get_peer(tp0, &tt0);
- if (pr0->get_codec)
- codec0 = pr0->get_codec(c0);
- if ((inaddrcmp(&t1, &ac1)) ||
- (vp1 && inaddrcmp(&vt1, &vac1)) ||
- (tp1 && inaddrcmp(&tt1, &tac1)) ||
- (codec1 != oldcodec1)) {
- ast_debug(2, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
- c1->name, ast_inet_ntoa(t1.sin_addr), ntohs(t1.sin_port), codec1);
- ast_debug(2, "Oooh, '%s' changed end vaddress to %s:%d (format %d)\n",
- c1->name, ast_inet_ntoa(vt1.sin_addr), ntohs(vt1.sin_port), codec1);
- ast_debug(2, "Oooh, '%s' changed end taddress to %s:%d (format %d)\n",
- c1->name, ast_inet_ntoa(tt1.sin_addr), ntohs(tt1.sin_port), codec1);
- ast_debug(2, "Oooh, '%s' was %s:%d/(format %d)\n",
- c1->name, ast_inet_ntoa(ac1.sin_addr), ntohs(ac1.sin_port), oldcodec1);
- ast_debug(2, "Oooh, '%s' was %s:%d/(format %d)\n",
- c1->name, ast_inet_ntoa(vac1.sin_addr), ntohs(vac1.sin_port), oldcodec1);
- ast_debug(2, "Oooh, '%s' was %s:%d/(format %d)\n",
- c1->name, ast_inet_ntoa(tac1.sin_addr), ntohs(tac1.sin_port), oldcodec1);
- if (pr0->set_rtp_peer(c0, t1.sin_addr.s_addr ? p1 : NULL, vt1.sin_addr.s_addr ? vp1 : NULL, tt1.sin_addr.s_addr ? tp1 : NULL, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE)))
- ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c0->name, c1->name);
- memcpy(&ac1, &t1, sizeof(ac1));
- memcpy(&vac1, &vt1, sizeof(vac1));
- memcpy(&tac1, &tt1, sizeof(tac1));
- oldcodec1 = codec1;
- }
- if ((inaddrcmp(&t0, &ac0)) ||
- (vp0 && inaddrcmp(&vt0, &vac0)) ||
- (tp0 && inaddrcmp(&tt0, &tac0))) {
- ast_debug(2, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
- c0->name, ast_inet_ntoa(t0.sin_addr), ntohs(t0.sin_port), codec0);
- ast_debug(2, "Oooh, '%s' was %s:%d/(format %d)\n",
- c0->name, ast_inet_ntoa(ac0.sin_addr), ntohs(ac0.sin_port), oldcodec0);
- if (pr1->set_rtp_peer(c1, t0.sin_addr.s_addr ? p0 : NULL, vt0.sin_addr.s_addr ? vp0 : NULL, tt0.sin_addr.s_addr ? tp0 : NULL, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE)))
- ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c1->name, c0->name);
- memcpy(&ac0, &t0, sizeof(ac0));
- memcpy(&vac0, &vt0, sizeof(vac0));
- memcpy(&tac0, &tt0, sizeof(tac0));
- oldcodec0 = codec0;
- }
-
- /* Wait for frame to come in on the channels */
- if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) {
- if (!timeoutms) {
- if (pr0->set_rtp_peer(c0, NULL, NULL, NULL, 0, 0))
- ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
- if (pr1->set_rtp_peer(c1, NULL, NULL, NULL, 0, 0))
- ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
- return AST_BRIDGE_RETRY;
- }
- ast_debug(1, "Ooh, empty read...\n");
- if (ast_check_hangup(c0) || ast_check_hangup(c1))
- break;
- continue;
- }
- fr = ast_read(who);
- other = (who == c0) ? c1 : c0;
- if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) &&
- (((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) ||
- ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) {
- /* Break out of bridge */
- *fo = fr;
- *rc = who;
- ast_debug(1, "Oooh, got a %s\n", fr ? "digit" : "hangup");
- if (c0->tech_pvt == pvt0)
- if (pr0->set_rtp_peer(c0, NULL, NULL, NULL, 0, 0))
- ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
- if (c1->tech_pvt == pvt1)
- if (pr1->set_rtp_peer(c1, NULL, NULL, NULL, 0, 0))
- ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
- ast_poll_channel_del(c0, c1);
- return AST_BRIDGE_COMPLETE;
- } else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
- if ((fr->subclass == AST_CONTROL_HOLD) ||
- (fr->subclass == AST_CONTROL_UNHOLD) ||
- (fr->subclass == AST_CONTROL_VIDUPDATE) ||
- (fr->subclass == AST_CONTROL_T38) ||
- (fr->subclass == AST_CONTROL_SRCUPDATE)) {
- if (fr->subclass == AST_CONTROL_HOLD) {
- /* If we someone went on hold we want the other side to reinvite back to us */
- if (who == c0)
- pr1->set_rtp_peer(c1, NULL, NULL, NULL, 0, 0);
- else
- pr0->set_rtp_peer(c0, NULL, NULL, NULL, 0, 0);
- } else if (fr->subclass == AST_CONTROL_UNHOLD) {
- /* If they went off hold they should go back to being direct */
- if (who == c0)
- pr1->set_rtp_peer(c1, p0, vp0, tp0, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE));
- else
- pr0->set_rtp_peer(c0, p1, vp1, tp1, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE));
- }
- /* Update local address information */
- ast_rtp_get_peer(p0, &t0);
- memcpy(&ac0, &t0, sizeof(ac0));
- ast_rtp_get_peer(p1, &t1);
- memcpy(&ac1, &t1, sizeof(ac1));
- /* Update codec information */
- if (pr0->get_codec && c0->tech_pvt)
- oldcodec0 = codec0 = pr0->get_codec(c0);
- if (pr1->get_codec && c1->tech_pvt)
- oldcodec1 = codec1 = pr1->get_codec(c1);
- ast_indicate_data(other, fr->subclass, fr->data.ptr, fr->datalen);
- ast_frfree(fr);
- } else {
- *fo = fr;
- *rc = who;
- ast_debug(1, "Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass, who->name);
- return AST_BRIDGE_COMPLETE;
- }
- } else {
- if ((fr->frametype == AST_FRAME_DTMF_BEGIN) ||
- (fr->frametype == AST_FRAME_DTMF_END) ||
- (fr->frametype == AST_FRAME_VOICE) ||
- (fr->frametype == AST_FRAME_VIDEO) ||
- (fr->frametype == AST_FRAME_IMAGE) ||
- (fr->frametype == AST_FRAME_HTML) ||
- (fr->frametype == AST_FRAME_MODEM) ||
- (fr->frametype == AST_FRAME_TEXT)) {
- ast_write(other, fr);
- }
- ast_frfree(fr);
- }
- /* Swap priority */
-#ifndef HAVE_EPOLL
- cs[2] = cs[0];
- cs[0] = cs[1];
- cs[1] = cs[2];
-#endif
- }
-
- ast_poll_channel_del(c0, c1);
-
- if (pr0->set_rtp_peer(c0, NULL, NULL, NULL, 0, 0))
- ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
- if (pr1->set_rtp_peer(c1, NULL, NULL, NULL, 0, 0))
- ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
-
- return AST_BRIDGE_FAILED;
-}
-
-/*! \brief P2P RTP Callback */
-#ifdef P2P_INTENSE
-static int p2p_rtp_callback(int *id, int fd, short events, void *cbdata)
-{
- int res = 0, hdrlen = 12;
- struct sockaddr_in sin;
- socklen_t len;
- unsigned int *header;
- struct ast_rtp *rtp = cbdata, *bridged = NULL;
-
- if (!rtp)
- return 1;
-
- len = sizeof(sin);
- if ((res = recvfrom(fd, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET, 0, (struct sockaddr *)&sin, &len)) < 0)
- return 1;
-
- header = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET);
-
- /* If NAT support is turned on, then see if we need to change their address */
- if ((rtp->nat) &&
- ((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
- (rtp->them.sin_port != sin.sin_port))) {
- rtp->them = sin;
- rtp->rxseqno = 0;
- ast_set_flag(rtp, FLAG_NAT_ACTIVE);
- if (option_debug || rtpdebug)
- ast_debug(0, "P2P RTP NAT: Got audio from other end. Now sending to address %s:%d\n", ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
- }
-
- /* Write directly out to other RTP stream if bridged */
- if ((bridged = ast_rtp_get_bridged(rtp)))
- bridge_p2p_rtp_write(rtp, bridged, header, res, hdrlen);
-
- return 1;
-}
-
-/*! \brief Helper function to switch a channel and RTP stream into callback mode */
-static int p2p_callback_enable(struct ast_channel *chan, struct ast_rtp *rtp, int **iod)
-{
- /* If we need DTMF, are looking for STUN, or we have no IO structure then we can't do direct callback */
- if (ast_test_flag(rtp, FLAG_P2P_NEED_DTMF) || ast_test_flag(rtp, FLAG_HAS_STUN) || !rtp->io)
- return 0;
-
- /* If the RTP structure is already in callback mode, remove it temporarily */
- if (rtp->ioid) {
- ast_io_remove(rtp->io, rtp->ioid);
- rtp->ioid = NULL;
- }
-
- /* Steal the file descriptors from the channel */
- chan->fds[0] = -1;
-
- /* Now, fire up callback mode */
- iod[0] = ast_io_add(rtp->io, ast_rtp_fd(rtp), p2p_rtp_callback, AST_IO_IN, rtp);
-
- return 1;
-}
-#else
-static int p2p_callback_enable(struct ast_channel *chan, struct ast_rtp *rtp, int **iod)
-{
- return 0;
-}
-#endif
-
-/*! \brief Helper function to switch a channel and RTP stream out of callback mode */
-static int p2p_callback_disable(struct ast_channel *chan, struct ast_rtp *rtp, int **iod)
-{
- ast_channel_lock(chan);
-
- /* Remove the callback from the IO context */
- ast_io_remove(rtp->io, iod[0]);
-
- /* Restore file descriptors */
- chan->fds[0] = ast_rtp_fd(rtp);
- ast_channel_unlock(chan);
-
- /* Restore callback mode if previously used */
- if (ast_test_flag(rtp, FLAG_CALLBACK_MODE))
- rtp->ioid = ast_io_add(rtp->io, ast_rtp_fd(rtp), rtpread, AST_IO_IN, rtp);
-
- return 0;
-}
-
-/*! \brief Helper function that sets what an RTP structure is bridged to */
-static void p2p_set_bridge(struct ast_rtp *rtp0, struct ast_rtp *rtp1)
-{
- rtp_bridge_lock(rtp0);
- rtp0->bridged = rtp1;
- rtp_bridge_unlock(rtp0);
-}
-
-/*! \brief Bridge loop for partial native bridge (packet2packet)
-
- In p2p mode, Asterisk is a very basic RTP proxy, just forwarding whatever
- rtp/rtcp we get in to the channel.
- \note this currently only works for Audio
-*/
-static enum ast_bridge_result bridge_p2p_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp *p0, struct ast_rtp *p1, int timeoutms, int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1)
-{
- struct ast_frame *fr = NULL;
- struct ast_channel *who = NULL, *other = NULL, *cs[3] = {NULL, };
- int *p0_iod[2] = {NULL, NULL}, *p1_iod[2] = {NULL, NULL};
- int p0_callback = 0, p1_callback = 0;
- enum ast_bridge_result res = AST_BRIDGE_FAILED;
-
- /* Okay, setup each RTP structure to do P2P forwarding */
- ast_clear_flag(p0, FLAG_P2P_SENT_MARK);
- p2p_set_bridge(p0, p1);
- ast_clear_flag(p1, FLAG_P2P_SENT_MARK);
- p2p_set_bridge(p1, p0);
-
- /* Activate callback modes if possible */
- p0_callback = p2p_callback_enable(c0, p0, &p0_iod[0]);
- p1_callback = p2p_callback_enable(c1, p1, &p1_iod[0]);
-
- /* Now let go of the channel locks and be on our way */
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
-
- ast_poll_channel_add(c0, c1);
-
- /* Go into a loop forwarding frames until we don't need to anymore */
- cs[0] = c0;
- cs[1] = c1;
- cs[2] = NULL;
- for (;;) {
- /* If the underlying formats have changed force this bridge to break */
- if ((c0->rawreadformat != c1->rawwriteformat) || (c1->rawreadformat != c0->rawwriteformat)) {
- ast_debug(3, "p2p-rtp-bridge: Oooh, formats changed, backing out\n");
- res = AST_BRIDGE_FAILED_NOWARN;
- break;
- }
- /* Check if anything changed */
- if ((c0->tech_pvt != pvt0) ||
- (c1->tech_pvt != pvt1) ||
- (c0->masq || c0->masqr || c1->masq || c1->masqr) ||
- (c0->monitor || c0->audiohooks || c1->monitor || c1->audiohooks)) {
- ast_debug(3, "p2p-rtp-bridge: Oooh, something is weird, backing out\n");
- /* If a masquerade needs to happen we have to try to read in a frame so that it actually happens. Without this we risk being called again and going into a loop */
- if ((c0->masq || c0->masqr) && (fr = ast_read(c0)))
- ast_frfree(fr);
- if ((c1->masq || c1->masqr) && (fr = ast_read(c1)))
- ast_frfree(fr);
- res = AST_BRIDGE_RETRY;
- break;
- }
- /* Wait on a channel to feed us a frame */
- if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) {
- if (!timeoutms) {
- res = AST_BRIDGE_RETRY;
- break;
- }
- if (option_debug > 2)
- ast_log(LOG_NOTICE, "p2p-rtp-bridge: Ooh, empty read...\n");
- if (ast_check_hangup(c0) || ast_check_hangup(c1))
- break;
- continue;
- }
- /* Read in frame from channel */
- fr = ast_read(who);
- other = (who == c0) ? c1 : c0;
- /* Depending on the frame we may need to break out of our bridge */
- if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) &&
- ((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) |
- ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)))) {
- /* Record received frame and who */
- *fo = fr;
- *rc = who;
- ast_debug(3, "p2p-rtp-bridge: Ooh, got a %s\n", fr ? "digit" : "hangup");
- res = AST_BRIDGE_COMPLETE;
- break;
- } else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
- if ((fr->subclass == AST_CONTROL_HOLD) ||
- (fr->subclass == AST_CONTROL_UNHOLD) ||
- (fr->subclass == AST_CONTROL_VIDUPDATE) ||
- (fr->subclass == AST_CONTROL_T38) ||
- (fr->subclass == AST_CONTROL_SRCUPDATE)) {
- /* If we are going on hold, then break callback mode and P2P bridging */
- if (fr->subclass == AST_CONTROL_HOLD) {
- if (p0_callback)
- p0_callback = p2p_callback_disable(c0, p0, &p0_iod[0]);
- if (p1_callback)
- p1_callback = p2p_callback_disable(c1, p1, &p1_iod[0]);
- p2p_set_bridge(p0, NULL);
- p2p_set_bridge(p1, NULL);
- } else if (fr->subclass == AST_CONTROL_UNHOLD) {
- /* If we are off hold, then go back to callback mode and P2P bridging */
- ast_clear_flag(p0, FLAG_P2P_SENT_MARK);
- p2p_set_bridge(p0, p1);
- ast_clear_flag(p1, FLAG_P2P_SENT_MARK);
- p2p_set_bridge(p1, p0);
- p0_callback = p2p_callback_enable(c0, p0, &p0_iod[0]);
- p1_callback = p2p_callback_enable(c1, p1, &p1_iod[0]);
- }
- ast_indicate_data(other, fr->subclass, fr->data.ptr, fr->datalen);
- ast_frfree(fr);
- } else {
- *fo = fr;
- *rc = who;
- ast_debug(3, "p2p-rtp-bridge: Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass, who->name);
- res = AST_BRIDGE_COMPLETE;
- break;
- }
- } else {
- if ((fr->frametype == AST_FRAME_DTMF_BEGIN) ||
- (fr->frametype == AST_FRAME_DTMF_END) ||
- (fr->frametype == AST_FRAME_VOICE) ||
- (fr->frametype == AST_FRAME_VIDEO) ||
- (fr->frametype == AST_FRAME_IMAGE) ||
- (fr->frametype == AST_FRAME_HTML) ||
- (fr->frametype == AST_FRAME_MODEM) ||
- (fr->frametype == AST_FRAME_TEXT)) {
- ast_write(other, fr);
- }
-
- ast_frfree(fr);
- }
- /* Swap priority */
-#ifndef HAVE_EPOLL
- cs[2] = cs[0];
- cs[0] = cs[1];
- cs[1] = cs[2];
-#endif
- }
-
- /* If we are totally avoiding the core, then restore our link to it */
- if (p0_callback)
- p0_callback = p2p_callback_disable(c0, p0, &p0_iod[0]);
- if (p1_callback)
- p1_callback = p2p_callback_disable(c1, p1, &p1_iod[0]);
-
- /* Break out of the direct bridge */
- p2p_set_bridge(p0, NULL);
- p2p_set_bridge(p1, NULL);
-
- ast_poll_channel_del(c0, c1);
-
- return res;
-}
-
-/*! \page AstRTPbridge The Asterisk RTP bridge
- The RTP bridge is called from the channel drivers that are using the RTP
- subsystem in Asterisk - like SIP, H.323 and Jingle/Google Talk.
-
- This bridge aims to offload the Asterisk server by setting up
- the media stream directly between the endpoints, keeping the
- signalling in Asterisk.
-
- It checks with the channel driver, using a callback function, if
- there are possibilities for a remote bridge.
-
- If this fails, the bridge hands off to the core bridge. Reasons
- can be NAT support needed, DTMF features in audio needed by
- the PBX for transfers or spying/monitoring on channels.
-
- If transcoding is needed - we can't do a remote bridge.
- If only NAT support is needed, we're using Asterisk in
- RTP proxy mode with the p2p RTP bridge, basically
- forwarding incoming audio packets to the outbound
- stream on a network level.
-
- References:
- - ast_rtp_bridge()
- - ast_channel_early_bridge()
- - ast_channel_bridge()
- - rtp.c
- - rtp.h
-*/
-/*! \brief Bridge calls. If possible and allowed, initiate
- re-invite so the peers exchange media directly outside
- of Asterisk.
-*/
-enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms)
-{
- struct ast_rtp *p0 = NULL, *p1 = NULL; /* Audio RTP Channels */
- struct ast_rtp *vp0 = NULL, *vp1 = NULL; /* Video RTP channels */
- struct ast_rtp *tp0 = NULL, *tp1 = NULL; /* Text RTP channels */
- struct ast_rtp_protocol *pr0 = NULL, *pr1 = NULL;
- enum ast_rtp_get_result audio_p0_res = AST_RTP_GET_FAILED, video_p0_res = AST_RTP_GET_FAILED, text_p0_res = AST_RTP_GET_FAILED;
- enum ast_rtp_get_result audio_p1_res = AST_RTP_GET_FAILED, video_p1_res = AST_RTP_GET_FAILED, text_p1_res = AST_RTP_GET_FAILED;
- enum ast_bridge_result res = AST_BRIDGE_FAILED;
- int codec0 = 0, codec1 = 0;
- void *pvt0 = NULL, *pvt1 = NULL;
-
- /* Lock channels */
- ast_channel_lock(c0);
- while (ast_channel_trylock(c1)) {
- ast_channel_unlock(c0);
- usleep(1);
- ast_channel_lock(c0);
- }
-
- /* Ensure neither channel got hungup during lock avoidance */
- if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
- ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", c0->name, c1->name);
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
- return AST_BRIDGE_FAILED;
- }
-
- /* Find channel driver interfaces */
- if (!(pr0 = get_proto(c0))) {
- ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name);
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
- return AST_BRIDGE_FAILED;
- }
- if (!(pr1 = get_proto(c1))) {
- ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name);
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
- return AST_BRIDGE_FAILED;
- }
-
- /* Get channel specific interface structures */
- pvt0 = c0->tech_pvt;
- pvt1 = c1->tech_pvt;
-
- /* Get audio and video interface (if native bridge is possible) */
- audio_p0_res = pr0->get_rtp_info(c0, &p0);
- video_p0_res = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0, &vp0) : AST_RTP_GET_FAILED;
- text_p0_res = pr0->get_trtp_info ? pr0->get_trtp_info(c0, &vp0) : AST_RTP_GET_FAILED;
- audio_p1_res = pr1->get_rtp_info(c1, &p1);
- video_p1_res = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1, &vp1) : AST_RTP_GET_FAILED;
- text_p1_res = pr1->get_trtp_info ? pr1->get_trtp_info(c1, &vp1) : AST_RTP_GET_FAILED;
-
- /* If we are carrying video, and both sides are not reinviting... then fail the native bridge */
- if (video_p0_res != AST_RTP_GET_FAILED && (audio_p0_res != AST_RTP_TRY_NATIVE || video_p0_res != AST_RTP_TRY_NATIVE))
- audio_p0_res = AST_RTP_GET_FAILED;
- if (video_p1_res != AST_RTP_GET_FAILED && (audio_p1_res != AST_RTP_TRY_NATIVE || video_p1_res != AST_RTP_TRY_NATIVE))
- audio_p1_res = AST_RTP_GET_FAILED;
-
- /* Check if a bridge is possible (partial/native) */
- if (audio_p0_res == AST_RTP_GET_FAILED || audio_p1_res == AST_RTP_GET_FAILED) {
- /* Somebody doesn't want to play... */
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
- return AST_BRIDGE_FAILED_NOWARN;
- }
-
- /* If we need to feed DTMF frames into the core then only do a partial native bridge */
- if (ast_test_flag(p0, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) {
- ast_set_flag(p0, FLAG_P2P_NEED_DTMF);
- audio_p0_res = AST_RTP_TRY_PARTIAL;
- }
-
- if (ast_test_flag(p1, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)) {
- ast_set_flag(p1, FLAG_P2P_NEED_DTMF);
- audio_p1_res = AST_RTP_TRY_PARTIAL;
- }
-
- /* If both sides are not using the same method of DTMF transmission
- * (ie: one is RFC2833, other is INFO... then we can not do direct media.
- * --------------------------------------------------
- * | DTMF Mode | HAS_DTMF | Accepts Begin Frames |
- * |-----------|------------|-----------------------|
- * | Inband | False | True |
- * | RFC2833 | True | True |
- * | SIP INFO | False | False |
- * --------------------------------------------------
- * However, if DTMF from both channels is being monitored by the core, then
- * we can still do packet-to-packet bridging, because passing through the
- * core will handle DTMF mode translation.
- */
- if ((ast_test_flag(p0, FLAG_HAS_DTMF) != ast_test_flag(p1, FLAG_HAS_DTMF)) ||
- (!c0->tech->send_digit_begin != !c1->tech->send_digit_begin)) {
- if (!ast_test_flag(p0, FLAG_P2P_NEED_DTMF) || !ast_test_flag(p1, FLAG_P2P_NEED_DTMF)) {
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
- return AST_BRIDGE_FAILED_NOWARN;
- }
- audio_p0_res = AST_RTP_TRY_PARTIAL;
- audio_p1_res = AST_RTP_TRY_PARTIAL;
- }
-
- /* If we need to feed frames into the core don't do a P2P bridge */
- if ((audio_p0_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p0, FLAG_P2P_NEED_DTMF)) ||
- (audio_p1_res == AST_RTP_TRY_PARTIAL && ast_test_flag(p1, FLAG_P2P_NEED_DTMF))) {
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
- return AST_BRIDGE_FAILED_NOWARN;
- }
-
- /* Get codecs from both sides */
- codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0;
- codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0;
- if (codec0 && codec1 && !(codec0 & codec1)) {
- /* Hey, we can't do native bridging if both parties speak different codecs */
- ast_debug(3, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
- return AST_BRIDGE_FAILED_NOWARN;
- }
-
- /* If either side can only do a partial bridge, then don't try for a true native bridge */
- if (audio_p0_res == AST_RTP_TRY_PARTIAL || audio_p1_res == AST_RTP_TRY_PARTIAL) {
- struct ast_format_list fmt0, fmt1;
-
- /* In order to do Packet2Packet bridging both sides must be in the same rawread/rawwrite */
- if (c0->rawreadformat != c1->rawwriteformat || c1->rawreadformat != c0->rawwriteformat) {
- ast_debug(1, "Cannot packet2packet bridge - raw formats are incompatible\n");
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
- return AST_BRIDGE_FAILED_NOWARN;
- }
- /* They must also be using the same packetization */
- fmt0 = ast_codec_pref_getsize(&p0->pref, c0->rawreadformat);
- fmt1 = ast_codec_pref_getsize(&p1->pref, c1->rawreadformat);
- if (fmt0.cur_ms != fmt1.cur_ms) {
- ast_debug(1, "Cannot packet2packet bridge - packetization settings prevent it\n");
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
- return AST_BRIDGE_FAILED_NOWARN;
- }
-
- ast_verb(3, "Packet2Packet bridging %s and %s\n", c0->name, c1->name);
- res = bridge_p2p_loop(c0, c1, p0, p1, timeoutms, flags, fo, rc, pvt0, pvt1);
- } else {
- ast_verb(3, "Native bridging %s and %s\n", c0->name, c1->name);
- res = bridge_native_loop(c0, c1, p0, p1, vp0, vp1, tp0, tp1, pr0, pr1, codec0, codec1, timeoutms, flags, fo, rc, pvt0, pvt1);
- }
-
- return res;
-}
-
-static char *rtp_do_debug_ip(struct ast_cli_args *a)
-{
- struct hostent *hp;
- struct ast_hostent ahp;
- int port = 0;
- char *p, *arg;
-
- arg = a->argv[3];
- p = strstr(arg, ":");
- if (p) {
- *p = '\0';
- p++;
- port = atoi(p);
- }
- hp = ast_gethostbyname(arg, &ahp);
- if (hp == NULL) {
- ast_cli(a->fd, "Lookup failed for '%s'\n", arg);
- return CLI_FAILURE;
- }
- rtpdebugaddr.sin_family = AF_INET;
- memcpy(&rtpdebugaddr.sin_addr, hp->h_addr, sizeof(rtpdebugaddr.sin_addr));
- rtpdebugaddr.sin_port = htons(port);
- if (port == 0)
- ast_cli(a->fd, "RTP Debugging Enabled for IP: %s\n", ast_inet_ntoa(rtpdebugaddr.sin_addr));
- else
- ast_cli(a->fd, "RTP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(rtpdebugaddr.sin_addr), port);
- rtpdebug = 1;
- return CLI_SUCCESS;
-}
-
-static char *rtcp_do_debug_ip(struct ast_cli_args *a)
-{
- struct hostent *hp;
- struct ast_hostent ahp;
- int port = 0;
- char *p, *arg;
-
- arg = a->argv[3];
- p = strstr(arg, ":");
- if (p) {
- *p = '\0';
- p++;
- port = atoi(p);
- }
- hp = ast_gethostbyname(arg, &ahp);
- if (hp == NULL) {
- ast_cli(a->fd, "Lookup failed for '%s'\n", arg);
- return CLI_FAILURE;
- }
- rtcpdebugaddr.sin_family = AF_INET;
- memcpy(&rtcpdebugaddr.sin_addr, hp->h_addr, sizeof(rtcpdebugaddr.sin_addr));
- rtcpdebugaddr.sin_port = htons(port);
- if (port == 0)
- ast_cli(a->fd, "RTCP Debugging Enabled for IP: %s\n", ast_inet_ntoa(rtcpdebugaddr.sin_addr));
- else
- ast_cli(a->fd, "RTCP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(rtcpdebugaddr.sin_addr), port);
- rtcpdebug = 1;
- return CLI_SUCCESS;
-}
-
-static char *handle_cli_rtp_set_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- switch (cmd) {
- case CLI_INIT:
- e->command = "rtp set debug {on|off|ip}";
- e->usage =
- "Usage: rtp set debug {on|off|ip host[:port]}\n"
- " Enable/Disable dumping of all RTP packets. If 'ip' is\n"
- " specified, limit the dumped packets to those to and from\n"
- " the specified 'host' with optional port.\n";
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
-
- if (a->argc == e->args) { /* set on or off */
- if (!strncasecmp(a->argv[e->args-1], "on", 2)) {
- rtpdebug = 1;
- memset(&rtpdebugaddr, 0, sizeof(rtpdebugaddr));
- ast_cli(a->fd, "RTP Debugging Enabled\n");
- return CLI_SUCCESS;
- } else if (!strncasecmp(a->argv[e->args-1], "off", 3)) {
- rtpdebug = 0;
- ast_cli(a->fd, "RTP Debugging Disabled\n");
- return CLI_SUCCESS;
- }
- } else if (a->argc == e->args +1) { /* ip */
- return rtp_do_debug_ip(a);
- }
-
- return CLI_SHOWUSAGE; /* default, failure */
-}
-
-static char *handle_cli_rtcp_set_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- switch (cmd) {
- case CLI_INIT:
- e->command = "rtcp set debug {on|off|ip}";
- e->usage =
- "Usage: rtcp set debug {on|off|ip host[:port]}\n"
- " Enable/Disable dumping of all RTCP packets. If 'ip' is\n"
- " specified, limit the dumped packets to those to and from\n"
- " the specified 'host' with optional port.\n";
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
-
- if (a->argc == e->args) { /* set on or off */
- if (!strncasecmp(a->argv[e->args-1], "on", 2)) {
- rtcpdebug = 1;
- memset(&rtcpdebugaddr, 0, sizeof(rtcpdebugaddr));
- ast_cli(a->fd, "RTCP Debugging Enabled\n");
- return CLI_SUCCESS;
- } else if (!strncasecmp(a->argv[e->args-1], "off", 3)) {
- rtcpdebug = 0;
- ast_cli(a->fd, "RTCP Debugging Disabled\n");
- return CLI_SUCCESS;
- }
- } else if (a->argc == e->args +1) { /* ip */
- return rtcp_do_debug_ip(a);
- }
-
- return CLI_SHOWUSAGE; /* default, failure */
-}
-
-static char *handle_cli_rtcp_set_stats(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- switch (cmd) {
- case CLI_INIT:
- e->command = "rtcp set stats {on|off}";
- e->usage =
- "Usage: rtcp set stats {on|off}\n"
- " Enable/Disable dumping of RTCP stats.\n";
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
-
- if (a->argc != e->args)
- return CLI_SHOWUSAGE;
-
- if (!strncasecmp(a->argv[e->args-1], "on", 2))
- rtcpstats = 1;
- else if (!strncasecmp(a->argv[e->args-1], "off", 3))
- rtcpstats = 0;
- else
- return CLI_SHOWUSAGE;
-
- ast_cli(a->fd, "RTCP Stats %s\n", rtcpstats ? "Enabled" : "Disabled");
- return CLI_SUCCESS;
-}
-
-static char *handle_cli_stun_set_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- switch (cmd) {
- case CLI_INIT:
- e->command = "stun set debug {on|off}";
- e->usage =
- "Usage: stun set debug {on|off}\n"
- " Enable/Disable STUN (Simple Traversal of UDP through NATs)\n"
- " debugging\n";
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
-
- if (a->argc != e->args)
- return CLI_SHOWUSAGE;
-
- if (!strncasecmp(a->argv[e->args-1], "on", 2))
- stundebug = 1;
- else if (!strncasecmp(a->argv[e->args-1], "off", 3))
- stundebug = 0;
- else
- return CLI_SHOWUSAGE;
-
- ast_cli(a->fd, "STUN Debugging %s\n", stundebug ? "Enabled" : "Disabled");
- return CLI_SUCCESS;
-}
-
-static struct ast_cli_entry cli_rtp[] = {
- AST_CLI_DEFINE(handle_cli_rtp_set_debug, "Enable/Disable RTP debugging"),
- AST_CLI_DEFINE(handle_cli_rtcp_set_debug, "Enable/Disable RTCP debugging"),
- AST_CLI_DEFINE(handle_cli_rtcp_set_stats, "Enable/Disable RTCP stats"),
- AST_CLI_DEFINE(handle_cli_stun_set_debug, "Enable/Disable STUN debugging"),
-};
-
-static int __ast_rtp_reload(int reload)
-{
- struct ast_config *cfg;
- const char *s;
- struct ast_flags config_flags = { reload ? CONFIG_FLAG_FILEUNCHANGED : 0 };
-
- cfg = ast_config_load2("rtp.conf", "rtp", config_flags);
- if (cfg == CONFIG_STATUS_FILEMISSING || cfg == CONFIG_STATUS_FILEUNCHANGED || cfg == CONFIG_STATUS_FILEINVALID) {
- return 0;
- }
-
- rtpstart = 5000;
- rtpend = 31000;
- dtmftimeout = DEFAULT_DTMF_TIMEOUT;
- strictrtp = STRICT_RTP_OPEN;
- if (cfg) {
- if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) {
- rtpstart = atoi(s);
- if (rtpstart < 1024)
- rtpstart = 1024;
- if (rtpstart > 65535)
- rtpstart = 65535;
- }
- if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) {
- rtpend = atoi(s);
- if (rtpend < 1024)
- rtpend = 1024;
- if (rtpend > 65535)
- rtpend = 65535;
- }
- if ((s = ast_variable_retrieve(cfg, "general", "rtcpinterval"))) {
- rtcpinterval = atoi(s);
- if (rtcpinterval == 0)
- rtcpinterval = 0; /* Just so we're clear... it's zero */
- if (rtcpinterval < RTCP_MIN_INTERVALMS)
- rtcpinterval = RTCP_MIN_INTERVALMS; /* This catches negative numbers too */
- if (rtcpinterval > RTCP_MAX_INTERVALMS)
- rtcpinterval = RTCP_MAX_INTERVALMS;
- }
- if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) {
-#ifdef SO_NO_CHECK
- if (ast_false(s))
- nochecksums = 1;
- else
- nochecksums = 0;
-#else
- if (ast_false(s))
- ast_log(LOG_WARNING, "Disabling RTP checksums is not supported on this operating system!\n");
-#endif
- }
- if ((s = ast_variable_retrieve(cfg, "general", "dtmftimeout"))) {
- dtmftimeout = atoi(s);
- if ((dtmftimeout < 0) || (dtmftimeout > 20000)) {
- ast_log(LOG_WARNING, "DTMF timeout of '%d' outside range, using default of '%d' instead\n",
- dtmftimeout, DEFAULT_DTMF_TIMEOUT);
- dtmftimeout = DEFAULT_DTMF_TIMEOUT;
- };
- }
- if ((s = ast_variable_retrieve(cfg, "general", "strictrtp"))) {
- strictrtp = ast_true(s);
- }
- ast_config_destroy(cfg);
- }
- if (rtpstart >= rtpend) {
- ast_log(LOG_WARNING, "Unreasonable values for RTP start/end port in rtp.conf\n");
- rtpstart = 5000;
- rtpend = 31000;
- }
- ast_verb(2, "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend);
- return 0;
-}
-
-int ast_rtp_reload(void)
-{
- return __ast_rtp_reload(1);
-}
-
-/*! \brief Initialize the RTP system in Asterisk */
-void ast_rtp_init(void)
-{
- ast_cli_register_multiple(cli_rtp, sizeof(cli_rtp) / sizeof(struct ast_cli_entry));
- __ast_rtp_reload(0);
-}
-
-/*! \brief Write t140 redundacy frame
- * \param data primary data to be buffered
- */
-static int red_write(const void *data)
-{
- struct ast_rtp *rtp = (struct ast_rtp*) data;
-
- ast_rtp_write(rtp, &rtp->red->t140);
-
- return 1;
-}
-
-/*! \brief Construct a redundant frame
- * \param red redundant data structure
- */
-static struct ast_frame *red_t140_to_red(struct rtp_red *red) {
- unsigned char *data = red->t140red.data.ptr;
- int len = 0;
- int i;
-
- /* replace most aged generation */
- if (red->len[0]) {
- for (i = 1; i < red->num_gen+1; i++)
- len += red->len[i];
-
- memmove(&data[red->hdrlen], &data[red->hdrlen+red->len[0]], len);
- }
-
- /* Store length of each generation and primary data length*/
- for (i = 0; i < red->num_gen; i++)
- red->len[i] = red->len[i+1];
- red->len[i] = red->t140.datalen;
-
- /* write each generation length in red header */
- len = red->hdrlen;
- for (i = 0; i < red->num_gen; i++)
- len += data[i*4+3] = red->len[i];
-
- /* add primary data to buffer */
- memcpy(&data[len], red->t140.data.ptr, red->t140.datalen);
- red->t140red.datalen = len + red->t140.datalen;
-
- /* no primary data and no generations to send */
- if (len == red->hdrlen && !red->t140.datalen)
- return NULL;
-
- /* reset t.140 buffer */
- red->t140.datalen = 0;
-
- return &red->t140red;
-}
-
-/*! \brief Initialize t140 redundancy
- * \param rtp
- * \param ti buffer t140 for ti (msecs) before sending redundant frame
- * \param red_data_pt Payloadtypes for primary- and generation-data
- * \param num_gen numbers of generations (primary generation not encounted)
- *
-*/
-int ast_rtp_red_init(struct ast_rtp *rtp, int ti, int *red_data_pt, int num_gen)
-{
- struct rtp_red *r;
- int x;
-
- if (!(r = ast_calloc(1, sizeof(struct rtp_red))))
- return -1;
-
- r->t140.frametype = AST_FRAME_TEXT;
- r->t140.subclass = AST_FORMAT_T140RED;
- r->t140.data.ptr = &r->buf_data;
-
- r->t140.ts = 0;
- r->t140red = r->t140;
- r->t140red.data.ptr = &r->t140red_data;
- r->t140red.datalen = 0;
- r->ti = ti;
- r->num_gen = num_gen;
- r->hdrlen = num_gen * 4 + 1;
- r->prev_ts = 0;
-
- for (x = 0; x < num_gen; x++) {
- r->pt[x] = red_data_pt[x];
- r->pt[x] |= 1 << 7; /* mark redundant generations pt */
- r->t140red_data[x*4] = r->pt[x];
- }
- r->t140red_data[x*4] = r->pt[x] = red_data_pt[x]; /* primary pt */
- r->schedid = ast_sched_add(rtp->sched, ti, red_write, rtp);
- rtp->red = r;
-
- r->t140.datalen = 0;
-
- return 0;
-}
-
-/*! \brief Buffer t140 from chan_sip
- * \param rtp
- * \param f frame
- */
-void ast_red_buffer_t140(struct ast_rtp *rtp, struct ast_frame *f)
-{
- if (f->datalen > -1) {
- struct rtp_red *red = rtp->red;
- memcpy(&red->buf_data[red->t140.datalen], f->data.ptr, f->datalen);
- red->t140.datalen += f->datalen;
- red->t140.ts = f->ts;
- }
-}
diff --git a/main/rtp_engine.c b/main/rtp_engine.c
new file mode 100644
index 0000000000..fd448b849b
--- /dev/null
+++ b/main/rtp_engine.c
@@ -0,0 +1,1572 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2008, Digium, Inc.
+ *
+ * Joshua Colp <jcolp@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Pluggable RTP Architecture
+ *
+ * \author Joshua Colp <jcolp@digium.com>
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include <math.h>
+
+#include "asterisk/channel.h"
+#include "asterisk/frame.h"
+#include "asterisk/module.h"
+#include "asterisk/rtp_engine.h"
+#include "asterisk/manager.h"
+#include "asterisk/options.h"
+#include "asterisk/astobj2.h"
+#include "asterisk/pbx.h"
+
+/*! Structure that represents an RTP session (instance) */
+struct ast_rtp_instance {
+ /*! Engine that is handling this RTP instance */
+ struct ast_rtp_engine *engine;
+ /*! Data unique to the RTP engine */
+ void *data;
+ /*! RTP properties that have been set and their value */
+ int properties[AST_RTP_PROPERTY_MAX];
+ /*! Address that we are expecting RTP to come in to */
+ struct sockaddr_in local_address;
+ /*! Address that we are sending RTP to */
+ struct sockaddr_in remote_address;
+ /*! Instance that we are bridged to if doing remote or local bridging */
+ struct ast_rtp_instance *bridged;
+ /*! Payload and packetization information */
+ struct ast_rtp_codecs codecs;
+ /*! RTP timeout time (negative or zero means disabled, negative value means temporarily disabled) */
+ int timeout;
+ /*! RTP timeout when on hold (negative or zero means disabled, negative value means temporarily disabled). */
+ int holdtimeout;
+ /*! DTMF mode in use */
+ enum ast_rtp_dtmf_mode dtmf_mode;
+};
+
+/*! List of RTP engines that are currently registered */
+static AST_RWLIST_HEAD_STATIC(engines, ast_rtp_engine);
+
+/*! List of RTP glues */
+static AST_RWLIST_HEAD_STATIC(glues, ast_rtp_glue);
+
+/*! The following array defines the MIME Media type (and subtype) for each
+ of our codecs, or RTP-specific data type. */
+static const struct ast_rtp_mime_type {
+ struct ast_rtp_payload_type payload_type;
+ char *type;
+ char *subtype;
+ unsigned int sample_rate;
+} ast_rtp_mime_types[] = {
+ {{1, AST_FORMAT_G723_1}, "audio", "G723", 8000},
+ {{1, AST_FORMAT_GSM}, "audio", "GSM", 8000},
+ {{1, AST_FORMAT_ULAW}, "audio", "PCMU", 8000},
+ {{1, AST_FORMAT_ULAW}, "audio", "G711U", 8000},
+ {{1, AST_FORMAT_ALAW}, "audio", "PCMA", 8000},
+ {{1, AST_FORMAT_ALAW}, "audio", "G711A", 8000},
+ {{1, AST_FORMAT_G726}, "audio", "G726-32", 8000},
+ {{1, AST_FORMAT_ADPCM}, "audio", "DVI4", 8000},
+ {{1, AST_FORMAT_SLINEAR}, "audio", "L16", 8000},
+ {{1, AST_FORMAT_LPC10}, "audio", "LPC", 8000},
+ {{1, AST_FORMAT_G729A}, "audio", "G729", 8000},
+ {{1, AST_FORMAT_G729A}, "audio", "G729A", 8000},
+ {{1, AST_FORMAT_G729A}, "audio", "G.729", 8000},
+ {{1, AST_FORMAT_SPEEX}, "audio", "speex", 8000},
+ {{1, AST_FORMAT_ILBC}, "audio", "iLBC", 8000},
+ /* this is the sample rate listed in the RTP profile for the G.722
+ codec, *NOT* the actual sample rate of the media stream
+ */
+ {{1, AST_FORMAT_G722}, "audio", "G722", 8000},
+ {{1, AST_FORMAT_G726_AAL2}, "audio", "AAL2-G726-32", 8000},
+ {{0, AST_RTP_DTMF}, "audio", "telephone-event", 8000},
+ {{0, AST_RTP_CISCO_DTMF}, "audio", "cisco-telephone-event", 8000},
+ {{0, AST_RTP_CN}, "audio", "CN", 8000},
+ {{1, AST_FORMAT_JPEG}, "video", "JPEG", 90000},
+ {{1, AST_FORMAT_PNG}, "video", "PNG", 90000},
+ {{1, AST_FORMAT_H261}, "video", "H261", 90000},
+ {{1, AST_FORMAT_H263}, "video", "H263", 90000},
+ {{1, AST_FORMAT_H263_PLUS}, "video", "h263-1998", 90000},
+ {{1, AST_FORMAT_H264}, "video", "H264", 90000},
+ {{1, AST_FORMAT_MP4_VIDEO}, "video", "MP4V-ES", 90000},
+ {{1, AST_FORMAT_T140RED}, "text", "RED", 1000},
+ {{1, AST_FORMAT_T140}, "text", "T140", 1000},
+ {{1, AST_FORMAT_SIREN7}, "audio", "G7221", 16000},
+ {{1, AST_FORMAT_SIREN14}, "audio", "G7221", 32000},
+};
+
+/*!
+ * \brief Mapping between Asterisk codecs and rtp payload types
+ *
+ * Static (i.e., well-known) RTP payload types for our "AST_FORMAT..."s:
+ * also, our own choices for dynamic payload types. This is our master
+ * table for transmission
+ *
+ * See http://www.iana.org/assignments/rtp-parameters for a list of
+ * assigned values
+ */
+static const struct ast_rtp_payload_type static_RTP_PT[AST_RTP_MAX_PT] = {
+ [0] = {1, AST_FORMAT_ULAW},
+ #ifdef USE_DEPRECATED_G726
+ [2] = {1, AST_FORMAT_G726}, /* Technically this is G.721, but if Cisco can do it, so can we... */
+ #endif
+ [3] = {1, AST_FORMAT_GSM},
+ [4] = {1, AST_FORMAT_G723_1},
+ [5] = {1, AST_FORMAT_ADPCM}, /* 8 kHz */
+ [6] = {1, AST_FORMAT_ADPCM}, /* 16 kHz */
+ [7] = {1, AST_FORMAT_LPC10},
+ [8] = {1, AST_FORMAT_ALAW},
+ [9] = {1, AST_FORMAT_G722},
+ [10] = {1, AST_FORMAT_SLINEAR}, /* 2 channels */
+ [11] = {1, AST_FORMAT_SLINEAR}, /* 1 channel */
+ [13] = {0, AST_RTP_CN},
+ [16] = {1, AST_FORMAT_ADPCM}, /* 11.025 kHz */
+ [17] = {1, AST_FORMAT_ADPCM}, /* 22.050 kHz */
+ [18] = {1, AST_FORMAT_G729A},
+ [19] = {0, AST_RTP_CN}, /* Also used for CN */
+ [26] = {1, AST_FORMAT_JPEG},
+ [31] = {1, AST_FORMAT_H261},
+ [34] = {1, AST_FORMAT_H263},
+ [97] = {1, AST_FORMAT_ILBC},
+ [98] = {1, AST_FORMAT_H263_PLUS},
+ [99] = {1, AST_FORMAT_H264},
+ [101] = {0, AST_RTP_DTMF},
+ [102] = {1, AST_FORMAT_SIREN7},
+ [103] = {1, AST_FORMAT_H263_PLUS},
+ [104] = {1, AST_FORMAT_MP4_VIDEO},
+ [105] = {1, AST_FORMAT_T140RED}, /* Real time text chat (with redundancy encoding) */
+ [106] = {1, AST_FORMAT_T140}, /* Real time text chat */
+ [110] = {1, AST_FORMAT_SPEEX},
+ [111] = {1, AST_FORMAT_G726},
+ [112] = {1, AST_FORMAT_G726_AAL2},
+ [115] = {1, AST_FORMAT_SIREN14},
+ [121] = {0, AST_RTP_CISCO_DTMF}, /* Must be type 121 */
+};
+
+int ast_rtp_engine_register2(struct ast_rtp_engine *engine, struct ast_module *module)
+{
+ struct ast_rtp_engine *current_engine;
+
+ /* Perform a sanity check on the engine structure to make sure it has the basics */
+ if (ast_strlen_zero(engine->name) || !engine->new || !engine->destroy || !engine->write || !engine->read) {
+ ast_log(LOG_WARNING, "RTP Engine '%s' failed sanity check so it was not registered.\n", !ast_strlen_zero(engine->name) ? engine->name : "Unknown");
+ return -1;
+ }
+
+ /* Link owner module to the RTP engine for reference counting purposes */
+ engine->mod = module;
+
+ AST_RWLIST_WRLOCK(&engines);
+
+ /* Ensure that no two modules with the same name are registered at the same time */
+ AST_RWLIST_TRAVERSE(&engines, current_engine, entry) {
+ if (!strcmp(current_engine->name, engine->name)) {
+ ast_log(LOG_WARNING, "An RTP engine with the name '%s' has already been registered.\n", engine->name);
+ AST_RWLIST_UNLOCK(&engines);
+ return -1;
+ }
+ }
+
+ /* The engine survived our critique. Off to the list it goes to be used */
+ AST_RWLIST_INSERT_TAIL(&engines, engine, entry);
+
+ AST_RWLIST_UNLOCK(&engines);
+
+ ast_verb(2, "Registered RTP engine '%s'\n", engine->name);
+
+ return 0;
+}
+
+int ast_rtp_engine_unregister(struct ast_rtp_engine *engine)
+{
+ struct ast_rtp_engine *current_engine = NULL;
+
+ AST_RWLIST_WRLOCK(&engines);
+
+ if ((current_engine = AST_RWLIST_REMOVE(&engines, engine, entry))) {
+ ast_verb(2, "Unregistered RTP engine '%s'\n", engine->name);
+ }
+
+ AST_RWLIST_UNLOCK(&engines);
+
+ return current_engine ? 0 : -1;
+}
+
+int ast_rtp_glue_register2(struct ast_rtp_glue *glue, struct ast_module *module)
+{
+ struct ast_rtp_glue *current_glue = NULL;
+
+ if (ast_strlen_zero(glue->type)) {
+ return -1;
+ }
+
+ glue->mod = module;
+
+ AST_RWLIST_WRLOCK(&glues);
+
+ AST_RWLIST_TRAVERSE(&glues, current_glue, entry) {
+ if (!strcasecmp(current_glue->type, glue->type)) {
+ ast_log(LOG_WARNING, "RTP glue with the name '%s' has already been registered.\n", glue->type);
+ AST_RWLIST_UNLOCK(&glues);
+ return -1;
+ }
+ }
+
+ AST_RWLIST_INSERT_TAIL(&glues, glue, entry);
+
+ AST_RWLIST_UNLOCK(&glues);
+
+ ast_verb(2, "Registered RTP glue '%s'\n", glue->type);
+
+ return 0;
+}
+
+int ast_rtp_glue_unregister(struct ast_rtp_glue *glue)
+{
+ struct ast_rtp_glue *current_glue = NULL;
+
+ AST_RWLIST_WRLOCK(&glues);
+
+ if ((current_glue = AST_RWLIST_REMOVE(&glues, glue, entry))) {
+ ast_verb(2, "Unregistered RTP glue '%s'\n", glue->type);
+ }
+
+ AST_RWLIST_UNLOCK(&glues);
+
+ return current_glue ? 0 : -1;
+}
+
+static void instance_destructor(void *obj)
+{
+ struct ast_rtp_instance *instance = obj;
+
+ /* Pass us off to the engine to destroy */
+ if (instance->data && instance->engine->destroy(instance)) {
+ ast_debug(1, "Engine '%s' failed to destroy RTP instance '%p'\n", instance->engine->name, instance);
+ return;
+ }
+
+ /* Drop our engine reference */
+ ast_module_unref(instance->engine->mod);
+
+ ast_debug(1, "Destroyed RTP instance '%p'\n", instance);
+}
+
+int ast_rtp_instance_destroy(struct ast_rtp_instance *instance)
+{
+ ao2_ref(instance, -1);
+
+ return 0;
+}
+
+struct ast_rtp_instance *ast_rtp_instance_new(const char *engine_name, struct sched_context *sched, struct sockaddr_in *sin, void *data)
+{
+ struct ast_rtp_instance *instance = NULL;
+ struct ast_rtp_engine *engine = NULL;
+
+ AST_RWLIST_RDLOCK(&engines);
+
+ /* If an engine name was specified try to use it or otherwise use the first one registered */
+ if (!ast_strlen_zero(engine_name)) {
+ AST_RWLIST_TRAVERSE(&engines, engine, entry) {
+ if (!strcmp(engine->name, engine_name)) {
+ break;
+ }
+ }
+ } else {
+ engine = AST_RWLIST_FIRST(&engines);
+ }
+
+ /* If no engine was actually found bail out now */
+ if (!engine) {
+ ast_log(LOG_ERROR, "No RTP engine was found. Do you have one loaded?\n");
+ AST_RWLIST_UNLOCK(&engines);
+ return NULL;
+ }
+
+ /* Bump up the reference count before we return so the module can not be unloaded */
+ ast_module_ref(engine->mod);
+
+ AST_RWLIST_UNLOCK(&engines);
+
+ /* Allocate a new RTP instance */
+ if (!(instance = ao2_alloc(sizeof(*instance), instance_destructor))) {
+ ast_module_unref(engine->mod);
+ return NULL;
+ }
+ instance->engine = engine;
+ memcpy(&instance->local_address, sin, sizeof(instance->local_address));
+
+ ast_debug(1, "Using engine '%s' for RTP instance '%p'\n", engine->name, instance);
+
+ /* And pass it off to the engine to setup */
+ if (instance->engine->new(instance, sched, sin, data)) {
+ ast_debug(1, "Engine '%s' failed to setup RTP instance '%p'\n", engine->name, instance);
+ ao2_ref(instance, -1);
+ return NULL;
+ }
+
+ ast_debug(1, "RTP instance '%p' is setup and ready to go\n", instance);
+
+ return instance;
+}
+
+void ast_rtp_instance_set_data(struct ast_rtp_instance *instance, void *data)
+{
+ instance->data = data;
+}
+
+void *ast_rtp_instance_get_data(struct ast_rtp_instance *instance)
+{
+ return instance->data;
+}
+
+int ast_rtp_instance_write(struct ast_rtp_instance *instance, struct ast_frame *frame)
+{
+ return instance->engine->write(instance, frame);
+}
+
+struct ast_frame *ast_rtp_instance_read(struct ast_rtp_instance *instance, int rtcp)
+{
+ return instance->engine->read(instance, rtcp);
+}
+
+int ast_rtp_instance_set_local_address(struct ast_rtp_instance *instance, struct sockaddr_in *address)
+{
+ memcpy(&instance->local_address, address, sizeof(instance->local_address));
+ return 0;
+}
+
+int ast_rtp_instance_set_remote_address(struct ast_rtp_instance *instance, struct sockaddr_in *address)
+{
+ if (&instance->remote_address != address) {
+ memcpy(&instance->remote_address, address, sizeof(instance->remote_address));
+ }
+
+ /* moo */
+
+ if (instance->engine->remote_address_set) {
+ instance->engine->remote_address_set(instance, address);
+ }
+
+ return 0;
+}
+
+int ast_rtp_instance_get_local_address(struct ast_rtp_instance *instance, struct sockaddr_in *address)
+{
+ if ((address->sin_family != AF_INET) ||
+ (address->sin_port != instance->local_address.sin_port) ||
+ (address->sin_addr.s_addr != instance->local_address.sin_addr.s_addr)) {
+ memcpy(address, &instance->local_address, sizeof(address));
+ return 1;
+ }
+
+ return 0;
+}
+
+int ast_rtp_instance_get_remote_address(struct ast_rtp_instance *instance, struct sockaddr_in *address)
+{
+ if ((address->sin_family != AF_INET) ||
+ (address->sin_port != instance->remote_address.sin_port) ||
+ (address->sin_addr.s_addr != instance->remote_address.sin_addr.s_addr)) {
+ memcpy(address, &instance->remote_address, sizeof(address));
+ return 1;
+ }
+
+ return 0;
+}
+
+void ast_rtp_instance_set_extended_prop(struct ast_rtp_instance *instance, int property, void *value)
+{
+ if (instance->engine->extended_prop_set) {
+ instance->engine->extended_prop_set(instance, property, value);
+ }
+}
+
+void *ast_rtp_instance_get_extended_prop(struct ast_rtp_instance *instance, int property)
+{
+ if (instance->engine->extended_prop_get) {
+ return instance->engine->extended_prop_get(instance, property);
+ }
+
+ return NULL;
+}
+
+void ast_rtp_instance_set_prop(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value)
+{
+ instance->properties[property] = value;
+
+ if (instance->engine->prop_set) {
+ instance->engine->prop_set(instance, property, value);
+ }
+}
+
+int ast_rtp_instance_get_prop(struct ast_rtp_instance *instance, enum ast_rtp_property property)
+{
+ return instance->properties[property];
+}
+
+struct ast_rtp_codecs *ast_rtp_instance_get_codecs(struct ast_rtp_instance *instance)
+{
+ return &instance->codecs;
+}
+
+void ast_rtp_codecs_payloads_clear(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance)
+{
+ int i;
+
+ for (i = 0; i < AST_RTP_MAX_PT; i++) {
+ ast_debug(2, "Clearing payload %d on %p\n", i, codecs);
+ codecs->payloads[i].asterisk_format = 0;
+ codecs->payloads[i].code = 0;
+ if (instance && instance->engine && instance->engine->payload_set) {
+ instance->engine->payload_set(instance, i, 0, 0);
+ }
+ }
+}
+
+void ast_rtp_codecs_payloads_default(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance)
+{
+ int i;
+
+ for (i = 0; i < AST_RTP_MAX_PT; i++) {
+ if (static_RTP_PT[i].code) {
+ ast_debug(2, "Set default payload %d on %p\n", i, codecs);
+ codecs->payloads[i].asterisk_format = static_RTP_PT[i].asterisk_format;
+ codecs->payloads[i].code = static_RTP_PT[i].code;
+ if (instance && instance->engine && instance->engine->payload_set) {
+ instance->engine->payload_set(instance, i, codecs->payloads[i].asterisk_format, codecs->payloads[i].code);
+ }
+ }
+ }
+}
+
+void ast_rtp_codecs_payloads_copy(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance)
+{
+ int i;
+
+ for (i = 0; i < AST_RTP_MAX_PT; i++) {
+ if (src->payloads[i].code) {
+ ast_debug(2, "Copying payload %d from %p to %p\n", i, src, dest);
+ dest->payloads[i].asterisk_format = src->payloads[i].asterisk_format;
+ dest->payloads[i].code = src->payloads[i].code;
+ if (instance && instance->engine && instance->engine->payload_set) {
+ instance->engine->payload_set(instance, i, dest->payloads[i].asterisk_format, dest->payloads[i].code);
+ }
+ }
+ }
+}
+
+void ast_rtp_codecs_payloads_set_m_type(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload)
+{
+ if (payload < 0 || payload > AST_RTP_MAX_PT || !static_RTP_PT[payload].code) {
+ return;
+ }
+
+ codecs->payloads[payload].asterisk_format = static_RTP_PT[payload].asterisk_format;
+ codecs->payloads[payload].code = static_RTP_PT[payload].code;
+
+ ast_debug(1, "Setting payload %d based on m type on %p\n", payload, codecs);
+
+ if (instance && instance->engine && instance->engine->payload_set) {
+ instance->engine->payload_set(instance, payload, codecs->payloads[payload].asterisk_format, codecs->payloads[payload].code);
+ }
+}
+
+int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int pt,
+ char *mimetype, char *mimesubtype,
+ enum ast_rtp_options options,
+ unsigned int sample_rate)
+{
+ unsigned int i;
+ int found = 0;
+
+ if (pt < 0 || pt > AST_RTP_MAX_PT)
+ return -1; /* bogus payload type */
+
+ for (i = 0; i < ARRAY_LEN(ast_rtp_mime_types); ++i) {
+ const struct ast_rtp_mime_type *t = &ast_rtp_mime_types[i];
+
+ if (strcasecmp(mimesubtype, t->subtype)) {
+ continue;
+ }
+
+ if (strcasecmp(mimetype, t->type)) {
+ continue;
+ }
+
+ /* if both sample rates have been supplied, and they don't match,
+ then this not a match; if one has not been supplied, then the
+ rates are not compared */
+ if (sample_rate && t->sample_rate &&
+ (sample_rate != t->sample_rate)) {
+ continue;
+ }
+
+ found = 1;
+ codecs->payloads[pt] = t->payload_type;
+
+ if ((t->payload_type.code == AST_FORMAT_G726) &&
+ t->payload_type.asterisk_format &&
+ (options & AST_RTP_OPT_G726_NONSTANDARD)) {
+ codecs->payloads[pt].code = AST_FORMAT_G726_AAL2;
+ }
+
+ if (instance && instance->engine && instance->engine->payload_set) {
+ instance->engine->payload_set(instance, pt, codecs->payloads[i].asterisk_format, codecs->payloads[i].code);
+ }
+
+ break;
+ }
+
+ return (found ? 0 : -2);
+}
+
+int ast_rtp_codecs_payloads_set_rtpmap_type(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload, char *mimetype, char *mimesubtype, enum ast_rtp_options options)
+{
+ return ast_rtp_codecs_payloads_set_rtpmap_type_rate(codecs, instance, payload, mimetype, mimesubtype, options, 0);
+}
+
+void ast_rtp_codecs_payloads_unset(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload)
+{
+ if (payload < 0 || payload > AST_RTP_MAX_PT) {
+ return;
+ }
+
+ ast_debug(2, "Unsetting payload %d on %p\n", payload, codecs);
+
+ codecs->payloads[payload].asterisk_format = 0;
+ codecs->payloads[payload].code = 0;
+
+ if (instance && instance->engine && instance->engine->payload_set) {
+ instance->engine->payload_set(instance, payload, 0, 0);
+ }
+}
+
+struct ast_rtp_payload_type ast_rtp_codecs_payload_lookup(struct ast_rtp_codecs *codecs, int payload)
+{
+ struct ast_rtp_payload_type result = { .asterisk_format = 0, };
+
+ if (payload < 0 || payload > AST_RTP_MAX_PT) {
+ return result;
+ }
+
+ result.asterisk_format = codecs->payloads[payload].asterisk_format;
+ result.code = codecs->payloads[payload].code;
+
+ if (!result.code) {
+ result = static_RTP_PT[payload];
+ }
+
+ return result;
+}
+
+void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, int *astformats, int *nonastformats)
+{
+ int i;
+
+ *astformats = *nonastformats = 0;
+
+ for (i = 0; i < AST_RTP_MAX_PT; i++) {
+ if (codecs->payloads[i].code) {
+ ast_debug(1, "Incorporating payload %d on %p\n", i, codecs);
+ }
+ if (codecs->payloads[i].asterisk_format) {
+ *astformats |= codecs->payloads[i].code;
+ } else {
+ *nonastformats |= codecs->payloads[i].code;
+ }
+ }
+}
+
+int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, const int asterisk_format, const int code)
+{
+ int i;
+
+ for (i = 0; i < AST_RTP_MAX_PT; i++) {
+ if (codecs->payloads[i].asterisk_format == asterisk_format && codecs->payloads[i].code == code) {
+ ast_debug(2, "Found code %d at payload %d on %p\n", code, i, codecs);
+ return i;
+ }
+ }
+
+ for (i = 0; i < AST_RTP_MAX_PT; i++) {
+ if (static_RTP_PT[i].asterisk_format == asterisk_format && static_RTP_PT[i].code == code) {
+ return i;
+ }
+ }
+
+ return -1;
+}
+
+const char *ast_rtp_lookup_mime_subtype2(const int asterisk_format, const int code, enum ast_rtp_options options)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_LEN(ast_rtp_mime_types); i++) {
+ if (ast_rtp_mime_types[i].payload_type.code == code && ast_rtp_mime_types[i].payload_type.asterisk_format == asterisk_format) {
+ if (asterisk_format && (code == AST_FORMAT_G726_AAL2) && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
+ return "G726-32";
+ } else {
+ return ast_rtp_mime_types[i].subtype;
+ }
+ }
+ }
+
+ return "";
+}
+
+unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, int code)
+{
+ unsigned int i;
+
+ for (i = 0; i < ARRAY_LEN(ast_rtp_mime_types); ++i) {
+ if ((ast_rtp_mime_types[i].payload_type.code == code) && (ast_rtp_mime_types[i].payload_type.asterisk_format == asterisk_format)) {
+ return ast_rtp_mime_types[i].sample_rate;
+ }
+ }
+
+ return 0;
+}
+
+char *ast_rtp_lookup_mime_multiple2(struct ast_str *buf, const int capability, const int asterisk_format, enum ast_rtp_options options)
+{
+ int format, found = 0;
+
+ if (!buf) {
+ return NULL;
+ }
+
+ ast_str_append(&buf, 0, "0x%x (", capability);
+
+ for (format = 1; format < AST_RTP_MAX; format <<= 1) {
+ if (capability & format) {
+ const char *name = ast_rtp_lookup_mime_subtype2(asterisk_format, format, options);
+ ast_str_append(&buf, 0, "%s|", name);
+ found = 1;
+ }
+ }
+
+ ast_str_append(&buf, 0, "%s", found ? ")" : "nothing)");
+
+ return ast_str_buffer(buf);
+}
+
+void ast_rtp_codecs_packetization_set(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, struct ast_codec_pref *prefs)
+{
+ codecs->pref = *prefs;
+
+ if (instance && instance->engine->packetization_set) {
+ instance->engine->packetization_set(instance, &instance->codecs.pref);
+ }
+}
+
+int ast_rtp_instance_dtmf_begin(struct ast_rtp_instance *instance, char digit)
+{
+ return instance->engine->dtmf_begin ? instance->engine->dtmf_begin(instance, digit) : -1;
+}
+
+int ast_rtp_instance_dtmf_end(struct ast_rtp_instance *instance, char digit)
+{
+ return instance->engine->dtmf_end ? instance->engine->dtmf_end(instance, digit) : -1;
+}
+
+int ast_rtp_instance_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode)
+{
+ if (!instance->engine->dtmf_mode_set || instance->engine->dtmf_mode_set(instance, dtmf_mode)) {
+ return -1;
+ }
+
+ instance->dtmf_mode = dtmf_mode;
+
+ return 0;
+}
+
+enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *instance)
+{
+ return instance->dtmf_mode;
+}
+
+void ast_rtp_instance_new_source(struct ast_rtp_instance *instance)
+{
+ if (instance->engine->new_source) {
+ instance->engine->new_source(instance);
+ }
+}
+
+int ast_rtp_instance_set_qos(struct ast_rtp_instance *instance, int tos, int cos, const char *desc)
+{
+ return instance->engine->qos ? instance->engine->qos(instance, tos, cos, desc) : -1;
+}
+
+void ast_rtp_instance_stop(struct ast_rtp_instance *instance)
+{
+ if (instance->engine->stop) {
+ instance->engine->stop(instance);
+ }
+}
+
+int ast_rtp_instance_fd(struct ast_rtp_instance *instance, int rtcp)
+{
+ return instance->engine->fd ? instance->engine->fd(instance, rtcp) : -1;
+}
+
+struct ast_rtp_glue *ast_rtp_instance_get_glue(const char *type)
+{
+ struct ast_rtp_glue *glue = NULL;
+
+ AST_RWLIST_RDLOCK(&glues);
+
+ AST_RWLIST_TRAVERSE(&glues, glue, entry) {
+ if (!strcasecmp(glue->type, type)) {
+ break;
+ }
+ }
+
+ AST_RWLIST_UNLOCK(&glues);
+
+ return glue;
+}
+
+static enum ast_bridge_result local_bridge_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1, int timeoutms, int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1)
+{
+ enum ast_bridge_result res = AST_BRIDGE_FAILED;
+ struct ast_channel *who = NULL, *other = NULL, *cs[3] = { NULL, };
+ struct ast_frame *fr = NULL;
+
+ /* Start locally bridging both instances */
+ if (instance0->engine->local_bridge && instance0->engine->local_bridge(instance0, instance1)) {
+ ast_debug(1, "Failed to locally bridge %s to %s, backing out.\n", c0->name, c1->name);
+ ast_channel_unlock(c0);
+ ast_channel_unlock(c1);
+ return AST_BRIDGE_FAILED_NOWARN;
+ }
+ if (instance1->engine->local_bridge && instance1->engine->local_bridge(instance1, instance0)) {
+ ast_debug(1, "Failed to locally bridge %s to %s, backing out.\n", c1->name, c0->name);
+ if (instance0->engine->local_bridge) {
+ instance0->engine->local_bridge(instance0, NULL);
+ }
+ ast_channel_unlock(c0);
+ ast_channel_unlock(c1);
+ return AST_BRIDGE_FAILED_NOWARN;
+ }
+
+ ast_channel_unlock(c0);
+ ast_channel_unlock(c1);
+
+ instance0->bridged = instance1;
+ instance1->bridged = instance0;
+
+ ast_poll_channel_add(c0, c1);
+
+ /* Hop into a loop waiting for a frame from either channel */
+ cs[0] = c0;
+ cs[1] = c1;
+ cs[2] = NULL;
+ for (;;) {
+ /* If the underlying formats have changed force this bridge to break */
+ if ((c0->rawreadformat != c1->rawwriteformat) || (c1->rawreadformat != c0->rawwriteformat)) {
+ ast_debug(1, "rtp-engine-local-bridge: Oooh, formats changed, backing out\n");
+ res = AST_BRIDGE_FAILED_NOWARN;
+ break;
+ }
+ /* Check if anything changed */
+ if ((c0->tech_pvt != pvt0) ||
+ (c1->tech_pvt != pvt1) ||
+ (c0->masq || c0->masqr || c1->masq || c1->masqr) ||
+ (c0->monitor || c0->audiohooks || c1->monitor || c1->audiohooks)) {
+ ast_debug(1, "rtp-engine-local-bridge: Oooh, something is weird, backing out\n");
+ /* If a masquerade needs to happen we have to try to read in a frame so that it actually happens. Without this we risk being called again and going into a loop */
+ if ((c0->masq || c0->masqr) && (fr = ast_read(c0))) {
+ ast_frfree(fr);
+ }
+ if ((c1->masq || c1->masqr) && (fr = ast_read(c1))) {
+ ast_frfree(fr);
+ }
+ res = AST_BRIDGE_RETRY;
+ break;
+ }
+ /* Wait on a channel to feed us a frame */
+ if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) {
+ if (!timeoutms) {
+ res = AST_BRIDGE_RETRY;
+ break;
+ }
+ ast_debug(2, "rtp-engine-local-bridge: Ooh, empty read...\n");
+ if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
+ break;
+ }
+ continue;
+ }
+ /* Read in frame from channel */
+ fr = ast_read(who);
+ other = (who == c0) ? c1 : c0;
+ /* Depending on the frame we may need to break out of our bridge */
+ if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) &&
+ ((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) |
+ ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)))) {
+ /* Record received frame and who */
+ *fo = fr;
+ *rc = who;
+ ast_debug(1, "rtp-engine-local-bridge: Ooh, got a %s\n", fr ? "digit" : "hangup");
+ res = AST_BRIDGE_COMPLETE;
+ break;
+ } else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
+ if ((fr->subclass == AST_CONTROL_HOLD) ||
+ (fr->subclass == AST_CONTROL_UNHOLD) ||
+ (fr->subclass == AST_CONTROL_VIDUPDATE) ||
+ (fr->subclass == AST_CONTROL_T38) ||
+ (fr->subclass == AST_CONTROL_SRCUPDATE)) {
+ /* If we are going on hold, then break callback mode and P2P bridging */
+ if (fr->subclass == AST_CONTROL_HOLD) {
+ if (instance0->engine->local_bridge) {
+ instance0->engine->local_bridge(instance0, NULL);
+ }
+ if (instance1->engine->local_bridge) {
+ instance1->engine->local_bridge(instance1, NULL);
+ }
+ instance0->bridged = NULL;
+ instance1->bridged = NULL;
+ } else if (fr->subclass == AST_CONTROL_UNHOLD) {
+ if (instance0->engine->local_bridge) {
+ instance0->engine->local_bridge(instance0, instance1);
+ }
+ if (instance1->engine->local_bridge) {
+ instance1->engine->local_bridge(instance1, instance0);
+ }
+ instance0->bridged = instance1;
+ instance1->bridged = instance0;
+ }
+ ast_indicate_data(other, fr->subclass, fr->data.ptr, fr->datalen);
+ ast_frfree(fr);
+ } else {
+ *fo = fr;
+ *rc = who;
+ ast_debug(1, "rtp-engine-local-bridge: Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass, who->name);
+ res = AST_BRIDGE_COMPLETE;
+ break;
+ }
+ } else {
+ if ((fr->frametype == AST_FRAME_DTMF_BEGIN) ||
+ (fr->frametype == AST_FRAME_DTMF_END) ||
+ (fr->frametype == AST_FRAME_VOICE) ||
+ (fr->frametype == AST_FRAME_VIDEO) ||
+ (fr->frametype == AST_FRAME_IMAGE) ||
+ (fr->frametype == AST_FRAME_HTML) ||
+ (fr->frametype == AST_FRAME_MODEM) ||
+ (fr->frametype == AST_FRAME_TEXT)) {
+ ast_write(other, fr);
+ }
+
+ ast_frfree(fr);
+ }
+ /* Swap priority */
+ cs[2] = cs[0];
+ cs[0] = cs[1];
+ cs[1] = cs[2];
+ }
+
+ /* Stop locally bridging both instances */
+ if (instance0->engine->local_bridge) {
+ instance0->engine->local_bridge(instance0, NULL);
+ }
+ if (instance1->engine->local_bridge) {
+ instance1->engine->local_bridge(instance1, NULL);
+ }
+
+ instance0->bridged = NULL;
+ instance1->bridged = NULL;
+
+ ast_poll_channel_del(c0, c1);
+
+ return res;
+}
+
+static enum ast_bridge_result remote_bridge_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1,
+ struct ast_rtp_instance *vinstance0, struct ast_rtp_instance *vinstance1, struct ast_rtp_instance *tinstance0,
+ struct ast_rtp_instance *tinstance1, struct ast_rtp_glue *glue0, struct ast_rtp_glue *glue1, int codec0, int codec1, int timeoutms,
+ int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1)
+{
+ enum ast_bridge_result res = AST_BRIDGE_FAILED;
+ struct ast_channel *who = NULL, *other = NULL, *cs[3] = { NULL, };
+ int oldcodec0 = codec0, oldcodec1 = codec1;
+ struct sockaddr_in ac1 = {0,}, vac1 = {0,}, tac1 = {0,}, ac0 = {0,}, vac0 = {0,}, tac0 = {0,};
+ struct sockaddr_in t1 = {0,}, vt1 = {0,}, tt1 = {0,}, t0 = {0,}, vt0 = {0,}, tt0 = {0,};
+ struct ast_frame *fr = NULL;
+
+ /* Test the first channel */
+ if (!(glue0->update_peer(c0, instance1, vinstance1, tinstance1, codec1, 0))) {
+ ast_rtp_instance_get_remote_address(instance1, &ac1);
+ if (vinstance1) {
+ ast_rtp_instance_get_remote_address(vinstance1, &vac1);
+ }
+ if (tinstance1) {
+ ast_rtp_instance_get_remote_address(tinstance1, &tac1);
+ }
+ } else {
+ ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name);
+ }
+
+ /* Test the second channel */
+ if (!(glue1->update_peer(c1, instance0, vinstance0, tinstance0, codec0, 0))) {
+ ast_rtp_instance_get_remote_address(instance0, &ac0);
+ if (vinstance0) {
+ ast_rtp_instance_get_remote_address(instance0, &vac0);
+ }
+ if (tinstance0) {
+ ast_rtp_instance_get_remote_address(instance0, &tac0);
+ }
+ } else {
+ ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c1->name, c0->name);
+ }
+
+ ast_channel_unlock(c0);
+ ast_channel_unlock(c1);
+
+ instance0->bridged = instance1;
+ instance1->bridged = instance0;
+
+ ast_poll_channel_add(c0, c1);
+
+ /* Go into a loop handling any stray frames that may come in */
+ cs[0] = c0;
+ cs[1] = c1;
+ cs[2] = NULL;
+ for (;;) {
+ /* Check if anything changed */
+ if ((c0->tech_pvt != pvt0) ||
+ (c1->tech_pvt != pvt1) ||
+ (c0->masq || c0->masqr || c1->masq || c1->masqr) ||
+ (c0->monitor || c0->audiohooks || c1->monitor || c1->audiohooks)) {
+ ast_debug(1, "Oooh, something is weird, backing out\n");
+ res = AST_BRIDGE_RETRY;
+ break;
+ }
+
+ /* Check if they have changed their address */
+ ast_rtp_instance_get_remote_address(instance1, &t1);
+ if (vinstance1) {
+ ast_rtp_instance_get_remote_address(vinstance1, &vt1);
+ }
+ if (tinstance1) {
+ ast_rtp_instance_get_remote_address(tinstance1, &tt1);
+ }
+ if (glue1->get_codec) {
+ codec1 = glue1->get_codec(c1);
+ }
+
+ ast_rtp_instance_get_remote_address(instance0, &t0);
+ if (vinstance0) {
+ ast_rtp_instance_get_remote_address(vinstance0, &vt0);
+ }
+ if (tinstance0) {
+ ast_rtp_instance_get_remote_address(tinstance0, &tt0);
+ }
+ if (glue0->get_codec) {
+ codec0 = glue0->get_codec(c0);
+ }
+
+ if ((inaddrcmp(&t1, &ac1)) ||
+ (vinstance1 && inaddrcmp(&vt1, &vac1)) ||
+ (tinstance1 && inaddrcmp(&tt1, &tac1)) ||
+ (codec1 != oldcodec1)) {
+ ast_debug(1, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
+ c1->name, ast_inet_ntoa(t1.sin_addr), ntohs(t1.sin_port), codec1);
+ ast_debug(1, "Oooh, '%s' changed end vaddress to %s:%d (format %d)\n",
+ c1->name, ast_inet_ntoa(vt1.sin_addr), ntohs(vt1.sin_port), codec1);
+ ast_debug(1, "Oooh, '%s' changed end taddress to %s:%d (format %d)\n",
+ c1->name, ast_inet_ntoa(tt1.sin_addr), ntohs(tt1.sin_port), codec1);
+ ast_debug(1, "Oooh, '%s' was %s:%d/(format %d)\n",
+ c1->name, ast_inet_ntoa(ac1.sin_addr), ntohs(ac1.sin_port), oldcodec1);
+ ast_debug(1, "Oooh, '%s' was %s:%d/(format %d)\n",
+ c1->name, ast_inet_ntoa(vac1.sin_addr), ntohs(vac1.sin_port), oldcodec1);
+ ast_debug(1, "Oooh, '%s' was %s:%d/(format %d)\n",
+ c1->name, ast_inet_ntoa(tac1.sin_addr), ntohs(tac1.sin_port), oldcodec1);
+ if (glue0->update_peer(c0, t1.sin_addr.s_addr ? instance1 : NULL, vt1.sin_addr.s_addr ? vinstance1 : NULL, tt1.sin_addr.s_addr ? tinstance1 : NULL, codec1, 0)) {
+ ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c0->name, c1->name);
+ }
+ memcpy(&ac1, &t1, sizeof(ac1));
+ memcpy(&vac1, &vt1, sizeof(vac1));
+ memcpy(&tac1, &tt1, sizeof(tac1));
+ oldcodec1 = codec1;
+ }
+ if ((inaddrcmp(&t0, &ac0)) ||
+ (vinstance0 && inaddrcmp(&vt0, &vac0)) ||
+ (tinstance0 && inaddrcmp(&tt0, &tac0))) {
+ ast_debug(1, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
+ c0->name, ast_inet_ntoa(t0.sin_addr), ntohs(t0.sin_port), codec0);
+ ast_debug(1, "Oooh, '%s' was %s:%d/(format %d)\n",
+ c0->name, ast_inet_ntoa(ac0.sin_addr), ntohs(ac0.sin_port), oldcodec0);
+ if (glue1->update_peer(c1, t0.sin_addr.s_addr ? instance0 : NULL, vt0.sin_addr.s_addr ? vinstance0 : NULL, tt0.sin_addr.s_addr ? tinstance0 : NULL, codec0, 0)) {
+ ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c1->name, c0->name);
+ }
+ memcpy(&ac0, &t0, sizeof(ac0));
+ memcpy(&vac0, &vt0, sizeof(vac0));
+ memcpy(&tac0, &tt0, sizeof(tac0));
+ oldcodec0 = codec0;
+ }
+
+ /* Wait for frame to come in on the channels */
+ if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) {
+ if (!timeoutms) {
+ res = AST_BRIDGE_RETRY;
+ break;
+ }
+ ast_debug(1, "Ooh, empty read...\n");
+ if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
+ break;
+ }
+ continue;
+ }
+ fr = ast_read(who);
+ other = (who == c0) ? c1 : c0;
+ if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) &&
+ (((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) ||
+ ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) {
+ /* Break out of bridge */
+ *fo = fr;
+ *rc = who;
+ ast_debug(1, "Oooh, got a %s\n", fr ? "digit" : "hangup");
+ res = AST_BRIDGE_COMPLETE;
+ break;
+ } else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
+ if ((fr->subclass == AST_CONTROL_HOLD) ||
+ (fr->subclass == AST_CONTROL_UNHOLD) ||
+ (fr->subclass == AST_CONTROL_VIDUPDATE) ||
+ (fr->subclass == AST_CONTROL_T38) ||
+ (fr->subclass == AST_CONTROL_SRCUPDATE)) {
+ if (fr->subclass == AST_CONTROL_HOLD) {
+ /* If we someone went on hold we want the other side to reinvite back to us */
+ if (who == c0) {
+ glue1->update_peer(c1, NULL, NULL, NULL, 0, 0);
+ } else {
+ glue0->update_peer(c0, NULL, NULL, NULL, 0, 0);
+ }
+ } else if (fr->subclass == AST_CONTROL_UNHOLD) {
+ /* If they went off hold they should go back to being direct */
+ if (who == c0) {
+ glue1->update_peer(c1, instance0, vinstance0, tinstance0, codec0, 0);
+ } else {
+ glue0->update_peer(c0, instance1, vinstance1, tinstance1, codec1, 0);
+ }
+ }
+ /* Update local address information */
+ ast_rtp_instance_get_remote_address(instance0, &t0);
+ memcpy(&ac0, &t0, sizeof(ac0));
+ ast_rtp_instance_get_remote_address(instance1, &t1);
+ memcpy(&ac1, &t1, sizeof(ac1));
+ /* Update codec information */
+ if (glue0->get_codec && c0->tech_pvt) {
+ oldcodec0 = codec0 = glue0->get_codec(c0);
+ }
+ if (glue1->get_codec && c1->tech_pvt) {
+ oldcodec1 = codec1 = glue1->get_codec(c1);
+ }
+ ast_indicate_data(other, fr->subclass, fr->data.ptr, fr->datalen);
+ ast_frfree(fr);
+ } else {
+ *fo = fr;
+ *rc = who;
+ ast_debug(1, "Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass, who->name);
+ return AST_BRIDGE_COMPLETE;
+ }
+ } else {
+ if ((fr->frametype == AST_FRAME_DTMF_BEGIN) ||
+ (fr->frametype == AST_FRAME_DTMF_END) ||
+ (fr->frametype == AST_FRAME_VOICE) ||
+ (fr->frametype == AST_FRAME_VIDEO) ||
+ (fr->frametype == AST_FRAME_IMAGE) ||
+ (fr->frametype == AST_FRAME_HTML) ||
+ (fr->frametype == AST_FRAME_MODEM) ||
+ (fr->frametype == AST_FRAME_TEXT)) {
+ ast_write(other, fr);
+ }
+ ast_frfree(fr);
+ }
+ /* Swap priority */
+ cs[2] = cs[0];
+ cs[0] = cs[1];
+ cs[1] = cs[2];
+ }
+
+ if (glue0->update_peer(c0, NULL, NULL, NULL, 0, 0)) {
+ ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
+ }
+ if (glue1->update_peer(c1, NULL, NULL, NULL, 0, 0)) {
+ ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
+ }
+
+ instance0->bridged = NULL;
+ instance1->bridged = NULL;
+
+ ast_poll_channel_del(c0, c1);
+
+ return res;
+}
+
+/*!
+ * \brief Conditionally unref an rtp instance
+ */
+static void unref_instance_cond(struct ast_rtp_instance **instance)
+{
+ if (*instance) {
+ ao2_ref(*instance, -1);
+ *instance = NULL;
+ }
+}
+
+enum ast_bridge_result ast_rtp_instance_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms)
+{
+ struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
+ *vinstance0 = NULL, *vinstance1 = NULL,
+ *tinstance0 = NULL, *tinstance1 = NULL;
+ struct ast_rtp_glue *glue0, *glue1;
+ enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID, text_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
+ enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID, text_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
+ enum ast_bridge_result res = AST_BRIDGE_FAILED;
+ int codec0 = 0, codec1 = 0;
+ int unlock_chans = 1;
+
+ /* Lock both channels so we can look for the glue that binds them together */
+ ast_channel_lock(c0);
+ while (ast_channel_trylock(c1)) {
+ ast_channel_unlock(c0);
+ usleep(1);
+ ast_channel_lock(c0);
+ }
+
+ /* Ensure neither channel got hungup during lock avoidance */
+ if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
+ ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", c0->name, c1->name);
+ goto done;
+ }
+
+ /* Grab glue that binds each channel to something using the RTP engine */
+ if (!(glue0 = ast_rtp_instance_get_glue(c0->tech->type)) || !(glue1 = ast_rtp_instance_get_glue(c1->tech->type))) {
+ ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", glue0 ? c1->name : c0->name);
+ goto done;
+ }
+
+ audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
+ video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
+ text_glue0_res = glue0->get_trtp_info ? glue0->get_trtp_info(c0, &tinstance0) : AST_RTP_GLUE_RESULT_FORBID;
+
+ audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
+ video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
+ text_glue1_res = glue1->get_trtp_info ? glue1->get_trtp_info(c1, &tinstance1) : AST_RTP_GLUE_RESULT_FORBID;
+
+ /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
+ if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) {
+ audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
+ }
+ if (video_glue1_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue1_res != AST_RTP_GLUE_RESULT_REMOTE)) {
+ audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
+ }
+
+ /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
+ if (audio_glue0_res == AST_RTP_GLUE_RESULT_FORBID || audio_glue1_res == AST_RTP_GLUE_RESULT_FORBID) {
+ res = AST_BRIDGE_FAILED_NOWARN;
+ goto done;
+ }
+
+ /* If we have gotten to a local bridge make sure that both sides have the same local bridge callback and that they are DTMF compatible */
+ if ((audio_glue0_res == AST_RTP_GLUE_RESULT_LOCAL || audio_glue1_res == AST_RTP_GLUE_RESULT_LOCAL) && ((instance0->engine->local_bridge != instance1->engine->local_bridge) || (instance0->engine->dtmf_compatible && !instance0->engine->dtmf_compatible(c0, instance0, c1, instance1)))) {
+ res = AST_BRIDGE_FAILED_NOWARN;
+ goto done;
+ }
+
+ /* Make sure that codecs match */
+ codec0 = glue0->get_codec ? glue0->get_codec(c0) : 0;
+ codec1 = glue1->get_codec ? glue1->get_codec(c1) : 0;
+ if (codec0 && codec1 && !(codec0 & codec1)) {
+ ast_debug(1, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
+ res = AST_BRIDGE_FAILED_NOWARN;
+ goto done;
+ }
+
+ /* Depending on the end result for bridging either do a local bridge or remote bridge */
+ if (audio_glue0_res == AST_RTP_GLUE_RESULT_LOCAL || audio_glue1_res == AST_RTP_GLUE_RESULT_LOCAL) {
+ ast_verbose(VERBOSE_PREFIX_3 "Locally bridging %s and %s\n", c0->name, c1->name);
+ res = local_bridge_loop(c0, c1, instance0, instance1, timeoutms, flags, fo, rc, c0->tech_pvt, c1->tech_pvt);
+ } else {
+ ast_verbose(VERBOSE_PREFIX_3 "Remotely bridging %s and %s\n", c0->name, c1->name);
+ res = remote_bridge_loop(c0, c1, instance0, instance1, vinstance0, vinstance1,
+ tinstance0, tinstance1, glue0, glue1, codec0, codec1, timeoutms, flags,
+ fo, rc, c0->tech_pvt, c1->tech_pvt);
+ }
+
+ unlock_chans = 0;
+
+done:
+ if (unlock_chans) {
+ ast_channel_unlock(c0);
+ ast_channel_unlock(c1);
+ }
+
+ unref_instance_cond(&instance0);
+ unref_instance_cond(&instance1);
+ unref_instance_cond(&vinstance0);
+ unref_instance_cond(&vinstance1);
+ unref_instance_cond(&tinstance0);
+ unref_instance_cond(&tinstance1);
+
+ return res;
+}
+
+struct ast_rtp_instance *ast_rtp_instance_get_bridged(struct ast_rtp_instance *instance)
+{
+ return instance->bridged;
+}
+
+void ast_rtp_instance_early_bridge_make_compatible(struct ast_channel *c0, struct ast_channel *c1)
+{
+ struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
+ *vinstance0 = NULL, *vinstance1 = NULL,
+ *tinstance0 = NULL, *tinstance1 = NULL;
+ struct ast_rtp_glue *glue0, *glue1;
+ enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID, text_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
+ enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID, text_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
+ int codec0 = 0, codec1 = 0;
+ int res = 0;
+
+ /* Lock both channels so we can look for the glue that binds them together */
+ ast_channel_lock(c0);
+ while (ast_channel_trylock(c1)) {
+ ast_channel_unlock(c0);
+ usleep(1);
+ ast_channel_lock(c0);
+ }
+
+ /* Grab glue that binds each channel to something using the RTP engine */
+ if (!(glue0 = ast_rtp_instance_get_glue(c0->tech->type)) || !(glue1 = ast_rtp_instance_get_glue(c1->tech->type))) {
+ ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", glue0 ? c1->name : c0->name);
+ goto done;
+ }
+
+ audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
+ video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
+ text_glue0_res = glue0->get_trtp_info ? glue0->get_trtp_info(c0, &tinstance0) : AST_RTP_GLUE_RESULT_FORBID;
+
+ audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
+ video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
+ text_glue1_res = glue1->get_trtp_info ? glue1->get_trtp_info(c1, &tinstance1) : AST_RTP_GLUE_RESULT_FORBID;
+
+ /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
+ if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) {
+ audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
+ }
+ if (video_glue1_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue1_res != AST_RTP_GLUE_RESULT_REMOTE)) {
+ audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
+ }
+ if (audio_glue0_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue0_res == AST_RTP_GLUE_RESULT_FORBID || video_glue0_res == AST_RTP_GLUE_RESULT_REMOTE) && glue0->get_codec(c0)) {
+ codec0 = glue0->get_codec(c0);
+ }
+ if (audio_glue1_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue1_res == AST_RTP_GLUE_RESULT_FORBID || video_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) && glue1->get_codec(c1)) {
+ codec1 = glue1->get_codec(c1);
+ }
+
+ /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
+ if (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE) {
+ goto done;
+ }
+
+ /* Make sure we have matching codecs */
+ if (!(codec0 & codec1)) {
+ goto done;
+ }
+
+ ast_rtp_codecs_payloads_copy(&instance0->codecs, &instance1->codecs, instance1);
+
+ if (vinstance0 && vinstance1) {
+ ast_rtp_codecs_payloads_copy(&vinstance0->codecs, &vinstance1->codecs, vinstance1);
+ }
+ if (tinstance0 && tinstance1) {
+ ast_rtp_codecs_payloads_copy(&tinstance0->codecs, &tinstance1->codecs, tinstance1);
+ }
+
+ res = 0;
+
+done:
+ ast_channel_unlock(c0);
+ ast_channel_unlock(c1);
+
+ unref_instance_cond(&instance0);
+ unref_instance_cond(&instance1);
+ unref_instance_cond(&vinstance0);
+ unref_instance_cond(&vinstance1);
+ unref_instance_cond(&tinstance0);
+ unref_instance_cond(&tinstance1);
+
+ if (!res) {
+ ast_debug(1, "Seeded SDP of '%s' with that of '%s'\n", c0->name, c1 ? c1->name : "<unspecified>");
+ }
+}
+
+int ast_rtp_instance_early_bridge(struct ast_channel *c0, struct ast_channel *c1)
+{
+ struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
+ *vinstance0 = NULL, *vinstance1 = NULL,
+ *tinstance0 = NULL, *tinstance1 = NULL;
+ struct ast_rtp_glue *glue0, *glue1;
+ enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID, text_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
+ enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID, text_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
+ int codec0 = 0, codec1 = 0;
+ int res = 0;
+
+ /* If there is no second channel just immediately bail out, we are of no use in that scenario */
+ if (!c1) {
+ return -1;
+ }
+
+ /* Lock both channels so we can look for the glue that binds them together */
+ ast_channel_lock(c0);
+ while (ast_channel_trylock(c1)) {
+ ast_channel_unlock(c0);
+ usleep(1);
+ ast_channel_lock(c0);
+ }
+
+ /* Grab glue that binds each channel to something using the RTP engine */
+ if (!(glue0 = ast_rtp_instance_get_glue(c0->tech->type)) || !(glue1 = ast_rtp_instance_get_glue(c1->tech->type))) {
+ ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", glue0 ? c1->name : c0->name);
+ goto done;
+ }
+
+ audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
+ video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
+ text_glue0_res = glue0->get_trtp_info ? glue0->get_trtp_info(c0, &tinstance0) : AST_RTP_GLUE_RESULT_FORBID;
+
+ audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
+ video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
+ text_glue1_res = glue1->get_trtp_info ? glue1->get_trtp_info(c1, &tinstance1) : AST_RTP_GLUE_RESULT_FORBID;
+
+ /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
+ if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) {
+ audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
+ }
+ if (video_glue1_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue1_res != AST_RTP_GLUE_RESULT_REMOTE)) {
+ audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
+ }
+ if (audio_glue0_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue0_res == AST_RTP_GLUE_RESULT_FORBID || video_glue0_res == AST_RTP_GLUE_RESULT_REMOTE) && glue0->get_codec(c0)) {
+ codec0 = glue0->get_codec(c0);
+ }
+ if (audio_glue1_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue1_res == AST_RTP_GLUE_RESULT_FORBID || video_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) && glue1->get_codec(c1)) {
+ codec1 = glue1->get_codec(c1);
+ }
+
+ /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
+ if (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE) {
+ goto done;
+ }
+
+ /* Make sure we have matching codecs */
+ if (!(codec0 & codec1)) {
+ goto done;
+ }
+
+ /* Bridge media early */
+ if (glue0->update_peer(c0, instance1, vinstance1, tinstance1, codec1, 0)) {
+ ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", c0->name, c1 ? c1->name : "<unspecified>");
+ }
+
+ res = 0;
+
+done:
+ ast_channel_unlock(c0);
+ ast_channel_unlock(c1);
+
+ unref_instance_cond(&instance0);
+ unref_instance_cond(&instance1);
+ unref_instance_cond(&vinstance0);
+ unref_instance_cond(&vinstance1);
+ unref_instance_cond(&tinstance0);
+ unref_instance_cond(&tinstance1);
+
+ if (!res) {
+ ast_debug(1, "Setting early bridge SDP of '%s' with that of '%s'\n", c0->name, c1 ? c1->name : "<unspecified>");
+ }
+
+ return res;
+}
+
+int ast_rtp_red_init(struct ast_rtp_instance *instance, int buffer_time, int *payloads, int generations)
+{
+ return instance->engine->red_init ? instance->engine->red_init(instance, buffer_time, payloads, generations) : -1;
+}
+
+int ast_rtp_red_buffer(struct ast_rtp_instance *instance, struct ast_frame *frame)
+{
+ return instance->engine->red_buffer ? instance->engine->red_buffer(instance, frame) : -1;
+}
+
+int ast_rtp_instance_get_stats(struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat)
+{
+ return instance->engine->get_stat ? instance->engine->get_stat(instance, stats, stat) : -1;
+}
+
+char *ast_rtp_instance_get_quality(struct ast_rtp_instance *instance, enum ast_rtp_instance_stat_field field, char *buf, size_t size)
+{
+ struct ast_rtp_instance_stats stats;
+ enum ast_rtp_instance_stat stat;
+
+ /* Determine what statistics we will need to retrieve based on field passed in */
+ if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) {
+ stat = AST_RTP_INSTANCE_STAT_ALL;
+ } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) {
+ stat = AST_RTP_INSTANCE_STAT_COMBINED_JITTER;
+ } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) {
+ stat = AST_RTP_INSTANCE_STAT_COMBINED_LOSS;
+ } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) {
+ stat = AST_RTP_INSTANCE_STAT_COMBINED_RTT;
+ } else {
+ return NULL;
+ }
+
+ /* Attempt to actually retrieve the statistics we need to generate the quality string */
+ if (ast_rtp_instance_get_stats(instance, &stats, stat)) {
+ return NULL;
+ }
+
+ /* Now actually fill the buffer with the good information */
+ if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) {
+ snprintf(buf, size, "ssrc=%i;themssrc=%u;lp=%u;rxjitter=%u;rxcount=%u;txjitter=%u;txcount=%u;rlp=%u;rtt=%u",
+ stats.local_ssrc, stats.remote_ssrc, stats.rxploss, stats.txjitter, stats.rxcount, stats.rxjitter, stats.txcount, stats.txploss, stats.rtt);
+ } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) {
+ snprintf(buf, size, "minrxjitter=%f;maxrxjitter=%f;avgrxjitter=%f;stdevrxjitter=%f;reported_minjitter=%f;reported_maxjitter=%f;reported_avgjitter=%f;reported_stdevjitter=%f;",
+ stats.local_minjitter, stats.local_maxjitter, stats.local_normdevjitter, sqrt(stats.local_stdevjitter), stats.remote_minjitter, stats.remote_maxjitter, stats.remote_normdevjitter, sqrt(stats.remote_stdevjitter));
+ } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) {
+ snprintf(buf, size, "minrxlost=%f;maxrxlost=%f;avgrxlost=%f;stdevrxlost=%f;reported_minlost=%f;reported_maxlost=%f;reported_avglost=%f;reported_stdevlost=%f;",
+ stats.local_minrxploss, stats.local_maxrxploss, stats.local_normdevrxploss, sqrt(stats.local_stdevrxploss), stats.remote_minrxploss, stats.remote_maxrxploss, stats.remote_normdevrxploss, sqrt(stats.remote_stdevrxploss));
+ } else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) {
+ snprintf(buf, size, "minrtt=%f;maxrtt=%f;avgrtt=%f;stdevrtt=%f;", stats.minrtt, stats.maxrtt, stats.normdevrtt, stats.stdevrtt);
+ }
+
+ return buf;
+}
+
+void ast_rtp_instance_set_stats_vars(struct ast_channel *chan, struct ast_rtp_instance *instance)
+{
+ char quality_buf[AST_MAX_USER_FIELD], *quality;
+ struct ast_channel *bridge = ast_bridged_channel(chan);
+
+ if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
+ pbx_builtin_setvar_helper(chan, "RTPAUDIOQOS", quality);
+ if (bridge) {
+ pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSBRIDGED", quality);
+ }
+ }
+
+ if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf)))) {
+ pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSJITTER", quality);
+ if (bridge) {
+ pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSJITTERBRIDGED", quality);
+ }
+ }
+
+ if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf)))) {
+ pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSLOSS", quality);
+ if (bridge) {
+ pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSLOSSBRIDGED", quality);
+ }
+ }
+
+ if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf)))) {
+ pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSRTT", quality);
+ if (bridge) {
+ pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSRTTBRIDGED", quality);
+ }
+ }
+}
+
+int ast_rtp_instance_set_read_format(struct ast_rtp_instance *instance, int format)
+{
+ return instance->engine->set_read_format ? instance->engine->set_read_format(instance, format) : -1;
+}
+
+int ast_rtp_instance_set_write_format(struct ast_rtp_instance *instance, int format)
+{
+ return instance->engine->set_write_format ? instance->engine->set_write_format(instance, format) : -1;
+}
+
+int ast_rtp_instance_make_compatible(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_channel *peer)
+{
+ struct ast_rtp_glue *glue;
+ struct ast_rtp_instance *peer_instance = NULL;
+ int res = -1;
+
+ if (!instance->engine->make_compatible) {
+ return -1;
+ }
+
+ ast_channel_lock(peer);
+
+ if (!(glue = ast_rtp_instance_get_glue(peer->tech->type))) {
+ ast_channel_unlock(peer);
+ return -1;
+ }
+
+ glue->get_rtp_info(peer, &peer_instance);
+
+ if (!peer_instance || peer_instance->engine != instance->engine) {
+ ast_channel_unlock(peer);
+ peer_instance = (ao2_ref(peer_instance, -1), NULL);
+ return -1;
+ }
+
+ res = instance->engine->make_compatible(chan, instance, peer, peer_instance);
+
+ ast_channel_unlock(peer);
+
+ peer_instance = (ao2_ref(peer_instance, -1), NULL);
+
+ return res;
+}
+
+int ast_rtp_instance_activate(struct ast_rtp_instance *instance)
+{
+ return instance->engine->activate ? instance->engine->activate(instance) : 0;
+}
+
+void ast_rtp_instance_stun_request(struct ast_rtp_instance *instance, struct sockaddr_in *suggestion, const char *username)
+{
+ if (instance->engine->stun_request) {
+ instance->engine->stun_request(instance, suggestion, username);
+ }
+}
+
+void ast_rtp_instance_set_timeout(struct ast_rtp_instance *instance, int timeout)
+{
+ instance->timeout = timeout;
+}
+
+void ast_rtp_instance_set_hold_timeout(struct ast_rtp_instance *instance, int timeout)
+{
+ instance->holdtimeout = timeout;
+}
+
+int ast_rtp_instance_get_timeout(struct ast_rtp_instance *instance)
+{
+ return instance->timeout;
+}
+
+int ast_rtp_instance_get_hold_timeout(struct ast_rtp_instance *instance)
+{
+ return instance->holdtimeout;
+}
diff --git a/main/stun.c b/main/stun.c
new file mode 100644
index 0000000000..264430718b
--- /dev/null
+++ b/main/stun.c
@@ -0,0 +1,475 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2008, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*!
+ * \file
+ *
+ * \brief STUN Support
+ *
+ * \author Mark Spencer <markster@digium.com>
+ *
+ * \note STUN is defined in RFC 3489.
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision: 124370 $")
+
+#include "asterisk/_private.h"
+#include "asterisk/stun.h"
+#include "asterisk/cli.h"
+#include "asterisk/utils.h"
+#include "asterisk/channel.h"
+
+static int stundebug; /*!< Are we debugging stun? */
+
+/*!
+ * \brief STUN support code
+ *
+ * This code provides some support for doing STUN transactions.
+ * Eventually it should be moved elsewhere as other protocols
+ * than RTP can benefit from it - e.g. SIP.
+ * STUN is described in RFC3489 and it is based on the exchange
+ * of UDP packets between a client and one or more servers to
+ * determine the externally visible address (and port) of the client
+ * once it has gone through the NAT boxes that connect it to the
+ * outside.
+ * The simplest request packet is just the header defined in
+ * struct stun_header, and from the response we may just look at
+ * one attribute, STUN_MAPPED_ADDRESS, that we find in the response.
+ * By doing more transactions with different server addresses we
+ * may determine more about the behaviour of the NAT boxes, of
+ * course - the details are in the RFC.
+ *
+ * All STUN packets start with a simple header made of a type,
+ * length (excluding the header) and a 16-byte random transaction id.
+ * Following the header we may have zero or more attributes, each
+ * structured as a type, length and a value (whose format depends
+ * on the type, but often contains addresses).
+ * Of course all fields are in network format.
+ */
+
+typedef struct { unsigned int id[4]; } __attribute__((packed)) stun_trans_id;
+
+struct stun_header {
+ unsigned short msgtype;
+ unsigned short msglen;
+ stun_trans_id id;
+ unsigned char ies[0];
+} __attribute__((packed));
+
+struct stun_attr {
+ unsigned short attr;
+ unsigned short len;
+ unsigned char value[0];
+} __attribute__((packed));
+
+/*
+ * The format normally used for addresses carried by STUN messages.
+ */
+struct stun_addr {
+ unsigned char unused;
+ unsigned char family;
+ unsigned short port;
+ unsigned int addr;
+} __attribute__((packed));
+
+/*! \brief STUN message types
+ * 'BIND' refers to transactions used to determine the externally
+ * visible addresses. 'SEC' refers to transactions used to establish
+ * a session key for subsequent requests.
+ * 'SEC' functionality is not supported here.
+ */
+
+#define STUN_BINDREQ 0x0001
+#define STUN_BINDRESP 0x0101
+#define STUN_BINDERR 0x0111
+#define STUN_SECREQ 0x0002
+#define STUN_SECRESP 0x0102
+#define STUN_SECERR 0x0112
+
+/*! \brief Basic attribute types in stun messages.
+ * Messages can also contain custom attributes (codes above 0x7fff)
+ */
+#define STUN_MAPPED_ADDRESS 0x0001
+#define STUN_RESPONSE_ADDRESS 0x0002
+#define STUN_CHANGE_REQUEST 0x0003
+#define STUN_SOURCE_ADDRESS 0x0004
+#define STUN_CHANGED_ADDRESS 0x0005
+#define STUN_USERNAME 0x0006
+#define STUN_PASSWORD 0x0007
+#define STUN_MESSAGE_INTEGRITY 0x0008
+#define STUN_ERROR_CODE 0x0009
+#define STUN_UNKNOWN_ATTRIBUTES 0x000a
+#define STUN_REFLECTED_FROM 0x000b
+
+/*! \brief helper function to print message names */
+static const char *stun_msg2str(int msg)
+{
+ switch (msg) {
+ case STUN_BINDREQ:
+ return "Binding Request";
+ case STUN_BINDRESP:
+ return "Binding Response";
+ case STUN_BINDERR:
+ return "Binding Error Response";
+ case STUN_SECREQ:
+ return "Shared Secret Request";
+ case STUN_SECRESP:
+ return "Shared Secret Response";
+ case STUN_SECERR:
+ return "Shared Secret Error Response";
+ }
+ return "Non-RFC3489 Message";
+}
+
+/*! \brief helper function to print attribute names */
+static const char *stun_attr2str(int msg)
+{
+ switch (msg) {
+ case STUN_MAPPED_ADDRESS:
+ return "Mapped Address";
+ case STUN_RESPONSE_ADDRESS:
+ return "Response Address";
+ case STUN_CHANGE_REQUEST:
+ return "Change Request";
+ case STUN_SOURCE_ADDRESS:
+ return "Source Address";
+ case STUN_CHANGED_ADDRESS:
+ return "Changed Address";
+ case STUN_USERNAME:
+ return "Username";
+ case STUN_PASSWORD:
+ return "Password";
+ case STUN_MESSAGE_INTEGRITY:
+ return "Message Integrity";
+ case STUN_ERROR_CODE:
+ return "Error Code";
+ case STUN_UNKNOWN_ATTRIBUTES:
+ return "Unknown Attributes";
+ case STUN_REFLECTED_FROM:
+ return "Reflected From";
+ }
+ return "Non-RFC3489 Attribute";
+}
+
+/*! \brief here we store credentials extracted from a message */
+struct stun_state {
+ const char *username;
+ const char *password;
+};
+
+static int stun_process_attr(struct stun_state *state, struct stun_attr *attr)
+{
+ if (stundebug)
+ ast_verbose("Found STUN Attribute %s (%04x), length %d\n",
+ stun_attr2str(ntohs(attr->attr)), ntohs(attr->attr), ntohs(attr->len));
+ switch (ntohs(attr->attr)) {
+ case STUN_USERNAME:
+ state->username = (const char *) (attr->value);
+ break;
+ case STUN_PASSWORD:
+ state->password = (const char *) (attr->value);
+ break;
+ default:
+ if (stundebug)
+ ast_verbose("Ignoring STUN attribute %s (%04x), length %d\n",
+ stun_attr2str(ntohs(attr->attr)), ntohs(attr->attr), ntohs(attr->len));
+ }
+ return 0;
+}
+
+/*! \brief append a string to an STUN message */
+static void append_attr_string(struct stun_attr **attr, int attrval, const char *s, int *len, int *left)
+{
+ int size = sizeof(**attr) + strlen(s);
+ if (*left > size) {
+ (*attr)->attr = htons(attrval);
+ (*attr)->len = htons(strlen(s));
+ memcpy((*attr)->value, s, strlen(s));
+ (*attr) = (struct stun_attr *)((*attr)->value + strlen(s));
+ *len += size;
+ *left -= size;
+ }
+}
+
+/*! \brief append an address to an STUN message */
+static void append_attr_address(struct stun_attr **attr, int attrval, struct sockaddr_in *sin, int *len, int *left)
+{
+ int size = sizeof(**attr) + 8;
+ struct stun_addr *addr;
+ if (*left > size) {
+ (*attr)->attr = htons(attrval);
+ (*attr)->len = htons(8);
+ addr = (struct stun_addr *)((*attr)->value);
+ addr->unused = 0;
+ addr->family = 0x01;
+ addr->port = sin->sin_port;
+ addr->addr = sin->sin_addr.s_addr;
+ (*attr) = (struct stun_attr *)((*attr)->value + 8);
+ *len += size;
+ *left -= size;
+ }
+}
+
+/*! \brief wrapper to send an STUN message */
+static int stun_send(int s, struct sockaddr_in *dst, struct stun_header *resp)
+{
+ return sendto(s, resp, ntohs(resp->msglen) + sizeof(*resp), 0,
+ (struct sockaddr *)dst, sizeof(*dst));
+}
+
+/*! \brief helper function to generate a random request id */
+static void stun_req_id(struct stun_header *req)
+{
+ int x;
+ for (x = 0; x < 4; x++)
+ req->id.id[x] = ast_random();
+}
+
+/*! \brief handle an incoming STUN message.
+ *
+ * Do some basic sanity checks on packet size and content,
+ * try to extract a bit of information, and possibly reply.
+ * At the moment this only processes BIND requests, and returns
+ * the externally visible address of the request.
+ * If a callback is specified, invoke it with the attribute.
+ */
+int ast_stun_handle_packet(int s, struct sockaddr_in *src, unsigned char *data, size_t len, stun_cb_f *stun_cb, void *arg)
+{
+ struct stun_header *hdr = (struct stun_header *)data;
+ struct stun_attr *attr;
+ struct stun_state st;
+ int ret = AST_STUN_IGNORE;
+ int x;
+
+ /* On entry, 'len' is the length of the udp payload. After the
+ * initial checks it becomes the size of unprocessed options,
+ * while 'data' is advanced accordingly.
+ */
+ if (len < sizeof(struct stun_header)) {
+ ast_debug(1, "Runt STUN packet (only %d, wanting at least %d)\n", (int) len, (int) sizeof(struct stun_header));
+ return -1;
+ }
+ len -= sizeof(struct stun_header);
+ data += sizeof(struct stun_header);
+ x = ntohs(hdr->msglen); /* len as advertised in the message */
+ if (stundebug)
+ ast_verbose("STUN Packet, msg %s (%04x), length: %d\n", stun_msg2str(ntohs(hdr->msgtype)), ntohs(hdr->msgtype), x);
+ if (x > len) {
+ ast_debug(1, "Scrambled STUN packet length (got %d, expecting %d)\n", x, (int)len);
+ } else
+ len = x;
+ memset(&st, 0, sizeof(st));
+ while (len) {
+ if (len < sizeof(struct stun_attr)) {
+ ast_debug(1, "Runt Attribute (got %d, expecting %d)\n", (int)len, (int) sizeof(struct stun_attr));
+ break;
+ }
+ attr = (struct stun_attr *)data;
+ /* compute total attribute length */
+ x = ntohs(attr->len) + sizeof(struct stun_attr);
+ if (x > len) {
+ ast_debug(1, "Inconsistent Attribute (length %d exceeds remaining msg len %d)\n", x, (int)len);
+ break;
+ }
+ if (stun_cb)
+ stun_cb(attr, arg);
+ if (stun_process_attr(&st, attr)) {
+ ast_debug(1, "Failed to handle attribute %s (%04x)\n", stun_attr2str(ntohs(attr->attr)), ntohs(attr->attr));
+ break;
+ }
+ /* Clear attribute id: in case previous entry was a string,
+ * this will act as the terminator for the string.
+ */
+ attr->attr = 0;
+ data += x;
+ len -= x;
+ }
+ /* Null terminate any string.
+ * XXX NOTE, we write past the size of the buffer passed by the
+ * caller, so this is potentially dangerous. The only thing that
+ * saves us is that usually we read the incoming message in a
+ * much larger buffer in the struct ast_rtp
+ */
+ *data = '\0';
+
+ /* Now prepare to generate a reply, which at the moment is done
+ * only for properly formed (len == 0) STUN_BINDREQ messages.
+ */
+ if (len == 0) {
+ unsigned char respdata[1024];
+ struct stun_header *resp = (struct stun_header *)respdata;
+ int resplen = 0; /* len excluding header */
+ int respleft = sizeof(respdata) - sizeof(struct stun_header);
+
+ resp->id = hdr->id;
+ resp->msgtype = 0;
+ resp->msglen = 0;
+ attr = (struct stun_attr *)resp->ies;
+ switch (ntohs(hdr->msgtype)) {
+ case STUN_BINDREQ:
+ if (stundebug)
+ ast_verbose("STUN Bind Request, username: %s\n",
+ st.username ? st.username : "<none>");
+ if (st.username)
+ append_attr_string(&attr, STUN_USERNAME, st.username, &resplen, &respleft);
+ append_attr_address(&attr, STUN_MAPPED_ADDRESS, src, &resplen, &respleft);
+ resp->msglen = htons(resplen);
+ resp->msgtype = htons(STUN_BINDRESP);
+ stun_send(s, src, resp);
+ ret = AST_STUN_ACCEPT;
+ break;
+ default:
+ if (stundebug)
+ ast_verbose("Dunno what to do with STUN message %04x (%s)\n", ntohs(hdr->msgtype), stun_msg2str(ntohs(hdr->msgtype)));
+ }
+ }
+ return ret;
+}
+
+/*! \brief Extract the STUN_MAPPED_ADDRESS from the stun response.
+ * This is used as a callback for stun_handle_response
+ * when called from ast_stun_request.
+ */
+static int stun_get_mapped(struct stun_attr *attr, void *arg)
+{
+ struct stun_addr *addr = (struct stun_addr *)(attr + 1);
+ struct sockaddr_in *sa = (struct sockaddr_in *)arg;
+
+ if (ntohs(attr->attr) != STUN_MAPPED_ADDRESS || ntohs(attr->len) != 8)
+ return 1; /* not us. */
+ sa->sin_port = addr->port;
+ sa->sin_addr.s_addr = addr->addr;
+ return 0;
+}
+
+/*! \brief Generic STUN request
+ * Send a generic stun request to the server specified,
+ * possibly waiting for a reply and filling the 'reply' field with
+ * the externally visible address. Note that in this case the request
+ * will be blocking.
+ * (Note, the interface may change slightly in the future).
+ *
+ * \param s the socket used to send the request
+ * \param dst the address of the STUN server
+ * \param username if non null, add the username in the request
+ * \param answer if non null, the function waits for a response and
+ * puts here the externally visible address.
+ * \return 0 on success, other values on error.
+ */
+int ast_stun_request(int s, struct sockaddr_in *dst,
+ const char *username, struct sockaddr_in *answer)
+{
+ struct stun_header *req;
+ unsigned char reqdata[1024];
+ int reqlen, reqleft;
+ struct stun_attr *attr;
+ int res = 0;
+ int retry;
+
+ req = (struct stun_header *)reqdata;
+ stun_req_id(req);
+ reqlen = 0;
+ reqleft = sizeof(reqdata) - sizeof(struct stun_header);
+ req->msgtype = 0;
+ req->msglen = 0;
+ attr = (struct stun_attr *)req->ies;
+ if (username)
+ append_attr_string(&attr, STUN_USERNAME, username, &reqlen, &reqleft);
+ req->msglen = htons(reqlen);
+ req->msgtype = htons(STUN_BINDREQ);
+ for (retry = 0; retry < 3; retry++) { /* XXX make retries configurable */
+ /* send request, possibly wait for reply */
+ unsigned char reply_buf[1024];
+ fd_set rfds;
+ struct timeval to = { 3, 0 }; /* timeout, make it configurable */
+ struct sockaddr_in src;
+ socklen_t srclen;
+
+ res = stun_send(s, dst, req);
+ if (res < 0) {
+ ast_log(LOG_WARNING, "ast_stun_request send #%d failed error %d, retry\n",
+ retry, res);
+ continue;
+ }
+ if (answer == NULL)
+ break;
+ FD_ZERO(&rfds);
+ FD_SET(s, &rfds);
+ res = ast_select(s + 1, &rfds, NULL, NULL, &to);
+ if (res <= 0) /* timeout or error */
+ continue;
+ memset(&src, 0, sizeof(src));
+ srclen = sizeof(src);
+ /* XXX pass -1 in the size, because stun_handle_packet might
+ * write past the end of the buffer.
+ */
+ res = recvfrom(s, reply_buf, sizeof(reply_buf) - 1,
+ 0, (struct sockaddr *)&src, &srclen);
+ if (res < 0) {
+ ast_log(LOG_WARNING, "ast_stun_request recvfrom #%d failed error %d, retry\n",
+ retry, res);
+ continue;
+ }
+ memset(answer, 0, sizeof(struct sockaddr_in));
+ ast_stun_handle_packet(s, &src, reply_buf, res,
+ stun_get_mapped, answer);
+ res = 0; /* signal regular exit */
+ break;
+ }
+ return res;
+}
+
+static char *handle_cli_stun_set_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
+{
+ switch (cmd) {
+ case CLI_INIT:
+ e->command = "stun set debug {on|off}";
+ e->usage =
+ "Usage: stun set debug {on|off}\n"
+ " Enable/Disable STUN (Simple Traversal of UDP through NATs)\n"
+ " debugging\n";
+ return NULL;
+ case CLI_GENERATE:
+ return NULL;
+ }
+
+ if (a->argc != e->args)
+ return CLI_SHOWUSAGE;
+
+ if (!strncasecmp(a->argv[e->args-1], "on", 2))
+ stundebug = 1;
+ else if (!strncasecmp(a->argv[e->args-1], "off", 3))
+ stundebug = 0;
+ else
+ return CLI_SHOWUSAGE;
+
+ ast_cli(a->fd, "STUN Debugging %s\n", stundebug ? "Enabled" : "Disabled");
+ return CLI_SUCCESS;
+}
+
+static struct ast_cli_entry cli_stun[] = {
+ AST_CLI_DEFINE(handle_cli_stun_set_debug, "Enable/Disable STUN debugging"),
+};
+
+/*! \brief Initialize the STUN system in Asterisk */
+void ast_stun_init(void)
+{
+ ast_cli_register_multiple(cli_stun, sizeof(cli_stun) / sizeof(struct ast_cli_entry));
+}
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
new file mode 100644
index 0000000000..e16088d6e8
--- /dev/null
+++ b/res/res_rtp_asterisk.c
@@ -0,0 +1,2579 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2008, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*!
+ * \file
+ *
+ * \brief Supports RTP and RTCP with Symmetric RTP support for NAT traversal.
+ *
+ * \author Mark Spencer <markster@digium.com>
+ *
+ * \note RTP is defined in RFC 3550.
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision: 138083 $")
+
+#include <sys/time.h>
+#include <signal.h>
+#include <fcntl.h>
+#include <math.h>
+
+#include "asterisk/stun.h"
+#include "asterisk/pbx.h"
+#include "asterisk/frame.h"
+#include "asterisk/channel.h"
+#include "asterisk/acl.h"
+#include "asterisk/config.h"
+#include "asterisk/lock.h"
+#include "asterisk/utils.h"
+#include "asterisk/netsock.h"
+#include "asterisk/cli.h"
+#include "asterisk/manager.h"
+#include "asterisk/unaligned.h"
+#include "asterisk/module.h"
+#include "asterisk/rtp_engine.h"
+
+#define MAX_TIMESTAMP_SKEW 640
+
+#define RTP_SEQ_MOD (1<<16) /*!< A sequence number can't be more than 16 bits */
+#define RTCP_DEFAULT_INTERVALMS 5000 /*!< Default milli-seconds between RTCP reports we send */
+#define RTCP_MIN_INTERVALMS 500 /*!< Min milli-seconds between RTCP reports we send */
+#define RTCP_MAX_INTERVALMS 60000 /*!< Max milli-seconds between RTCP reports we send */
+
+#define DEFAULT_RTP_START 5000 /*!< Default port number to start allocating RTP ports from */
+#define DEFAULT_RTP_END 31000 /*!< Default maximum port number to end allocating RTP ports at */
+
+#define MINIMUM_RTP_PORT 1024 /*!< Minimum port number to accept */
+#define MAXIMUM_RTP_PORT 65535 /*!< Maximum port number to accept */
+
+#define RTCP_PT_FUR 192
+#define RTCP_PT_SR 200
+#define RTCP_PT_RR 201
+#define RTCP_PT_SDES 202
+#define RTCP_PT_BYE 203
+#define RTCP_PT_APP 204
+
+#define RTP_MTU 1200
+
+#define DEFAULT_DTMF_TIMEOUT 3000 /*!< samples */
+
+#define ZFONE_PROFILE_ID 0x505a
+
+static int dtmftimeout = DEFAULT_DTMF_TIMEOUT;
+
+static int rtpstart = DEFAULT_RTP_START; /*!< First port for RTP sessions (set in rtp.conf) */
+static int rtpend = DEFAULT_RTP_END; /*!< Last port for RTP sessions (set in rtp.conf) */
+static int rtpdebug; /*!< Are we debugging? */
+static int rtcpdebug; /*!< Are we debugging RTCP? */
+static int rtcpstats; /*!< Are we debugging RTCP? */
+static int rtcpinterval = RTCP_DEFAULT_INTERVALMS; /*!< Time between rtcp reports in millisecs */
+static struct sockaddr_in rtpdebugaddr; /*!< Debug packets to/from this host */
+static struct sockaddr_in rtcpdebugaddr; /*!< Debug RTCP packets to/from this host */
+#ifdef SO_NO_CHECK
+static int nochecksums;
+#endif
+static int strictrtp;
+
+enum strict_rtp_state {
+ STRICT_RTP_OPEN = 0, /*! No RTP packets should be dropped, all sources accepted */
+ STRICT_RTP_LEARN, /*! Accept next packet as source */
+ STRICT_RTP_CLOSED, /*! Drop all RTP packets not coming from source that was learned */
+};
+
+#define FLAG_3389_WARNING (1 << 0)
+#define FLAG_NAT_ACTIVE (3 << 1)
+#define FLAG_NAT_INACTIVE (0 << 1)
+#define FLAG_NAT_INACTIVE_NOWARN (1 << 1)
+#define FLAG_NEED_MARKER_BIT (1 << 3)
+#define FLAG_DTMF_COMPENSATE (1 << 4)
+
+/*! \brief RTP session description */
+struct ast_rtp {
+ int s;
+ struct ast_frame f;
+ unsigned char rawdata[8192 + AST_FRIENDLY_OFFSET];
+ unsigned int ssrc; /*!< Synchronization source, RFC 3550, page 10. */
+ unsigned int themssrc; /*!< Their SSRC */
+ unsigned int rxssrc;
+ unsigned int lastts;
+ unsigned int lastrxts;
+ unsigned int lastividtimestamp;
+ unsigned int lastovidtimestamp;
+ unsigned int lastitexttimestamp;
+ unsigned int lastotexttimestamp;
+ unsigned int lasteventseqn;
+ int lastrxseqno; /*!< Last received sequence number */
+ unsigned short seedrxseqno; /*!< What sequence number did they start with?*/
+ unsigned int seedrxts; /*!< What RTP timestamp did they start with? */
+ unsigned int rxcount; /*!< How many packets have we received? */
+ unsigned int rxoctetcount; /*!< How many octets have we received? should be rxcount *160*/
+ unsigned int txcount; /*!< How many packets have we sent? */
+ unsigned int txoctetcount; /*!< How many octets have we sent? (txcount*160)*/
+ unsigned int cycles; /*!< Shifted count of sequence number cycles */
+ double rxjitter; /*!< Interarrival jitter at the moment */
+ double rxtransit; /*!< Relative transit time for previous packet */
+ int lasttxformat;
+ int lastrxformat;
+
+ int rtptimeout; /*!< RTP timeout time (negative or zero means disabled, negative value means temporarily disabled) */
+ int rtpholdtimeout; /*!< RTP timeout when on hold (negative or zero means disabled, negative value means temporarily disabled). */
+ int rtpkeepalive; /*!< Send RTP comfort noice packets for keepalive */
+
+ /* DTMF Reception Variables */
+ char resp;
+ unsigned int lastevent;
+ int dtmfcount;
+ unsigned int dtmfsamples;
+ /* DTMF Transmission Variables */
+ unsigned int lastdigitts;
+ char sending_digit; /*!< boolean - are we sending digits */
+ char send_digit; /*!< digit we are sending */
+ int send_payload;
+ int send_duration;
+ unsigned int flags;
+ struct timeval rxcore;
+ struct timeval txcore;
+ double drxcore; /*!< The double representation of the first received packet */
+ struct timeval lastrx; /*!< timeval when we last received a packet */
+ struct timeval dtmfmute;
+ struct ast_smoother *smoother;
+ int *ioid;
+ unsigned short seqno; /*!< Sequence number, RFC 3550, page 13. */
+ unsigned short rxseqno;
+ struct sched_context *sched;
+ struct io_context *io;
+ void *data;
+ struct ast_rtcp *rtcp;
+ struct ast_rtp *bridged; /*!< Who we are Packet bridged to */
+
+ enum strict_rtp_state strict_rtp_state; /*!< Current state that strict RTP protection is in */
+ struct sockaddr_in strict_rtp_address; /*!< Remote address information for strict RTP purposes */
+
+ struct rtp_red *red;
+};
+
+/*!
+ * \brief Structure defining an RTCP session.
+ *
+ * The concept "RTCP session" is not defined in RFC 3550, but since
+ * this structure is analogous to ast_rtp, which tracks a RTP session,
+ * it is logical to think of this as a RTCP session.
+ *
+ * RTCP packet is defined on page 9 of RFC 3550.
+ *
+ */
+struct ast_rtcp {
+ int rtcp_info;
+ int s; /*!< Socket */
+ struct sockaddr_in us; /*!< Socket representation of the local endpoint. */
+ struct sockaddr_in them; /*!< Socket representation of the remote endpoint. */
+ unsigned int soc; /*!< What they told us */
+ unsigned int spc; /*!< What they told us */
+ unsigned int themrxlsr; /*!< The middle 32 bits of the NTP timestamp in the last received SR*/
+ struct timeval rxlsr; /*!< Time when we got their last SR */
+ struct timeval txlsr; /*!< Time when we sent or last SR*/
+ unsigned int expected_prior; /*!< no. packets in previous interval */
+ unsigned int received_prior; /*!< no. packets received in previous interval */
+ int schedid; /*!< Schedid returned from ast_sched_add() to schedule RTCP-transmissions*/
+ unsigned int rr_count; /*!< number of RRs we've sent, not including report blocks in SR's */
+ unsigned int sr_count; /*!< number of SRs we've sent */
+ unsigned int lastsrtxcount; /*!< Transmit packet count when last SR sent */
+ double accumulated_transit; /*!< accumulated a-dlsr-lsr */
+ double rtt; /*!< Last reported rtt */
+ unsigned int reported_jitter; /*!< The contents of their last jitter entry in the RR */
+ unsigned int reported_lost; /*!< Reported lost packets in their RR */
+
+ double reported_maxjitter;
+ double reported_minjitter;
+ double reported_normdev_jitter;
+ double reported_stdev_jitter;
+ unsigned int reported_jitter_count;
+
+ double reported_maxlost;
+ double reported_minlost;
+ double reported_normdev_lost;
+ double reported_stdev_lost;
+
+ double rxlost;
+ double maxrxlost;
+ double minrxlost;
+ double normdev_rxlost;
+ double stdev_rxlost;
+ unsigned int rxlost_count;
+
+ double maxrxjitter;
+ double minrxjitter;
+ double normdev_rxjitter;
+ double stdev_rxjitter;
+ unsigned int rxjitter_count;
+ double maxrtt;
+ double minrtt;
+ double normdevrtt;
+ double stdevrtt;
+ unsigned int rtt_count;
+};
+
+struct rtp_red {
+ struct ast_frame t140; /*!< Primary data */
+ struct ast_frame t140red; /*!< Redundant t140*/
+ unsigned char pt[AST_RED_MAX_GENERATION]; /*!< Payload types for redundancy data */
+ unsigned char ts[AST_RED_MAX_GENERATION]; /*!< Time stamps */
+ unsigned char len[AST_RED_MAX_GENERATION]; /*!< length of each generation */
+ int num_gen; /*!< Number of generations */
+ int schedid; /*!< Timer id */
+ int ti; /*!< How long to buffer data before send */
+ unsigned char t140red_data[64000];
+ unsigned char buf_data[64000]; /*!< buffered primary data */
+ int hdrlen;
+ long int prev_ts;
+};
+
+/* Forward Declarations */
+static int ast_rtp_new(struct ast_rtp_instance *instance, struct sched_context *sched, struct sockaddr_in *sin, void *data);
+static int ast_rtp_destroy(struct ast_rtp_instance *instance);
+static int ast_rtp_dtmf_begin(struct ast_rtp_instance *instance, char digit);
+static int ast_rtp_dtmf_end(struct ast_rtp_instance *instance, char digit);
+static void ast_rtp_new_source(struct ast_rtp_instance *instance);
+static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame);
+static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtcp);
+static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value);
+static int ast_rtp_fd(struct ast_rtp_instance *instance, int rtcp);
+static void ast_rtp_remote_address_set(struct ast_rtp_instance *instance, struct sockaddr_in *sin);
+static int rtp_red_init(struct ast_rtp_instance *instance, int buffer_time, int *payloads, int generations);
+static int rtp_red_buffer(struct ast_rtp_instance *instance, struct ast_frame *frame);
+static int ast_rtp_local_bridge(struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1);
+static int ast_rtp_get_stat(struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat);
+static int ast_rtp_dtmf_compatible(struct ast_channel *chan0, struct ast_rtp_instance *instance0, struct ast_channel *chan1, struct ast_rtp_instance *instance1);
+static void ast_rtp_stun_request(struct ast_rtp_instance *instance, struct sockaddr_in *suggestion, const char *username);
+static void ast_rtp_stop(struct ast_rtp_instance *instance);
+
+/* RTP Engine Declaration */
+static struct ast_rtp_engine asterisk_rtp_engine = {
+ .name = "asterisk",
+ .new = ast_rtp_new,
+ .destroy = ast_rtp_destroy,
+ .dtmf_begin = ast_rtp_dtmf_begin,
+ .dtmf_end = ast_rtp_dtmf_end,
+ .new_source = ast_rtp_new_source,
+ .write = ast_rtp_write,
+ .read = ast_rtp_read,
+ .prop_set = ast_rtp_prop_set,
+ .fd = ast_rtp_fd,
+ .remote_address_set = ast_rtp_remote_address_set,
+ .red_init = rtp_red_init,
+ .red_buffer = rtp_red_buffer,
+ .local_bridge = ast_rtp_local_bridge,
+ .get_stat = ast_rtp_get_stat,
+ .dtmf_compatible = ast_rtp_dtmf_compatible,
+ .stun_request = ast_rtp_stun_request,
+ .stop = ast_rtp_stop,
+};
+
+static inline int rtp_debug_test_addr(struct sockaddr_in *addr)
+{
+ if (!rtpdebug) {
+ return 0;
+ }
+
+ if (rtpdebugaddr.sin_addr.s_addr) {
+ if (((ntohs(rtpdebugaddr.sin_port) != 0)
+ && (rtpdebugaddr.sin_port != addr->sin_port))
+ || (rtpdebugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
+ return 0;
+ }
+
+ return 1;
+}
+
+static inline int rtcp_debug_test_addr(struct sockaddr_in *addr)
+{
+ if (!rtcpdebug) {
+ return 0;
+ }
+
+ if (rtcpdebugaddr.sin_addr.s_addr) {
+ if (((ntohs(rtcpdebugaddr.sin_port) != 0)
+ && (rtcpdebugaddr.sin_port != addr->sin_port))
+ || (rtcpdebugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
+ return 0;
+ }
+
+ return 1;
+}
+
+static unsigned int ast_rtcp_calc_interval(struct ast_rtp *rtp)
+{
+ unsigned int interval;
+ /*! \todo XXX Do a more reasonable calculation on this one
+ * Look in RFC 3550 Section A.7 for an example*/
+ interval = rtcpinterval;
+ return interval;
+}
+
+/*! \brief Calculate normal deviation */
+static double normdev_compute(double normdev, double sample, unsigned int sample_count)
+{
+ normdev = normdev * sample_count + sample;
+ sample_count++;
+
+ return normdev / sample_count;
+}
+
+static double stddev_compute(double stddev, double sample, double normdev, double normdev_curent, unsigned int sample_count)
+{
+/*
+ for the formula check http://www.cs.umd.edu/~austinjp/constSD.pdf
+ return sqrt( (sample_count*pow(stddev,2) + sample_count*pow((sample-normdev)/(sample_count+1),2) + pow(sample-normdev_curent,2)) / (sample_count+1));
+ we can compute the sigma^2 and that way we would have to do the sqrt only 1 time at the end and would save another pow 2 compute
+ optimized formula
+*/
+#define SQUARE(x) ((x) * (x))
+
+ stddev = sample_count * stddev;
+ sample_count++;
+
+ return stddev +
+ ( sample_count * SQUARE( (sample - normdev) / sample_count ) ) +
+ ( SQUARE(sample - normdev_curent) / sample_count );
+
+#undef SQUARE
+}
+
+static int create_new_socket(const char *type)
+{
+ int sock = socket(AF_INET, SOCK_DGRAM, 0);
+
+ if (sock < 0) {
+ if (!type) {
+ type = "RTP/RTCP";
+ }
+ ast_log(LOG_WARNING, "Unable to allocate %s socket: %s\n", type, strerror(errno));
+ } else {
+ long flags = fcntl(sock, F_GETFL);
+ fcntl(sock, F_SETFL, flags | O_NONBLOCK);
+#ifdef SO_NO_CHECK
+ if (nochecksums) {
+ setsockopt(sock, SOL_SOCKET, SO_NO_CHECK, &nochecksums, sizeof(nochecksums));
+ }
+#endif
+ }
+
+ return sock;
+}
+
+static int ast_rtp_new(struct ast_rtp_instance *instance, struct sched_context *sched, struct sockaddr_in *sin, void *data)
+{
+ struct ast_rtp *rtp = NULL;
+ int x, startplace;
+
+ /* Create a new RTP structure to hold all of our data */
+ if (!(rtp = ast_calloc(1, sizeof(*rtp)))) {
+ return -1;
+ }
+
+ /* Set default parameters on the newly created RTP structure */
+ rtp->ssrc = ast_random();
+ rtp->seqno = ast_random() & 0xffff;
+ rtp->strict_rtp_state = (strictrtp ? STRICT_RTP_LEARN : STRICT_RTP_OPEN);
+
+ /* Create a new socket for us to listen on and use */
+ if ((rtp->s = create_new_socket("RTP")) < 0) {
+ ast_debug(1, "Failed to create a new socket for RTP instance '%p'\n", instance);
+ ast_free(rtp);
+ return -1;
+ }
+
+ /* Now actually find a free RTP port to use */
+ x = (rtpend == rtpstart) ? rtpstart : (ast_random() % (rtpend - rtpstart)) + rtpstart;
+ x = x & ~1;
+ startplace = x;
+
+ for (;;) {
+ struct sockaddr_in local_address = { 0, };
+
+ local_address.sin_port = htons(x);
+ /* Try to bind, this will tell us whether the port is available or not */
+ if (!bind(rtp->s, (struct sockaddr*)&local_address, sizeof(local_address))) {
+ ast_debug(1, "Allocated port %d for RTP instance '%p'\n", x, instance);
+ ast_rtp_instance_set_local_address(instance, &local_address);
+ break;
+ }
+
+ x += 2;
+ if (x > rtpend) {
+ x = (rtpstart + 1) & ~1;
+ }
+
+ /* See if we ran out of ports or if the bind actually failed because of something other than the address being in use */
+ if (x == startplace || errno != EADDRINUSE) {
+ ast_log(LOG_ERROR, "Oh dear... we couldn't allocate a port for RTP instance '%p'\n", instance);
+ return -1;
+ }
+ }
+
+ /* Record any information we may need */
+ rtp->sched = sched;
+
+ /* Associate the RTP structure with the RTP instance and be done */
+ ast_rtp_instance_set_data(instance, rtp);
+
+ return 0;
+}
+
+static int ast_rtp_destroy(struct ast_rtp_instance *instance)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+
+ /* Destroy the smoother that was smoothing out audio if present */
+ if (rtp->smoother) {
+ ast_smoother_free(rtp->smoother);
+ }
+
+ /* Close our own socket so we no longer get packets */
+ if (rtp->s > -1) {
+ close(rtp->s);
+ }
+
+ /* Destroy RTCP if it was being used */
+ if (rtp->rtcp) {
+ AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
+ close(rtp->rtcp->s);
+ ast_free(rtp->rtcp);
+ }
+
+ /* Destroy RED if it was being used */
+ if (rtp->red) {
+ AST_SCHED_DEL(rtp->sched, rtp->red->schedid);
+ ast_free(rtp->red);
+ }
+
+ /* Finally destroy ourselves */
+ ast_free(rtp);
+
+ return 0;
+}
+
+static int ast_rtp_dtmf_begin(struct ast_rtp_instance *instance, char digit)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+ struct sockaddr_in remote_address;
+ int hdrlen = 12, res = 0, i = 0, payload = 101;
+ char data[256];
+ unsigned int *rtpheader = (unsigned int*)data;
+
+ ast_rtp_instance_get_remote_address(instance, &remote_address);
+
+ /* If we have no remote address information bail out now */
+ if (!remote_address.sin_addr.s_addr || !remote_address.sin_port) {
+ return -1;
+ }
+
+ /* Convert given digit into what we want to transmit */
+ if ((digit <= '9') && (digit >= '0')) {
+ digit -= '0';
+ } else if (digit == '*') {
+ digit = 10;
+ } else if (digit == '#') {
+ digit = 11;
+ } else if ((digit >= 'A') && (digit <= 'D')) {
+ digit = digit - 'A' + 12;
+ } else if ((digit >= 'a') && (digit <= 'd')) {
+ digit = digit - 'a' + 12;
+ } else {
+ ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
+ return -1;
+ }
+
+ /* Grab the payload that they expect the RFC2833 packet to be received in */
+ payload = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance), 0, AST_RTP_DTMF);
+
+ rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
+ rtp->send_duration = 160;
+ rtp->lastdigitts = rtp->lastts + rtp->send_duration;
+
+ /* Create the actual packet that we will be sending */
+ rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno));
+ rtpheader[1] = htonl(rtp->lastdigitts);
+ rtpheader[2] = htonl(rtp->ssrc);
+
+ /* Actually send the packet */
+ for (i = 0; i < 2; i++) {
+ rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration));
+ res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &remote_address, sizeof(remote_address));
+ if (res < 0) {
+ ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n",
+ ast_inet_ntoa(remote_address.sin_addr), ntohs(remote_address.sin_port), strerror(errno));
+ }
+ if (rtp_debug_test_addr(&remote_address)) {
+ ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
+ ast_inet_ntoa(remote_address.sin_addr),
+ ntohs(remote_address.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
+ }
+ rtp->seqno++;
+ rtp->send_duration += 160;
+ rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno));
+ }
+
+ /* Record that we are in the process of sending a digit and information needed to continue doing so */
+ rtp->sending_digit = 1;
+ rtp->send_digit = digit;
+ rtp->send_payload = payload;
+
+ return 0;
+}
+
+static int ast_rtp_dtmf_continuation(struct ast_rtp_instance *instance)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+ struct sockaddr_in remote_address;
+ int hdrlen = 12, res = 0;
+ char data[256];
+ unsigned int *rtpheader = (unsigned int*)data;
+
+ ast_rtp_instance_get_remote_address(instance, &remote_address);
+
+ /* Make sure we know where the other side is so we can send them the packet */
+ if (!remote_address.sin_addr.s_addr || !remote_address.sin_port) {
+ return -1;
+ }
+
+ /* Actually create the packet we will be sending */
+ rtpheader[0] = htonl((2 << 30) | (1 << 23) | (rtp->send_payload << 16) | (rtp->seqno));
+ rtpheader[1] = htonl(rtp->lastdigitts);
+ rtpheader[2] = htonl(rtp->ssrc);
+ rtpheader[3] = htonl((rtp->send_digit << 24) | (0xa << 16) | (rtp->send_duration));
+ rtpheader[0] = htonl((2 << 30) | (rtp->send_payload << 16) | (rtp->seqno));
+
+ /* Boom, send it on out */
+ res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &remote_address, sizeof(remote_address));
+ if (res < 0) {
+ ast_log(LOG_ERROR, "RTP Transmission error to %s:%d: %s\n",
+ ast_inet_ntoa(remote_address.sin_addr),
+ ntohs(remote_address.sin_port), strerror(errno));
+ }
+
+ if (rtp_debug_test_addr(&remote_address)) {
+ ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
+ ast_inet_ntoa(remote_address.sin_addr),
+ ntohs(remote_address.sin_port), rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
+ }
+
+ /* And now we increment some values for the next time we swing by */
+ rtp->seqno++;
+ rtp->send_duration += 160;
+
+ return 0;
+}
+
+static int ast_rtp_dtmf_end(struct ast_rtp_instance *instance, char digit)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+ struct sockaddr_in remote_address;
+ int hdrlen = 12, res = 0, i = 0;
+ char data[256];
+ unsigned int *rtpheader = (unsigned int*)data;
+
+ ast_rtp_instance_get_remote_address(instance, &remote_address);
+
+ /* Make sure we know where the remote side is so we can send them the packet we construct */
+ if (!remote_address.sin_addr.s_addr || !remote_address.sin_port) {
+ return -1;
+ }
+
+ /* Convert the given digit to the one we are going to send */
+ if ((digit <= '9') && (digit >= '0')) {
+ digit -= '0';
+ } else if (digit == '*') {
+ digit = 10;
+ } else if (digit == '#') {
+ digit = 11;
+ } else if ((digit >= 'A') && (digit <= 'D')) {
+ digit = digit - 'A' + 12;
+ } else if ((digit >= 'a') && (digit <= 'd')) {
+ digit = digit - 'a' + 12;
+ } else {
+ ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
+ return -1;
+ }
+
+ rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
+
+ /* Construct the packet we are going to send */
+ rtpheader[0] = htonl((2 << 30) | (1 << 23) | (rtp->send_payload << 16) | (rtp->seqno));
+ rtpheader[1] = htonl(rtp->lastdigitts);
+ rtpheader[2] = htonl(rtp->ssrc);
+ rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration));
+ rtpheader[3] |= htonl((1 << 23));
+ rtpheader[0] = htonl((2 << 30) | (rtp->send_payload << 16) | (rtp->seqno));
+
+ /* Send it 3 times, that's the magical number */
+ for (i = 0; i < 3; i++) {
+ res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &remote_address, sizeof(remote_address));
+ if (res < 0) {
+ ast_log(LOG_ERROR, "RTP Transmission error to %s:%d: %s\n",
+ ast_inet_ntoa(remote_address.sin_addr),
+ ntohs(remote_address.sin_port), strerror(errno));
+ }
+ if (rtp_debug_test_addr(&remote_address)) {
+ ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
+ ast_inet_ntoa(remote_address.sin_addr),
+ ntohs(remote_address.sin_port), rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
+ }
+ }
+
+ /* Oh and we can't forget to turn off the stuff that says we are sending DTMF */
+ rtp->lastts += rtp->send_duration;
+ rtp->sending_digit = 0;
+ rtp->send_digit = 0;
+
+ return 0;
+}
+
+static void ast_rtp_new_source(struct ast_rtp_instance *instance)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+
+ /* We simply set this bit so that the next packet sent will have the marker bit turned on */
+ ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
+
+ return;
+}
+
+static unsigned int calc_txstamp(struct ast_rtp *rtp, struct timeval *delivery)
+{
+ struct timeval t;
+ long ms;
+
+ if (ast_tvzero(rtp->txcore)) {
+ rtp->txcore = ast_tvnow();
+ rtp->txcore.tv_usec -= rtp->txcore.tv_usec % 20000;
+ }
+
+ t = (delivery && !ast_tvzero(*delivery)) ? *delivery : ast_tvnow();
+ if ((ms = ast_tvdiff_ms(t, rtp->txcore)) < 0) {
+ ms = 0;
+ }
+ rtp->txcore = t;
+
+ return (unsigned int) ms;
+}
+
+static void timeval2ntp(struct timeval tv, unsigned int *msw, unsigned int *lsw)
+{
+ unsigned int sec, usec, frac;
+ sec = tv.tv_sec + 2208988800u; /* Sec between 1900 and 1970 */
+ usec = tv.tv_usec;
+ frac = (usec << 12) + (usec << 8) - ((usec * 3650) >> 6);
+ *msw = sec;
+ *lsw = frac;
+}
+
+/*! \brief Send RTCP recipient's report */
+static int ast_rtcp_write_rr(const void *data)
+{
+ struct ast_rtp *rtp = (struct ast_rtp *)data;
+ int res;
+ int len = 32;
+ unsigned int lost;
+ unsigned int extended;
+ unsigned int expected;
+ unsigned int expected_interval;
+ unsigned int received_interval;
+ int lost_interval;
+ struct timeval now;
+ unsigned int *rtcpheader;
+ char bdata[1024];
+ struct timeval dlsr;
+ int fraction;
+
+ double rxlost_current;
+
+ if (!rtp || !rtp->rtcp || (&rtp->rtcp->them.sin_addr == 0))
+ return 0;
+
+ if (!rtp->rtcp->them.sin_addr.s_addr) {
+ ast_log(LOG_ERROR, "RTCP RR transmission error, rtcp halted\n");
+ AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
+ return 0;
+ }
+
+ extended = rtp->cycles + rtp->lastrxseqno;
+ expected = extended - rtp->seedrxseqno + 1;
+ lost = expected - rtp->rxcount;
+ expected_interval = expected - rtp->rtcp->expected_prior;
+ rtp->rtcp->expected_prior = expected;
+ received_interval = rtp->rxcount - rtp->rtcp->received_prior;
+ rtp->rtcp->received_prior = rtp->rxcount;
+ lost_interval = expected_interval - received_interval;
+
+ if (lost_interval <= 0)
+ rtp->rtcp->rxlost = 0;
+ else rtp->rtcp->rxlost = rtp->rtcp->rxlost;
+ if (rtp->rtcp->rxlost_count == 0)
+ rtp->rtcp->minrxlost = rtp->rtcp->rxlost;
+ if (lost_interval < rtp->rtcp->minrxlost)
+ rtp->rtcp->minrxlost = rtp->rtcp->rxlost;
+ if (lost_interval > rtp->rtcp->maxrxlost)
+ rtp->rtcp->maxrxlost = rtp->rtcp->rxlost;
+
+ rxlost_current = normdev_compute(rtp->rtcp->normdev_rxlost, rtp->rtcp->rxlost, rtp->rtcp->rxlost_count);
+ rtp->rtcp->stdev_rxlost = stddev_compute(rtp->rtcp->stdev_rxlost, rtp->rtcp->rxlost, rtp->rtcp->normdev_rxlost, rxlost_current, rtp->rtcp->rxlost_count);
+ rtp->rtcp->normdev_rxlost = rxlost_current;
+ rtp->rtcp->rxlost_count++;
+
+ if (expected_interval == 0 || lost_interval <= 0)
+ fraction = 0;
+ else
+ fraction = (lost_interval << 8) / expected_interval;
+ gettimeofday(&now, NULL);
+ timersub(&now, &rtp->rtcp->rxlsr, &dlsr);
+ rtcpheader = (unsigned int *)bdata;
+ rtcpheader[0] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_RR << 16) | ((len/4)-1));
+ rtcpheader[1] = htonl(rtp->ssrc);
+ rtcpheader[2] = htonl(rtp->themssrc);
+ rtcpheader[3] = htonl(((fraction & 0xff) << 24) | (lost & 0xffffff));
+ rtcpheader[4] = htonl((rtp->cycles) | ((rtp->lastrxseqno & 0xffff)));
+ rtcpheader[5] = htonl((unsigned int)(rtp->rxjitter * 65536.));
+ rtcpheader[6] = htonl(rtp->rtcp->themrxlsr);
+ rtcpheader[7] = htonl((((dlsr.tv_sec * 1000) + (dlsr.tv_usec / 1000)) * 65536) / 1000);
+
+ /*! \note Insert SDES here. Probably should make SDES text equal to mimetypes[code].type (not subtype 'cos
+ it can change mid call, and SDES can't) */
+ rtcpheader[len/4] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_SDES << 16) | 2);
+ rtcpheader[(len/4)+1] = htonl(rtp->ssrc); /* Our SSRC */
+ rtcpheader[(len/4)+2] = htonl(0x01 << 24); /* Empty for the moment */
+ len += 12;
+
+ res = sendto(rtp->rtcp->s, (unsigned int *)rtcpheader, len, 0, (struct sockaddr *)&rtp->rtcp->them, sizeof(rtp->rtcp->them));
+
+ if (res < 0) {
+ ast_log(LOG_ERROR, "RTCP RR transmission error, rtcp halted: %s\n",strerror(errno));
+ /* Remove the scheduler */
+ AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
+ return 0;
+ }
+
+ rtp->rtcp->rr_count++;
+ if (rtcp_debug_test_addr(&rtp->rtcp->them)) {
+ ast_verbose("\n* Sending RTCP RR to %s:%d\n"
+ " Our SSRC: %u\nTheir SSRC: %u\niFraction lost: %d\nCumulative loss: %u\n"
+ " IA jitter: %.4f\n"
+ " Their last SR: %u\n"
+ " DLSR: %4.4f (sec)\n\n",
+ ast_inet_ntoa(rtp->rtcp->them.sin_addr),
+ ntohs(rtp->rtcp->them.sin_port),
+ rtp->ssrc, rtp->themssrc, fraction, lost,
+ rtp->rxjitter,
+ rtp->rtcp->themrxlsr,
+ (double)(ntohl(rtcpheader[7])/65536.0));
+ }
+
+ return res;
+}
+
+/*! \brief Send RTCP sender's report */
+static int ast_rtcp_write_sr(const void *data)
+{
+ struct ast_rtp *rtp = (struct ast_rtp *)data;
+ int res;
+ int len = 0;
+ struct timeval now;
+ unsigned int now_lsw;
+ unsigned int now_msw;
+ unsigned int *rtcpheader;
+ unsigned int lost;
+ unsigned int extended;
+ unsigned int expected;
+ unsigned int expected_interval;
+ unsigned int received_interval;
+ int lost_interval;
+ int fraction;
+ struct timeval dlsr;
+ char bdata[512];
+
+ /* Commented condition is always not NULL if rtp->rtcp is not NULL */
+ if (!rtp || !rtp->rtcp/* || (&rtp->rtcp->them.sin_addr == 0)*/)
+ return 0;
+
+ if (!rtp->rtcp->them.sin_addr.s_addr) { /* This'll stop rtcp for this rtp session */
+ ast_verbose("RTCP SR transmission error, rtcp halted\n");
+ AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
+ return 0;
+ }
+
+ gettimeofday(&now, NULL);
+ timeval2ntp(now, &now_msw, &now_lsw); /* fill thses ones in from utils.c*/
+ rtcpheader = (unsigned int *)bdata;
+ rtcpheader[1] = htonl(rtp->ssrc); /* Our SSRC */
+ rtcpheader[2] = htonl(now_msw); /* now, MSW. gettimeofday() + SEC_BETWEEN_1900_AND_1970*/
+ rtcpheader[3] = htonl(now_lsw); /* now, LSW */
+ rtcpheader[4] = htonl(rtp->lastts); /* FIXME shouldn't be that, it should be now */
+ rtcpheader[5] = htonl(rtp->txcount); /* No. packets sent */
+ rtcpheader[6] = htonl(rtp->txoctetcount); /* No. bytes sent */
+ len += 28;
+
+ extended = rtp->cycles + rtp->lastrxseqno;
+ expected = extended - rtp->seedrxseqno + 1;
+ if (rtp->rxcount > expected)
+ expected += rtp->rxcount - expected;
+ lost = expected - rtp->rxcount;
+ expected_interval = expected - rtp->rtcp->expected_prior;
+ rtp->rtcp->expected_prior = expected;
+ received_interval = rtp->rxcount - rtp->rtcp->received_prior;
+ rtp->rtcp->received_prior = rtp->rxcount;
+ lost_interval = expected_interval - received_interval;
+ if (expected_interval == 0 || lost_interval <= 0)
+ fraction = 0;
+ else
+ fraction = (lost_interval << 8) / expected_interval;
+ timersub(&now, &rtp->rtcp->rxlsr, &dlsr);
+ rtcpheader[7] = htonl(rtp->themssrc);
+ rtcpheader[8] = htonl(((fraction & 0xff) << 24) | (lost & 0xffffff));
+ rtcpheader[9] = htonl((rtp->cycles) | ((rtp->lastrxseqno & 0xffff)));
+ rtcpheader[10] = htonl((unsigned int)(rtp->rxjitter * 65536.));
+ rtcpheader[11] = htonl(rtp->rtcp->themrxlsr);
+ rtcpheader[12] = htonl((((dlsr.tv_sec * 1000) + (dlsr.tv_usec / 1000)) * 65536) / 1000);
+ len += 24;
+
+ rtcpheader[0] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_SR << 16) | ((len/4)-1));
+
+ /* Insert SDES here. Probably should make SDES text equal to mimetypes[code].type (not subtype 'cos */
+ /* it can change mid call, and SDES can't) */
+ rtcpheader[len/4] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_SDES << 16) | 2);
+ rtcpheader[(len/4)+1] = htonl(rtp->ssrc); /* Our SSRC */
+ rtcpheader[(len/4)+2] = htonl(0x01 << 24); /* Empty for the moment */
+ len += 12;
+
+ res = sendto(rtp->rtcp->s, (unsigned int *)rtcpheader, len, 0, (struct sockaddr *)&rtp->rtcp->them, sizeof(rtp->rtcp->them));
+ if (res < 0) {
+ ast_log(LOG_ERROR, "RTCP SR transmission error to %s:%d, rtcp halted %s\n",ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port), strerror(errno));
+ AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
+ return 0;
+ }
+
+ /* FIXME Don't need to get a new one */
+ gettimeofday(&rtp->rtcp->txlsr, NULL);
+ rtp->rtcp->sr_count++;
+
+ rtp->rtcp->lastsrtxcount = rtp->txcount;
+
+ if (rtcp_debug_test_addr(&rtp->rtcp->them)) {
+ ast_verbose("* Sent RTCP SR to %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
+ ast_verbose(" Our SSRC: %u\n", rtp->ssrc);
+ ast_verbose(" Sent(NTP): %u.%010u\n", (unsigned int)now.tv_sec, (unsigned int)now.tv_usec*4096);
+ ast_verbose(" Sent(RTP): %u\n", rtp->lastts);
+ ast_verbose(" Sent packets: %u\n", rtp->txcount);
+ ast_verbose(" Sent octets: %u\n", rtp->txoctetcount);
+ ast_verbose(" Report block:\n");
+ ast_verbose(" Fraction lost: %u\n", fraction);
+ ast_verbose(" Cumulative loss: %u\n", lost);
+ ast_verbose(" IA jitter: %.4f\n", rtp->rxjitter);
+ ast_verbose(" Their last SR: %u\n", rtp->rtcp->themrxlsr);
+ ast_verbose(" DLSR: %4.4f (sec)\n\n", (double)(ntohl(rtcpheader[12])/65536.0));
+ }
+ manager_event(EVENT_FLAG_REPORTING, "RTCPSent", "To %s:%d\r\n"
+ "OurSSRC: %u\r\n"
+ "SentNTP: %u.%010u\r\n"
+ "SentRTP: %u\r\n"
+ "SentPackets: %u\r\n"
+ "SentOctets: %u\r\n"
+ "ReportBlock:\r\n"
+ "FractionLost: %u\r\n"
+ "CumulativeLoss: %u\r\n"
+ "IAJitter: %.4f\r\n"
+ "TheirLastSR: %u\r\n"
+ "DLSR: %4.4f (sec)\r\n",
+ ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port),
+ rtp->ssrc,
+ (unsigned int)now.tv_sec, (unsigned int)now.tv_usec*4096,
+ rtp->lastts,
+ rtp->txcount,
+ rtp->txoctetcount,
+ fraction,
+ lost,
+ rtp->rxjitter,
+ rtp->rtcp->themrxlsr,
+ (double)(ntohl(rtcpheader[12])/65536.0));
+ return res;
+}
+
+/*! \brief Write and RTCP packet to the far end
+ * \note Decide if we are going to send an SR (with Reception Block) or RR
+ * RR is sent if we have not sent any rtp packets in the previous interval */
+static int ast_rtcp_write(const void *data)
+{
+ struct ast_rtp *rtp = (struct ast_rtp *)data;
+ int res;
+
+ if (!rtp || !rtp->rtcp)
+ return 0;
+
+ if (rtp->txcount > rtp->rtcp->lastsrtxcount)
+ res = ast_rtcp_write_sr(data);
+ else
+ res = ast_rtcp_write_rr(data);
+
+ return res;
+}
+
+static int ast_rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame *frame, int codec)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+ int pred, mark = 0;
+ unsigned int ms = calc_txstamp(rtp, &frame->delivery);
+ struct sockaddr_in remote_address;
+
+ if (rtp->sending_digit) {
+ return 0;
+ }
+
+ if (frame->frametype == AST_FRAME_VOICE) {
+ pred = rtp->lastts + frame->samples;
+
+ /* Re-calculate last TS */
+ rtp->lastts = rtp->lastts + ms * 8;
+ if (ast_tvzero(frame->delivery)) {
+ /* If this isn't an absolute delivery time, Check if it is close to our prediction,
+ and if so, go with our prediction */
+ if (abs(rtp->lastts - pred) < MAX_TIMESTAMP_SKEW) {
+ rtp->lastts = pred;
+ } else {
+ ast_debug(3, "Difference is %d, ms is %d\n", abs(rtp->lastts - pred), ms);
+ mark = 1;
+ }
+ }
+ } else if (frame->frametype == AST_FRAME_VIDEO) {
+ mark = frame->subclass & 0x1;
+ pred = rtp->lastovidtimestamp + frame->samples;
+ /* Re-calculate last TS */
+ rtp->lastts = rtp->lastts + ms * 90;
+ /* If it's close to our prediction, go for it */
+ if (ast_tvzero(frame->delivery)) {
+ if (abs(rtp->lastts - pred) < 7200) {
+ rtp->lastts = pred;
+ rtp->lastovidtimestamp += frame->samples;
+ } else {
+ ast_debug(3, "Difference is %d, ms is %d (%d), pred/ts/samples %d/%d/%d\n", abs(rtp->lastts - pred), ms, ms * 90, rtp->lastts, pred, frame->samples);
+ rtp->lastovidtimestamp = rtp->lastts;
+ }
+ }
+ } else {
+ pred = rtp->lastotexttimestamp + frame->samples;
+ /* Re-calculate last TS */
+ rtp->lastts = rtp->lastts + ms * 90;
+ /* If it's close to our prediction, go for it */
+ if (ast_tvzero(frame->delivery)) {
+ if (abs(rtp->lastts - pred) < 7200) {
+ rtp->lastts = pred;
+ rtp->lastotexttimestamp += frame->samples;
+ } else {
+ ast_debug(3, "Difference is %d, ms is %d (%d), pred/ts/samples %d/%d/%d\n", abs(rtp->lastts - pred), ms, ms * 90, rtp->lastts, pred, frame->samples);
+ rtp->lastotexttimestamp = rtp->lastts;
+ }
+ }
+ }
+
+ /* If we have been explicitly told to set the marker bit then do so */
+ if (ast_test_flag(rtp, FLAG_NEED_MARKER_BIT)) {
+ mark = 1;
+ ast_clear_flag(rtp, FLAG_NEED_MARKER_BIT);
+ }
+
+ /* If the timestamp for non-digt packets has moved beyond the timestamp for digits, update the digit timestamp */
+ if (rtp->lastts > rtp->lastdigitts) {
+ rtp->lastdigitts = rtp->lastts;
+ }
+
+ if (ast_test_flag(frame, AST_FRFLAG_HAS_TIMING_INFO)) {
+ rtp->lastts = frame->ts * 8;
+ }
+
+ ast_rtp_instance_get_remote_address(instance, &remote_address);
+
+ /* If we know the remote address construct a packet and send it out */
+ if (remote_address.sin_port && remote_address.sin_addr.s_addr) {
+ int hdrlen = 12, res;
+ unsigned char *rtpheader = (unsigned char *)(frame->data.ptr - hdrlen);
+
+ put_unaligned_uint32(rtpheader, htonl((2 << 30) | (codec << 16) | (rtp->seqno) | (mark << 23)));
+ put_unaligned_uint32(rtpheader + 4, htonl(rtp->lastts));
+ put_unaligned_uint32(rtpheader + 8, htonl(rtp->ssrc));
+
+ if ((res = sendto(rtp->s, (void *)rtpheader, frame->datalen + hdrlen, 0, (struct sockaddr *)&remote_address, sizeof(remote_address))) < 0) {
+ if (!ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT) || (ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT) && (ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_ACTIVE))) {
+ ast_debug(1, "RTP Transmission error of packet %d to %s:%d: %s\n", rtp->seqno, ast_inet_ntoa(remote_address.sin_addr), ntohs(remote_address.sin_port), strerror(errno));
+ } else if (((ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_INACTIVE) || rtpdebug) && !ast_test_flag(rtp, FLAG_NAT_INACTIVE_NOWARN)) {
+ /* Only give this error message once if we are not RTP debugging */
+ if (option_debug || rtpdebug)
+ ast_debug(0, "RTP NAT: Can't write RTP to private address %s:%d, waiting for other end to send audio...\n", ast_inet_ntoa(remote_address.sin_addr), ntohs(remote_address.sin_port));
+ ast_set_flag(rtp, FLAG_NAT_INACTIVE_NOWARN);
+ }
+ } else {
+ rtp->txcount++;
+ rtp->txoctetcount += (res - hdrlen);
+
+ if (rtp->rtcp && rtp->rtcp->schedid < 1) {
+ ast_debug(1, "Starting RTCP transmission on RTP instance '%p'\n", instance);
+ rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp);
+ }
+ }
+
+ if (rtp_debug_test_addr(&remote_address)) {
+ ast_verbose("Sent RTP packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
+ ast_inet_ntoa(remote_address.sin_addr), ntohs(remote_address.sin_port), codec, rtp->seqno, rtp->lastts, res - hdrlen);
+ }
+ }
+
+ rtp->seqno++;
+
+ return 0;
+}
+
+static struct ast_frame *red_t140_to_red(struct rtp_red *red) {
+ unsigned char *data = red->t140red.data.ptr;
+ int len = 0;
+ int i;
+
+ /* replace most aged generation */
+ if (red->len[0]) {
+ for (i = 1; i < red->num_gen+1; i++)
+ len += red->len[i];
+
+ memmove(&data[red->hdrlen], &data[red->hdrlen+red->len[0]], len);
+ }
+
+ /* Store length of each generation and primary data length*/
+ for (i = 0; i < red->num_gen; i++)
+ red->len[i] = red->len[i+1];
+ red->len[i] = red->t140.datalen;
+
+ /* write each generation length in red header */
+ len = red->hdrlen;
+ for (i = 0; i < red->num_gen; i++)
+ len += data[i*4+3] = red->len[i];
+
+ /* add primary data to buffer */
+ memcpy(&data[len], red->t140.data.ptr, red->t140.datalen);
+ red->t140red.datalen = len + red->t140.datalen;
+
+ /* no primary data and no generations to send */
+ if (len == red->hdrlen && !red->t140.datalen)
+ return NULL;
+
+ /* reset t.140 buffer */
+ red->t140.datalen = 0;
+
+ return &red->t140red;
+}
+
+static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+ struct sockaddr_in remote_address;
+ int codec, subclass;
+
+ ast_rtp_instance_get_remote_address(instance, &remote_address);
+
+ /* If we don't actually know the remote address don't even bother doing anything */
+ if (!remote_address.sin_addr.s_addr) {
+ ast_debug(1, "No remote address on RTP instance '%p' so dropping frame\n", instance);
+ return -1;
+ }
+
+ /* If there is no data length we can't very well send the packet */
+ if (!frame->datalen) {
+ ast_debug(1, "Received frame with no data for RTP instance '%p' so dropping frame\n", instance);
+ return -1;
+ }
+
+ /* If the packet is not one our RTP stack supports bail out */
+ if (frame->frametype != AST_FRAME_VOICE && frame->frametype != AST_FRAME_VIDEO && frame->frametype != AST_FRAME_TEXT) {
+ ast_log(LOG_WARNING, "RTP can only send voice, video, and text\n");
+ return -1;
+ }
+
+ if (rtp->red) {
+ /* return 0; */
+ /* no primary data or generations to send */
+ if ((frame = red_t140_to_red(rtp->red)) == NULL)
+ return 0;
+ }
+
+ /* Grab the subclass and look up the payload we are going to use */
+ subclass = frame->subclass;
+ if (frame->frametype == AST_FRAME_VIDEO) {
+ subclass &= ~0x1;
+ }
+ if ((codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance), 1, subclass)) < 0) {
+ ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(frame->subclass));
+ return -1;
+ }
+
+ /* Oh dear, if the format changed we will have to set up a new smoother */
+ if (rtp->lasttxformat != subclass) {
+ ast_debug(1, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass));
+ rtp->lasttxformat = subclass;
+ if (rtp->smoother) {
+ ast_smoother_free(rtp->smoother);
+ rtp->smoother = NULL;
+ }
+ }
+
+ /* If no smoother is present see if we have to set one up */
+ if (!rtp->smoother) {
+ struct ast_format_list fmt = ast_codec_pref_getsize(&ast_rtp_instance_get_codecs(instance)->pref, subclass);
+
+ switch (subclass) {
+ case AST_FORMAT_SPEEX:
+ case AST_FORMAT_G723_1:
+ case AST_FORMAT_SIREN7:
+ case AST_FORMAT_SIREN14:
+ /* these are all frame-based codecs and cannot be safely run through
+ a smoother */
+ break;
+ default:
+ if (fmt.inc_ms) {
+ if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) {
+ ast_log(LOG_WARNING, "Unable to create smoother: format %d ms: %d len: %d\n", subclass, fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms));
+ return -1;
+ }
+ if (fmt.flags) {
+ ast_smoother_set_flags(rtp->smoother, fmt.flags);
+ }
+ ast_debug(1, "Created smoother: format: %d ms: %d len: %d\n", subclass, fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms));
+ }
+ }
+ }
+
+ /* Feed audio frames into the actual function that will create a frame and send it */
+ if (rtp->smoother) {
+ struct ast_frame *f;
+
+ if (ast_smoother_test_flag(rtp->smoother, AST_SMOOTHER_FLAG_BE)) {
+ ast_smoother_feed_be(rtp->smoother, frame);
+ } else {
+ ast_smoother_feed(rtp->smoother, frame);
+ }
+
+ while ((f = ast_smoother_read(rtp->smoother)) && (f->data.ptr)) {
+ if (f->subclass == AST_FORMAT_G722) {
+ f->samples /= 2;
+ }
+
+ ast_rtp_raw_write(instance, f, codec);
+ }
+ } else {
+ int hdrlen = 12;
+ struct ast_frame *f = NULL;
+
+ if (frame->offset < hdrlen) {
+ f = ast_frdup(frame);
+ } else {
+ f = frame;
+ }
+ if (f->data.ptr) {
+ ast_rtp_raw_write(instance, f, codec);
+ }
+ if (f != frame) {
+ ast_frfree(f);
+ }
+
+ }
+
+ return 0;
+}
+
+static void calc_rxstamp(struct timeval *tv, struct ast_rtp *rtp, unsigned int timestamp, int mark)
+{
+ struct timeval now;
+ double transit;
+ double current_time;
+ double d;
+ double dtv;
+ double prog;
+
+ double normdev_rxjitter_current;
+ if ((!rtp->rxcore.tv_sec && !rtp->rxcore.tv_usec) || mark) {
+ gettimeofday(&rtp->rxcore, NULL);
+ rtp->drxcore = (double) rtp->rxcore.tv_sec + (double) rtp->rxcore.tv_usec / 1000000;
+ /* map timestamp to a real time */
+ rtp->seedrxts = timestamp; /* Their RTP timestamp started with this */
+ rtp->rxcore.tv_sec -= timestamp / 8000;
+ rtp->rxcore.tv_usec -= (timestamp % 8000) * 125;
+ /* Round to 0.1ms for nice, pretty timestamps */
+ rtp->rxcore.tv_usec -= rtp->rxcore.tv_usec % 100;
+ if (rtp->rxcore.tv_usec < 0) {
+ /* Adjust appropriately if necessary */
+ rtp->rxcore.tv_usec += 1000000;
+ rtp->rxcore.tv_sec -= 1;
+ }
+ }
+
+ gettimeofday(&now,NULL);
+ /* rxcore is the mapping between the RTP timestamp and _our_ real time from gettimeofday() */
+ tv->tv_sec = rtp->rxcore.tv_sec + timestamp / 8000;
+ tv->tv_usec = rtp->rxcore.tv_usec + (timestamp % 8000) * 125;
+ if (tv->tv_usec >= 1000000) {
+ tv->tv_usec -= 1000000;
+ tv->tv_sec += 1;
+ }
+ prog = (double)((timestamp-rtp->seedrxts)/8000.);
+ dtv = (double)rtp->drxcore + (double)(prog);
+ current_time = (double)now.tv_sec + (double)now.tv_usec/1000000;
+ transit = current_time - dtv;
+ d = transit - rtp->rxtransit;
+ rtp->rxtransit = transit;
+ if (d<0)
+ d=-d;
+ rtp->rxjitter += (1./16.) * (d - rtp->rxjitter);
+
+ if (rtp->rtcp) {
+ if (rtp->rxjitter > rtp->rtcp->maxrxjitter)
+ rtp->rtcp->maxrxjitter = rtp->rxjitter;
+ if (rtp->rtcp->rxjitter_count == 1)
+ rtp->rtcp->minrxjitter = rtp->rxjitter;
+ if (rtp->rtcp && rtp->rxjitter < rtp->rtcp->minrxjitter)
+ rtp->rtcp->minrxjitter = rtp->rxjitter;
+
+ normdev_rxjitter_current = normdev_compute(rtp->rtcp->normdev_rxjitter,rtp->rxjitter,rtp->rtcp->rxjitter_count);
+ rtp->rtcp->stdev_rxjitter = stddev_compute(rtp->rtcp->stdev_rxjitter,rtp->rxjitter,rtp->rtcp->normdev_rxjitter,normdev_rxjitter_current,rtp->rtcp->rxjitter_count);
+
+ rtp->rtcp->normdev_rxjitter = normdev_rxjitter_current;
+ rtp->rtcp->rxjitter_count++;
+ }
+}
+
+static struct ast_frame *send_dtmf(struct ast_rtp_instance *instance, enum ast_frame_type type, int compensate)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+ struct sockaddr_in remote_address;
+
+ ast_rtp_instance_get_remote_address(instance, &remote_address);
+
+ if (((compensate && type == AST_FRAME_DTMF_END) || (type == AST_FRAME_DTMF_BEGIN)) && ast_tvcmp(ast_tvnow(), rtp->dtmfmute) < 0) {
+ ast_debug(1, "Ignore potential DTMF echo from '%s'\n", ast_inet_ntoa(remote_address.sin_addr));
+ rtp->resp = 0;
+ rtp->dtmfsamples = 0;
+ return &ast_null_frame;
+ }
+ ast_debug(1, "Sending dtmf: %d (%c), at %s\n", rtp->resp, rtp->resp, ast_inet_ntoa(remote_address.sin_addr));
+ if (rtp->resp == 'X') {
+ rtp->f.frametype = AST_FRAME_CONTROL;
+ rtp->f.subclass = AST_CONTROL_FLASH;
+ } else {
+ rtp->f.frametype = type;
+ rtp->f.subclass = rtp->resp;
+ }
+ rtp->f.datalen = 0;
+ rtp->f.samples = 0;
+ rtp->f.mallocd = 0;
+ rtp->f.src = "RTP";
+
+ return &rtp->f;
+}
+
+static struct ast_frame *process_dtmf_rfc2833(struct ast_rtp_instance *instance, unsigned char *data, int len, unsigned int seqno, unsigned int timestamp, struct sockaddr_in *sin, int payloadtype, int mark)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+ struct sockaddr_in remote_address;
+ unsigned int event, event_end, samples;
+ char resp = 0;
+ struct ast_frame *f = NULL;
+
+ ast_rtp_instance_get_remote_address(instance, &remote_address);
+
+ /* Figure out event, event end, and samples */
+ event = ntohl(*((unsigned int *)(data)));
+ event >>= 24;
+ event_end = ntohl(*((unsigned int *)(data)));
+ event_end <<= 8;
+ event_end >>= 24;
+ samples = ntohl(*((unsigned int *)(data)));
+ samples &= 0xFFFF;
+
+ ast_verbose("Got RTP RFC2833 from %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u, mark %d, event %08x, end %d, duration %-5.5d) \n", ast_inet_ntoa(remote_address.sin_addr),
+ ntohs(remote_address.sin_port), payloadtype, seqno, timestamp, len, (mark?1:0), event, ((event_end & 0x80)?1:0), samples);
+
+ /* Print out debug if turned on */
+ if (rtpdebug || option_debug > 2)
+ ast_debug(0, "- RTP 2833 Event: %08x (len = %d)\n", event, len);
+
+ /* Figure out what digit was pressed */
+ if (event < 10) {
+ resp = '0' + event;
+ } else if (event < 11) {
+ resp = '*';
+ } else if (event < 12) {
+ resp = '#';
+ } else if (event < 16) {
+ resp = 'A' + (event - 12);
+ } else if (event < 17) { /* Event 16: Hook flash */
+ resp = 'X';
+ } else {
+ /* Not a supported event */
+ ast_log(LOG_DEBUG, "Ignoring RTP 2833 Event: %08x. Not a DTMF Digit.\n", event);
+ return &ast_null_frame;
+ }
+
+ if (ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_DTMF_COMPENSATE)) {
+ if ((rtp->lastevent != timestamp) || (rtp->resp && rtp->resp != resp)) {
+ rtp->resp = resp;
+ rtp->dtmfcount = 0;
+ f = send_dtmf(instance, AST_FRAME_DTMF_END, ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_DTMF_COMPENSATE));
+ f->len = 0;
+ rtp->lastevent = timestamp;
+ }
+ } else {
+ if ((!(rtp->resp) && (!(event_end & 0x80))) || (rtp->resp && rtp->resp != resp)) {
+ rtp->resp = resp;
+ f = send_dtmf(instance, AST_FRAME_DTMF_BEGIN, 0);
+ rtp->dtmfcount = dtmftimeout;
+ } else if ((event_end & 0x80) && (rtp->lastevent != seqno) && rtp->resp) {
+ f = send_dtmf(instance, AST_FRAME_DTMF_END, 0);
+ f->len = ast_tvdiff_ms(ast_samp2tv(samples, 8000), ast_tv(0, 0)); /* XXX hard coded 8kHz */
+ rtp->resp = 0;
+ rtp->dtmfcount = 0;
+ rtp->lastevent = seqno;
+ }
+ }
+
+ rtp->dtmfsamples = samples;
+
+ return f;
+}
+
+static struct ast_frame *process_dtmf_cisco(struct ast_rtp_instance *instance, unsigned char *data, int len, unsigned int seqno, unsigned int timestamp, struct sockaddr_in *sin, int payloadtype, int mark)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+ unsigned int event, flags, power;
+ char resp = 0;
+ unsigned char seq;
+ struct ast_frame *f = NULL;
+
+ if (len < 4) {
+ return NULL;
+ }
+
+ /* The format of Cisco RTP DTMF packet looks like next:
+ +0 - sequence number of DTMF RTP packet (begins from 1,
+ wrapped to 0)
+ +1 - set of flags
+ +1 (bit 0) - flaps by different DTMF digits delimited by audio
+ or repeated digit without audio???
+ +2 (+4,+6,...) - power level? (rises from 0 to 32 at begin of tone
+ then falls to 0 at its end)
+ +3 (+5,+7,...) - detected DTMF digit (0..9,*,#,A-D,...)
+ Repeated DTMF information (bytes 4/5, 6/7) is history shifted right
+ by each new packet and thus provides some redudancy.
+
+ Sample of Cisco RTP DTMF packet is (all data in hex):
+ 19 07 00 02 12 02 20 02
+ showing end of DTMF digit '2'.
+
+ The packets
+ 27 07 00 02 0A 02 20 02
+ 28 06 20 02 00 02 0A 02
+ shows begin of new digit '2' with very short pause (20 ms) after
+ previous digit '2'. Bit +1.0 flips at begin of new digit.
+
+ Cisco RTP DTMF packets comes as replacement of audio RTP packets
+ so its uses the same sequencing and timestamping rules as replaced
+ audio packets. Repeat interval of DTMF packets is 20 ms and not rely
+ on audio framing parameters. Marker bit isn't used within stream of
+ DTMFs nor audio stream coming immediately after DTMF stream. Timestamps
+ are not sequential at borders between DTMF and audio streams,
+ */
+
+ seq = data[0];
+ flags = data[1];
+ power = data[2];
+ event = data[3] & 0x1f;
+
+ if (option_debug > 2 || rtpdebug)
+ ast_debug(0, "Cisco DTMF Digit: %02x (len=%d, seq=%d, flags=%02x, power=%d, history count=%d)\n", event, len, seq, flags, power, (len - 4) / 2);
+ if (event < 10) {
+ resp = '0' + event;
+ } else if (event < 11) {
+ resp = '*';
+ } else if (event < 12) {
+ resp = '#';
+ } else if (event < 16) {
+ resp = 'A' + (event - 12);
+ } else if (event < 17) {
+ resp = 'X';
+ }
+ if ((!rtp->resp && power) || (rtp->resp && (rtp->resp != resp))) {
+ rtp->resp = resp;
+ /* Why we should care on DTMF compensation at reception? */
+ if (ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_DTMF_COMPENSATE)) {
+ f = send_dtmf(instance, AST_FRAME_DTMF_BEGIN, 0);
+ rtp->dtmfsamples = 0;
+ }
+ } else if ((rtp->resp == resp) && !power) {
+ f = send_dtmf(instance, AST_FRAME_DTMF_END, ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_DTMF_COMPENSATE));
+ f->samples = rtp->dtmfsamples * 8;
+ rtp->resp = 0;
+ } else if (rtp->resp == resp)
+ rtp->dtmfsamples += 20 * 8;
+ rtp->dtmfcount = dtmftimeout;
+
+ return f;
+}
+
+static struct ast_frame *process_cn_rfc3389(struct ast_rtp_instance *instance, unsigned char *data, int len, unsigned int seqno, unsigned int timestamp, struct sockaddr_in *sin, int payloadtype, int mark)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+
+ /* Convert comfort noise into audio with various codecs. Unfortunately this doesn't
+ totally help us out becuase we don't have an engine to keep it going and we are not
+ guaranteed to have it every 20ms or anything */
+ if (rtpdebug)
+ ast_debug(0, "- RTP 3389 Comfort noise event: Level %d (len = %d)\n", rtp->lastrxformat, len);
+
+ if (ast_test_flag(rtp, FLAG_3389_WARNING)) {
+ struct sockaddr_in remote_address;
+
+ ast_rtp_instance_get_remote_address(instance, &remote_address);
+
+ ast_log(LOG_NOTICE, "Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: %s\n",
+ ast_inet_ntoa(remote_address.sin_addr));
+ ast_set_flag(rtp, FLAG_3389_WARNING);
+ }
+
+ /* Must have at least one byte */
+ if (!len)
+ return NULL;
+ if (len < 24) {
+ rtp->f.data.ptr = rtp->rawdata + AST_FRIENDLY_OFFSET;
+ rtp->f.datalen = len - 1;
+ rtp->f.offset = AST_FRIENDLY_OFFSET;
+ memcpy(rtp->f.data.ptr, data + 1, len - 1);
+ } else {
+ rtp->f.data.ptr = NULL;
+ rtp->f.offset = 0;
+ rtp->f.datalen = 0;
+ }
+ rtp->f.frametype = AST_FRAME_CNG;
+ rtp->f.subclass = data[0] & 0x7f;
+ rtp->f.datalen = len - 1;
+ rtp->f.samples = 0;
+ rtp->f.delivery.tv_usec = rtp->f.delivery.tv_sec = 0;
+
+ return &rtp->f;
+}
+
+static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+ struct sockaddr_in sin;
+ socklen_t len = sizeof(sin);
+ unsigned int rtcpdata[8192 + AST_FRIENDLY_OFFSET];
+ unsigned int *rtcpheader = (unsigned int *)(rtcpdata + AST_FRIENDLY_OFFSET);
+ int res, packetwords, position = 0;
+ struct ast_frame *f = &ast_null_frame;
+
+ /* Read in RTCP data from the socket */
+ if ((res = recvfrom(rtp->rtcp->s, rtcpdata + AST_FRIENDLY_OFFSET, sizeof(rtcpdata) - sizeof(unsigned int) * AST_FRIENDLY_OFFSET, 0, (struct sockaddr *)&sin, &len)) < 0) {
+ ast_assert(errno != EBADF);
+ if (errno != EAGAIN) {
+ ast_log(LOG_WARNING, "RTCP Read error: %s. Hanging up.\n", strerror(errno));
+ return NULL;
+ }
+ return &ast_null_frame;
+ }
+
+ packetwords = res / 4;
+
+ if (ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT)) {
+ /* Send to whoever sent to us */
+ if ((rtp->rtcp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
+ (rtp->rtcp->them.sin_port != sin.sin_port)) {
+ memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them));
+ if (option_debug || rtpdebug)
+ ast_debug(0, "RTCP NAT: Got RTCP from other end. Now sending to address %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
+ }
+ }
+
+ ast_debug(1, "Got RTCP report of %d bytes\n", res);
+
+ while (position < packetwords) {
+ int i, pt, rc;
+ unsigned int length, dlsr, lsr, msw, lsw, comp;
+ struct timeval now;
+ double rttsec, reported_jitter, reported_normdev_jitter_current, normdevrtt_current, reported_lost, reported_normdev_lost_current;
+ uint64_t rtt = 0;
+
+ i = position;
+ length = ntohl(rtcpheader[i]);
+ pt = (length & 0xff0000) >> 16;
+ rc = (length & 0x1f000000) >> 24;
+ length &= 0xffff;
+
+ if ((i + length) > packetwords) {
+ if (option_debug || rtpdebug)
+ ast_log(LOG_DEBUG, "RTCP Read too short\n");
+ return &ast_null_frame;
+ }
+
+ if (rtcp_debug_test_addr(&sin)) {
+ ast_verbose("\n\nGot RTCP from %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
+ ast_verbose("PT: %d(%s)\n", pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown");
+ ast_verbose("Reception reports: %d\n", rc);
+ ast_verbose("SSRC of sender: %u\n", rtcpheader[i + 1]);
+ }
+
+ i += 2; /* Advance past header and ssrc */
+
+ switch (pt) {
+ case RTCP_PT_SR:
+ gettimeofday(&rtp->rtcp->rxlsr,NULL); /* To be able to populate the dlsr */
+ rtp->rtcp->spc = ntohl(rtcpheader[i+3]);
+ rtp->rtcp->soc = ntohl(rtcpheader[i + 4]);
+ rtp->rtcp->themrxlsr = ((ntohl(rtcpheader[i]) & 0x0000ffff) << 16) | ((ntohl(rtcpheader[i + 1]) & 0xffff0000) >> 16); /* Going to LSR in RR*/
+
+ if (rtcp_debug_test_addr(&sin)) {
+ ast_verbose("NTP timestamp: %lu.%010lu\n", (unsigned long) ntohl(rtcpheader[i]), (unsigned long) ntohl(rtcpheader[i + 1]) * 4096);
+ ast_verbose("RTP timestamp: %lu\n", (unsigned long) ntohl(rtcpheader[i + 2]));
+ ast_verbose("SPC: %lu\tSOC: %lu\n", (unsigned long) ntohl(rtcpheader[i + 3]), (unsigned long) ntohl(rtcpheader[i + 4]));
+ }
+ i += 5;
+ if (rc < 1)
+ break;
+ /* Intentional fall through */
+ case RTCP_PT_RR:
+ /* Don't handle multiple reception reports (rc > 1) yet */
+ /* Calculate RTT per RFC */
+ gettimeofday(&now, NULL);
+ timeval2ntp(now, &msw, &lsw);
+ if (ntohl(rtcpheader[i + 4]) && ntohl(rtcpheader[i + 5])) { /* We must have the LSR && DLSR */
+ comp = ((msw & 0xffff) << 16) | ((lsw & 0xffff0000) >> 16);
+ lsr = ntohl(rtcpheader[i + 4]);
+ dlsr = ntohl(rtcpheader[i + 5]);
+ rtt = comp - lsr - dlsr;
+
+ /* Convert end to end delay to usec (keeping the calculation in 64bit space)
+ sess->ee_delay = (eedelay * 1000) / 65536; */
+ if (rtt < 4294) {
+ rtt = (rtt * 1000000) >> 16;
+ } else {
+ rtt = (rtt * 1000) >> 16;
+ rtt *= 1000;
+ }
+ rtt = rtt / 1000.;
+ rttsec = rtt / 1000.;
+ rtp->rtcp->rtt = rttsec;
+
+ if (comp - dlsr >= lsr) {
+ rtp->rtcp->accumulated_transit += rttsec;
+
+ if (rtp->rtcp->rtt_count == 0)
+ rtp->rtcp->minrtt = rttsec;
+
+ if (rtp->rtcp->maxrtt<rttsec)
+ rtp->rtcp->maxrtt = rttsec;
+ if (rtp->rtcp->minrtt>rttsec)
+ rtp->rtcp->minrtt = rttsec;
+
+ normdevrtt_current = normdev_compute(rtp->rtcp->normdevrtt, rttsec, rtp->rtcp->rtt_count);
+
+ rtp->rtcp->stdevrtt = stddev_compute(rtp->rtcp->stdevrtt, rttsec, rtp->rtcp->normdevrtt, normdevrtt_current, rtp->rtcp->rtt_count);
+
+ rtp->rtcp->normdevrtt = normdevrtt_current;
+
+ rtp->rtcp->rtt_count++;
+ } else if (rtcp_debug_test_addr(&sin)) {
+ ast_verbose("Internal RTCP NTP clock skew detected: "
+ "lsr=%u, now=%u, dlsr=%u (%d:%03dms), "
+ "diff=%d\n",
+ lsr, comp, dlsr, dlsr / 65536,
+ (dlsr % 65536) * 1000 / 65536,
+ dlsr - (comp - lsr));
+ }
+ }
+
+ rtp->rtcp->reported_jitter = ntohl(rtcpheader[i + 3]);
+ reported_jitter = (double) rtp->rtcp->reported_jitter;
+
+ if (rtp->rtcp->reported_jitter_count == 0)
+ rtp->rtcp->reported_minjitter = reported_jitter;
+
+ if (reported_jitter < rtp->rtcp->reported_minjitter)
+ rtp->rtcp->reported_minjitter = reported_jitter;
+
+ if (reported_jitter > rtp->rtcp->reported_maxjitter)
+ rtp->rtcp->reported_maxjitter = reported_jitter;
+
+ reported_normdev_jitter_current = normdev_compute(rtp->rtcp->reported_normdev_jitter, reported_jitter, rtp->rtcp->reported_jitter_count);
+
+ rtp->rtcp->reported_stdev_jitter = stddev_compute(rtp->rtcp->reported_stdev_jitter, reported_jitter, rtp->rtcp->reported_normdev_jitter, reported_normdev_jitter_current, rtp->rtcp->reported_jitter_count);
+
+ rtp->rtcp->reported_normdev_jitter = reported_normdev_jitter_current;
+
+ rtp->rtcp->reported_lost = ntohl(rtcpheader[i + 1]) & 0xffffff;
+
+ reported_lost = (double) rtp->rtcp->reported_lost;
+
+ /* using same counter as for jitter */
+ if (rtp->rtcp->reported_jitter_count == 0)
+ rtp->rtcp->reported_minlost = reported_lost;
+
+ if (reported_lost < rtp->rtcp->reported_minlost)
+ rtp->rtcp->reported_minlost = reported_lost;
+
+ if (reported_lost > rtp->rtcp->reported_maxlost)
+ rtp->rtcp->reported_maxlost = reported_lost;
+ reported_normdev_lost_current = normdev_compute(rtp->rtcp->reported_normdev_lost, reported_lost, rtp->rtcp->reported_jitter_count);
+
+ rtp->rtcp->reported_stdev_lost = stddev_compute(rtp->rtcp->reported_stdev_lost, reported_lost, rtp->rtcp->reported_normdev_lost, reported_normdev_lost_current, rtp->rtcp->reported_jitter_count);
+
+ rtp->rtcp->reported_normdev_lost = reported_normdev_lost_current;
+
+ rtp->rtcp->reported_jitter_count++;
+
+ if (rtcp_debug_test_addr(&sin)) {
+ ast_verbose(" Fraction lost: %ld\n", (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24));
+ ast_verbose(" Packets lost so far: %d\n", rtp->rtcp->reported_lost);
+ ast_verbose(" Highest sequence number: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff));
+ ast_verbose(" Sequence number cycles: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16);
+ ast_verbose(" Interarrival jitter: %u\n", rtp->rtcp->reported_jitter);
+ ast_verbose(" Last SR(our NTP): %lu.%010lu\n",(unsigned long) ntohl(rtcpheader[i + 4]) >> 16,((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096);
+ ast_verbose(" DLSR: %4.4f (sec)\n",ntohl(rtcpheader[i + 5])/65536.0);
+ if (rtt)
+ ast_verbose(" RTT: %lu(sec)\n", (unsigned long) rtt);
+ }
+ if (rtt) {
+ manager_event(EVENT_FLAG_REPORTING, "RTCPReceived", "From %s:%d\r\n"
+ "PT: %d(%s)\r\n"
+ "ReceptionReports: %d\r\n"
+ "SenderSSRC: %u\r\n"
+ "FractionLost: %ld\r\n"
+ "PacketsLost: %d\r\n"
+ "HighestSequence: %ld\r\n"
+ "SequenceNumberCycles: %ld\r\n"
+ "IAJitter: %u\r\n"
+ "LastSR: %lu.%010lu\r\n"
+ "DLSR: %4.4f(sec)\r\n"
+ "RTT: %llu(sec)\r\n",
+ ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port),
+ pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown",
+ rc,
+ rtcpheader[i + 1],
+ (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24),
+ rtp->rtcp->reported_lost,
+ (long) (ntohl(rtcpheader[i + 2]) & 0xffff),
+ (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16,
+ rtp->rtcp->reported_jitter,
+ (unsigned long) ntohl(rtcpheader[i + 4]) >> 16, ((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096,
+ ntohl(rtcpheader[i + 5])/65536.0,
+ (unsigned long long)rtt);
+ } else {
+ manager_event(EVENT_FLAG_REPORTING, "RTCPReceived", "From %s:%d\r\n"
+ "PT: %d(%s)\r\n"
+ "ReceptionReports: %d\r\n"
+ "SenderSSRC: %u\r\n"
+ "FractionLost: %ld\r\n"
+ "PacketsLost: %d\r\n"
+ "HighestSequence: %ld\r\n"
+ "SequenceNumberCycles: %ld\r\n"
+ "IAJitter: %u\r\n"
+ "LastSR: %lu.%010lu\r\n"
+ "DLSR: %4.4f(sec)\r\n",
+ ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port),
+ pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown",
+ rc,
+ rtcpheader[i + 1],
+ (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24),
+ rtp->rtcp->reported_lost,
+ (long) (ntohl(rtcpheader[i + 2]) & 0xffff),
+ (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16,
+ rtp->rtcp->reported_jitter,
+ (unsigned long) ntohl(rtcpheader[i + 4]) >> 16,
+ ((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096,
+ ntohl(rtcpheader[i + 5])/65536.0);
+ }
+ break;
+ case RTCP_PT_FUR:
+ if (rtcp_debug_test_addr(&sin))
+ ast_verbose("Received an RTCP Fast Update Request\n");
+ rtp->f.frametype = AST_FRAME_CONTROL;
+ rtp->f.subclass = AST_CONTROL_VIDUPDATE;
+ rtp->f.datalen = 0;
+ rtp->f.samples = 0;
+ rtp->f.mallocd = 0;
+ rtp->f.src = "RTP";
+ f = &rtp->f;
+ break;
+ case RTCP_PT_SDES:
+ if (rtcp_debug_test_addr(&sin))
+ ast_verbose("Received an SDES from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
+ break;
+ case RTCP_PT_BYE:
+ if (rtcp_debug_test_addr(&sin))
+ ast_verbose("Received a BYE from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
+ break;
+ default:
+ ast_debug(1, "Unknown RTCP packet (pt=%d) received from %s:%d\n", pt, ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
+ break;
+ }
+ position += (length + 1);
+ }
+
+ rtp->rtcp->rtcp_info = 1;
+
+ return f;
+}
+
+static int bridge_p2p_rtp_write(struct ast_rtp_instance *instance, unsigned int *rtpheader, int len, int hdrlen)
+{
+ struct ast_rtp_instance *instance1 = ast_rtp_instance_get_bridged(instance);
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance), *bridged = ast_rtp_instance_get_data(instance1);
+ int res = 0, payload = 0, bridged_payload = 0, mark;
+ struct ast_rtp_payload_type payload_type;
+ int reconstruct = ntohl(rtpheader[0]);
+ struct sockaddr_in remote_address;
+
+ /* Get fields from packet */
+ payload = (reconstruct & 0x7f0000) >> 16;
+ mark = (((reconstruct & 0x800000) >> 23) != 0);
+
+ /* Check what the payload value should be */
+ payload_type = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(instance), payload);
+
+ /* Otherwise adjust bridged payload to match */
+ bridged_payload = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance1), payload_type.asterisk_format, payload_type.code);
+
+ /* If the payload coming in is not one of the negotiated ones then send it to the core, this will cause formats to change and the bridge to break */
+ if (!(ast_rtp_instance_get_codecs(instance1)->payloads[bridged_payload].code)) {
+ return -1;
+ }
+
+ /* If the marker bit has been explicitly set turn it on */
+ if (ast_test_flag(rtp, FLAG_NEED_MARKER_BIT)) {
+ mark = 1;
+ ast_clear_flag(rtp, FLAG_NEED_MARKER_BIT);
+ }
+
+ /* Reconstruct part of the packet */
+ reconstruct &= 0xFF80FFFF;
+ reconstruct |= (bridged_payload << 16);
+ reconstruct |= (mark << 23);
+ rtpheader[0] = htonl(reconstruct);
+
+ ast_rtp_instance_get_remote_address(instance1, &remote_address);
+
+ /* Send the packet back out */
+ res = sendto(bridged->s, (void *)rtpheader, len, 0, (struct sockaddr *)&remote_address, sizeof(remote_address));
+ if (res < 0) {
+ if (!ast_rtp_instance_get_prop(instance1, AST_RTP_PROPERTY_NAT) || (ast_rtp_instance_get_prop(instance1, AST_RTP_PROPERTY_NAT) && (ast_test_flag(bridged, FLAG_NAT_ACTIVE) == FLAG_NAT_ACTIVE))) {
+ ast_debug(1, "RTP Transmission error of packet to %s:%d: %s\n", ast_inet_ntoa(remote_address.sin_addr), ntohs(remote_address.sin_port), strerror(errno));
+ } else if (((ast_test_flag(bridged, FLAG_NAT_ACTIVE) == FLAG_NAT_INACTIVE) || rtpdebug) && !ast_test_flag(bridged, FLAG_NAT_INACTIVE_NOWARN)) {
+ if (option_debug || rtpdebug)
+ ast_debug(0, "RTP NAT: Can't write RTP to private address %s:%d, waiting for other end to send audio...\n", ast_inet_ntoa(remote_address.sin_addr), ntohs(remote_address.sin_port));
+ ast_set_flag(bridged, FLAG_NAT_INACTIVE_NOWARN);
+ }
+ return 0;
+ } else if (rtp_debug_test_addr(&remote_address)) {
+ ast_verbose("Sent RTP P2P packet to %s:%u (type %-2.2d, len %-6.6u)\n", ast_inet_ntoa(remote_address.sin_addr), ntohs(remote_address.sin_port), bridged_payload, len - hdrlen);
+ }
+
+ return 0;
+}
+
+static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtcp)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+ struct sockaddr_in sin;
+ socklen_t len = sizeof(sin);
+ int res, hdrlen = 12, version, payloadtype, padding, mark, ext, cc, prev_seqno;
+ unsigned int *rtpheader = (unsigned int*)(rtp->rawdata + AST_FRIENDLY_OFFSET), seqno, ssrc, timestamp;
+ struct ast_rtp_payload_type payload;
+ struct sockaddr_in remote_address;
+
+ /* If this is actually RTCP let's hop on over and handle it */
+ if (rtcp) {
+ if (rtp->rtcp) {
+ return ast_rtcp_read(instance);
+ }
+ return &ast_null_frame;
+ }
+
+ /* If we are currently sending DTMF to the remote party send a continuation packet */
+ if (rtp->sending_digit) {
+ ast_rtp_dtmf_continuation(instance);
+ }
+
+ /* Actually read in the data from the socket */
+ if ((res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET, 0, (struct sockaddr*)&sin, &len)) < 0) {
+ ast_assert(errno != EBADF);
+ if (errno != EAGAIN) {
+ ast_log(LOG_WARNING, "RTP Read error: %s. Hanging up.\n", strerror(errno));
+ return NULL;
+ }
+ return &ast_null_frame;
+ }
+
+ /* Make sure the data that was read in is actually enough to make up an RTP packet */
+ if (res < hdrlen) {
+ ast_log(LOG_WARNING, "RTP Read too short\n");
+ return &ast_null_frame;
+ }
+
+ /* If strict RTP protection is enabled see if we need to learn the remote address or if we need to drop the packet */
+ if (rtp->strict_rtp_state == STRICT_RTP_LEARN) {
+ memcpy(&rtp->strict_rtp_address, &sin, sizeof(rtp->strict_rtp_address));
+ rtp->strict_rtp_state = STRICT_RTP_CLOSED;
+ } else if (rtp->strict_rtp_state == STRICT_RTP_CLOSED) {
+ if ((rtp->strict_rtp_address.sin_addr.s_addr != sin.sin_addr.s_addr) || (rtp->strict_rtp_address.sin_port != sin.sin_port)) {
+ ast_debug(1, "Received RTP packet from %s:%d, dropping due to strict RTP protection. Expected it to be from %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), ast_inet_ntoa(rtp->strict_rtp_address.sin_addr), ntohs(rtp->strict_rtp_address.sin_port));
+ return &ast_null_frame;
+ }
+ }
+
+ /* Get fields and verify this is an RTP packet */
+ seqno = ntohl(rtpheader[0]);
+
+ ast_rtp_instance_get_remote_address(instance, &remote_address);
+
+ if (!(version = (seqno & 0xC0000000) >> 30)) {
+ if ((ast_stun_handle_packet(rtp->s, &sin, rtp->rawdata + AST_FRIENDLY_OFFSET, res, NULL, NULL) == AST_STUN_ACCEPT) &&
+ (!remote_address.sin_port && !remote_address.sin_addr.s_addr)) {
+ ast_rtp_instance_set_remote_address(instance, &sin);
+ }
+ return &ast_null_frame;
+ }
+
+ /* If symmetric RTP is enabled see if the remote side is not what we expected and change where we are sending audio */
+ if (ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT)) {
+ if ((remote_address.sin_addr.s_addr != sin.sin_addr.s_addr) ||
+ (remote_address.sin_port != sin.sin_port)) {
+ ast_rtp_instance_set_remote_address(instance, &sin);
+ memcpy(&remote_address, &sin, sizeof(remote_address));
+ if (rtp->rtcp) {
+ memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them));
+ rtp->rtcp->them.sin_port = htons(ntohs(sin.sin_port)+1);
+ }
+ rtp->rxseqno = 0;
+ ast_set_flag(rtp, FLAG_NAT_ACTIVE);
+ if (option_debug || rtpdebug)
+ ast_debug(0, "RTP NAT: Got audio from other end. Now sending to address %s:%d\n", ast_inet_ntoa(remote_address.sin_addr), ntohs(remote_address.sin_port));
+ }
+ }
+
+ /* If we are directly bridged to another instance send the audio directly out */
+ if (ast_rtp_instance_get_bridged(instance) && !bridge_p2p_rtp_write(instance, rtpheader, res, hdrlen)) {
+ return &ast_null_frame;
+ }
+
+ /* If the version is not what we expected by this point then just drop the packet */
+ if (version != 2) {
+ return &ast_null_frame;
+ }
+
+ /* Pull out the various other fields we will need */
+ payloadtype = (seqno & 0x7f0000) >> 16;
+ padding = seqno & (1 << 29);
+ mark = seqno & (1 << 23);
+ ext = seqno & (1 << 28);
+ cc = (seqno & 0xF000000) >> 24;
+ seqno &= 0xffff;
+ timestamp = ntohl(rtpheader[1]);
+ ssrc = ntohl(rtpheader[2]);
+
+ /* Force a marker bit if the SSRC changes */
+ if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) {
+ if (option_debug || rtpdebug) {
+ ast_debug(1, "Forcing Marker bit, because SSRC has changed\n");
+ }
+ mark = 1;
+ }
+
+ /* Remove any padding bytes that may be present */
+ if (padding) {
+ res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1];
+ }
+
+ /* Skip over any CSRC fields */
+ if (cc) {
+ hdrlen += cc * 4;
+ }
+
+ /* Look for any RTP extensions, currently we do not support any */
+ if (ext) {
+ hdrlen += (ntohl(rtpheader[hdrlen/4]) & 0xffff) << 2;
+ hdrlen += 4;
+ if (option_debug) {
+ int profile;
+ profile = (ntohl(rtpheader[3]) & 0xffff0000) >> 16;
+ if (profile == 0x505a)
+ ast_debug(1, "Found Zfone extension in RTP stream - zrtp - not supported.\n");
+ else
+ ast_debug(1, "Found unknown RTP Extensions %x\n", profile);
+ }
+ }
+
+ /* Make sure after we potentially mucked with the header length that it is once again valid */
+ if (res < hdrlen) {
+ ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d\n", res, hdrlen);
+ return &ast_null_frame;
+ }
+
+ rtp->rxcount++;
+ if (rtp->rxcount == 1) {
+ rtp->seedrxseqno = seqno;
+ }
+
+ /* Do not schedule RR if RTCP isn't run */
+ if (rtp->rtcp && rtp->rtcp->them.sin_addr.s_addr && rtp->rtcp->schedid < 1) {
+ /* Schedule transmission of Receiver Report */
+ rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp);
+ }
+ if ((int)rtp->lastrxseqno - (int)seqno > 100) /* if so it would indicate that the sender cycled; allow for misordering */
+ rtp->cycles += RTP_SEQ_MOD;
+
+ prev_seqno = rtp->lastrxseqno;
+ rtp->lastrxseqno = seqno;
+
+ if (!rtp->themssrc) {
+ rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */
+ }
+
+ if (rtp_debug_test_addr(&sin)) {
+ ast_verbose("Got RTP packet from %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
+ ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen);
+ }
+
+ payload = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(instance), payloadtype);
+
+ /* If the payload is not actually an Asterisk one but a special one pass it off to the respective handler */
+ if (!payload.asterisk_format) {
+ struct ast_frame *f = NULL;
+
+ if (payload.code == AST_RTP_DTMF) {
+ f = process_dtmf_rfc2833(instance, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp, &sin, payloadtype, mark);
+ } else if (payload.code == AST_RTP_CISCO_DTMF) {
+ f = process_dtmf_cisco(instance, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp, &sin, payloadtype, mark);
+ } else if (payload.code == AST_RTP_CN) {
+ f = process_cn_rfc3389(instance, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp, &sin, payloadtype, mark);
+ } else {
+ ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(remote_address.sin_addr));
+ }
+
+ return f ? f : &ast_null_frame;
+ }
+
+ rtp->lastrxformat = rtp->f.subclass = payload.code;
+ rtp->f.frametype = (rtp->f.subclass & AST_FORMAT_AUDIO_MASK) ? AST_FRAME_VOICE : (rtp->f.subclass & AST_FORMAT_VIDEO_MASK) ? AST_FRAME_VIDEO : AST_FRAME_TEXT;
+
+ rtp->rxseqno = seqno;
+ rtp->lastrxts = timestamp;
+
+ rtp->f.src = "RTP";
+ rtp->f.mallocd = 0;
+ rtp->f.datalen = res - hdrlen;
+ rtp->f.data.ptr = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET;
+ rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
+ rtp->f.seqno = seqno;
+
+ if (rtp->f.subclass == AST_FORMAT_T140 && (int)seqno - (prev_seqno+1) > 0 && (int)seqno - (prev_seqno+1) < 10) {
+ unsigned char *data = rtp->f.data.ptr;
+
+ memmove(rtp->f.data.ptr+3, rtp->f.data.ptr, rtp->f.datalen);
+ rtp->f.datalen +=3;
+ *data++ = 0xEF;
+ *data++ = 0xBF;
+ *data = 0xBD;
+ }
+
+ if (rtp->f.subclass == AST_FORMAT_T140RED) {
+ unsigned char *data = rtp->f.data.ptr;
+ unsigned char *header_end;
+ int num_generations;
+ int header_length;
+ int len;
+ int diff =(int)seqno - (prev_seqno+1); /* if diff = 0, no drop*/
+ int x;
+
+ rtp->f.subclass = AST_FORMAT_T140;
+ header_end = memchr(data, ((*data) & 0x7f), rtp->f.datalen);
+ header_end++;
+
+ header_length = header_end - data;
+ num_generations = header_length / 4;
+ len = header_length;
+
+ if (!diff) {
+ for (x = 0; x < num_generations; x++)
+ len += data[x * 4 + 3];
+
+ if (!(rtp->f.datalen - len))
+ return &ast_null_frame;
+
+ rtp->f.data.ptr += len;
+ rtp->f.datalen -= len;
+ } else if (diff > num_generations && diff < 10) {
+ len -= 3;
+ rtp->f.data.ptr += len;
+ rtp->f.datalen -= len;
+
+ data = rtp->f.data.ptr;
+ *data++ = 0xEF;
+ *data++ = 0xBF;
+ *data = 0xBD;
+ } else {
+ for ( x = 0; x < num_generations - diff; x++)
+ len += data[x * 4 + 3];
+
+ rtp->f.data.ptr += len;
+ rtp->f.datalen -= len;
+ }
+ }
+
+ if (rtp->f.subclass & AST_FORMAT_AUDIO_MASK) {
+ rtp->f.samples = ast_codec_get_samples(&rtp->f);
+ if (rtp->f.subclass == AST_FORMAT_SLINEAR)
+ ast_frame_byteswap_be(&rtp->f);
+ calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
+ /* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
+ ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);
+ rtp->f.ts = timestamp / 8;
+ rtp->f.len = rtp->f.samples / ((ast_format_rate(rtp->f.subclass) / 1000));
+ } else if (rtp->f.subclass & AST_FORMAT_VIDEO_MASK) {
+ /* Video -- samples is # of samples vs. 90000 */
+ if (!rtp->lastividtimestamp)
+ rtp->lastividtimestamp = timestamp;
+ rtp->f.samples = timestamp - rtp->lastividtimestamp;
+ rtp->lastividtimestamp = timestamp;
+ rtp->f.delivery.tv_sec = 0;
+ rtp->f.delivery.tv_usec = 0;
+ /* Pass the RTP marker bit as bit 0 in the subclass field.
+ * This is ok because subclass is actually a bitmask, and
+ * the low bits represent audio formats, that are not
+ * involved here since we deal with video.
+ */
+ if (mark)
+ rtp->f.subclass |= 0x1;
+ } else {
+ /* TEXT -- samples is # of samples vs. 1000 */
+ if (!rtp->lastitexttimestamp)
+ rtp->lastitexttimestamp = timestamp;
+ rtp->f.samples = timestamp - rtp->lastitexttimestamp;
+ rtp->lastitexttimestamp = timestamp;
+ rtp->f.delivery.tv_sec = 0;
+ rtp->f.delivery.tv_usec = 0;
+ }
+
+ return &rtp->f;
+}
+
+static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+
+ if (property == AST_RTP_PROPERTY_RTCP) {
+ if (rtp->rtcp) {
+ ast_debug(1, "Ignoring duplicate RTCP property on RTP instance '%p'\n", instance);
+ return;
+ }
+ if (!(rtp->rtcp = ast_calloc(1, sizeof(*rtp->rtcp)))) {
+ return;
+ }
+ if ((rtp->rtcp->s = create_new_socket("RTCP")) < 0) {
+ ast_debug(1, "Failed to create a new socket for RTCP on instance '%p'\n", instance);
+ ast_free(rtp->rtcp);
+ rtp->rtcp = NULL;
+ return;
+ }
+
+ /* Grab the IP address and port we are going to use */
+ ast_rtp_instance_get_local_address(instance, &rtp->rtcp->us);
+ rtp->rtcp->us.sin_port = htons(ntohs(rtp->rtcp->us.sin_port) + 1);
+
+ /* Try to actually bind to the IP address and port we are going to use for RTCP, if this fails we have to bail out */
+ if (bind(rtp->rtcp->s, (struct sockaddr*)&rtp->rtcp->us, sizeof(rtp->rtcp->us))) {
+ ast_debug(1, "Failed to setup RTCP on RTP instance '%p'\n", instance);
+ close(rtp->rtcp->s);
+ ast_free(rtp->rtcp);
+ rtp->rtcp = NULL;
+ return;
+ }
+
+ ast_debug(1, "Setup RTCP on RTP instance '%p'\n", instance);
+ rtp->rtcp->schedid = -1;
+
+ return;
+ }
+
+ return;
+}
+
+static int ast_rtp_fd(struct ast_rtp_instance *instance, int rtcp)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+
+ return rtcp ? (rtp->rtcp ? rtp->rtcp->s : -1) : rtp->s;
+}
+
+static void ast_rtp_remote_address_set(struct ast_rtp_instance *instance, struct sockaddr_in *sin)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+
+ if (rtp->rtcp) {
+ ast_debug(1, "Setting RTCP address on RTP instance '%p'\n", instance);
+ memcpy(&rtp->rtcp->them, sin, sizeof(rtp->rtcp->them));
+ rtp->rtcp->them.sin_port = htons(ntohs(sin->sin_port) + 1);
+ }
+
+ rtp->rxseqno = 0;
+
+ if (strictrtp) {
+ rtp->strict_rtp_state = STRICT_RTP_LEARN;
+ }
+
+ return;
+}
+
+/*! \brief Write t140 redundacy frame
+ * \param data primary data to be buffered
+ */
+static int red_write(const void *data)
+{
+ struct ast_rtp_instance *instance = (struct ast_rtp_instance*) data;
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+
+ ast_rtp_write(instance, &rtp->red->t140);
+
+ return 1;
+}
+
+static int rtp_red_init(struct ast_rtp_instance *instance, int buffer_time, int *payloads, int generations)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+ int x;
+
+ if (!(rtp->red = ast_calloc(1, sizeof(*rtp->red)))) {
+ return -1;
+ }
+
+ rtp->red->t140.frametype = AST_FRAME_TEXT;
+ rtp->red->t140.subclass = AST_FORMAT_T140RED;
+ rtp->red->t140.data.ptr = &rtp->red->buf_data;
+
+ rtp->red->t140.ts = 0;
+ rtp->red->t140red = rtp->red->t140;
+ rtp->red->t140red.data.ptr = &rtp->red->t140red_data;
+ rtp->red->t140red.datalen = 0;
+ rtp->red->ti = buffer_time;
+ rtp->red->num_gen = generations;
+ rtp->red->hdrlen = generations * 4 + 1;
+ rtp->red->prev_ts = 0;
+
+ for (x = 0; x < generations; x++) {
+ rtp->red->pt[x] = payloads[x];
+ rtp->red->pt[x] |= 1 << 7; /* mark redundant generations pt */
+ rtp->red->t140red_data[x*4] = rtp->red->pt[x];
+ }
+ rtp->red->t140red_data[x*4] = rtp->red->pt[x] = payloads[x]; /* primary pt */
+ rtp->red->schedid = ast_sched_add(rtp->sched, generations, red_write, instance);
+
+ rtp->red->t140.datalen = 0;
+
+ return 0;
+}
+
+static int rtp_red_buffer(struct ast_rtp_instance *instance, struct ast_frame *frame)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+
+ if (frame->datalen > -1) {
+ struct rtp_red *red = rtp->red;
+ memcpy(&red->buf_data[red->t140.datalen], frame->data.ptr, frame->datalen);
+ red->t140.datalen += frame->datalen;
+ red->t140.ts = frame->ts;
+ }
+
+ return 0;
+}
+
+static int ast_rtp_local_bridge(struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance0);
+
+ ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
+
+ return 0;
+}
+
+static int ast_rtp_get_stat(struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+
+ if (!rtp->rtcp) {
+ return -1;
+ }
+
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_TXCOUNT, -1, stats->txcount, rtp->txcount);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_RXCOUNT, -1, stats->rxcount, rtp->rxcount);
+
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_TXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->txploss, rtp->rtcp->reported_lost);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_RXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->rxploss, rtp->rtcp->expected_prior - rtp->rtcp->received_prior);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_MAXRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->remote_maxrxploss, rtp->rtcp->reported_maxlost);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_MINRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->remote_minrxploss, rtp->rtcp->reported_minlost);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->remote_normdevrxploss, rtp->rtcp->reported_normdev_lost);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_STDEVRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->remote_stdevrxploss, rtp->rtcp->reported_stdev_lost);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_MAXRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->local_maxrxploss, rtp->rtcp->maxrxlost);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_MINRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->local_minrxploss, rtp->rtcp->minrxlost);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->local_normdevrxploss, rtp->rtcp->normdev_rxlost);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_STDEVRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->local_stdevrxploss, rtp->rtcp->stdev_rxlost);
+ AST_RTP_STAT_TERMINATOR(AST_RTP_INSTANCE_STAT_COMBINED_LOSS);
+
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_TXJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->txjitter, rtp->rxjitter);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_RXJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->rxjitter, rtp->rtcp->reported_jitter / (unsigned int) 65536.0);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_MAXJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->remote_maxjitter, rtp->rtcp->reported_maxjitter);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_MINJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->remote_minjitter, rtp->rtcp->reported_minjitter);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->remote_normdevjitter, rtp->rtcp->reported_normdev_jitter);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_STDEVJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->remote_stdevjitter, rtp->rtcp->reported_stdev_jitter);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_MAXJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->local_maxjitter, rtp->rtcp->maxrxjitter);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_MINJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->local_minjitter, rtp->rtcp->minrxjitter);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->local_normdevjitter, rtp->rtcp->normdev_rxjitter);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_STDEVJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->local_stdevjitter, rtp->rtcp->stdev_rxjitter);
+ AST_RTP_STAT_TERMINATOR(AST_RTP_INSTANCE_STAT_COMBINED_JITTER);
+
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_RTT, AST_RTP_INSTANCE_STAT_COMBINED_RTT, stats->rtt, rtp->rtcp->rtt);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_MAX_RTT, AST_RTP_INSTANCE_STAT_COMBINED_RTT, stats->maxrtt, rtp->rtcp->maxrtt);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_MIN_RTT, AST_RTP_INSTANCE_STAT_COMBINED_RTT, stats->minrtt, rtp->rtcp->minrtt);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_NORMDEVRTT, AST_RTP_INSTANCE_STAT_COMBINED_RTT, stats->normdevrtt, rtp->rtcp->normdevrtt);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_STDEVRTT, AST_RTP_INSTANCE_STAT_COMBINED_RTT, stats->stdevrtt, rtp->rtcp->stdevrtt);
+ AST_RTP_STAT_TERMINATOR(AST_RTP_INSTANCE_STAT_COMBINED_RTT);
+
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_SSRC, -1, stats->local_ssrc, rtp->ssrc);
+ AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_SSRC, -1, stats->remote_ssrc, rtp->themssrc);
+
+ return 0;
+}
+
+static int ast_rtp_dtmf_compatible(struct ast_channel *chan0, struct ast_rtp_instance *instance0, struct ast_channel *chan1, struct ast_rtp_instance *instance1)
+{
+ /* If both sides are not using the same method of DTMF transmission
+ * (ie: one is RFC2833, other is INFO... then we can not do direct media.
+ * --------------------------------------------------
+ * | DTMF Mode | HAS_DTMF | Accepts Begin Frames |
+ * |-----------|------------|-----------------------|
+ * | Inband | False | True |
+ * | RFC2833 | True | True |
+ * | SIP INFO | False | False |
+ * --------------------------------------------------
+ */
+ return (((ast_rtp_instance_get_prop(instance0, AST_RTP_PROPERTY_DTMF) != ast_rtp_instance_get_prop(instance1, AST_RTP_PROPERTY_DTMF)) ||
+ (!chan0->tech->send_digit_begin != !chan1->tech->send_digit_begin)) ? 0 : 1);
+}
+
+static void ast_rtp_stun_request(struct ast_rtp_instance *instance, struct sockaddr_in *suggestion, const char *username)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+
+ ast_stun_request(rtp->s, suggestion, username, NULL);
+}
+
+static void ast_rtp_stop(struct ast_rtp_instance *instance)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+ struct sockaddr_in sin = { 0, };
+
+ if (rtp->rtcp) {
+ AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
+ }
+ if (rtp->red) {
+ AST_SCHED_DEL(rtp->sched, rtp->red->schedid);
+ free(rtp->red);
+ rtp->red = NULL;
+ }
+
+ ast_rtp_instance_set_remote_address(instance, &sin);
+ if (rtp->rtcp) {
+ memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->rtcp->them.sin_addr));
+ memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->rtcp->them.sin_port));
+ }
+
+ ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
+}
+
+static char *rtp_do_debug_ip(struct ast_cli_args *a)
+{
+ struct hostent *hp;
+ struct ast_hostent ahp;
+ int port = 0;
+ char *p, *arg;
+
+ arg = a->argv[3];
+ p = strstr(arg, ":");
+ if (p) {
+ *p = '\0';
+ p++;
+ port = atoi(p);
+ }
+ hp = ast_gethostbyname(arg, &ahp);
+ if (hp == NULL) {
+ ast_cli(a->fd, "Lookup failed for '%s'\n", arg);
+ return CLI_FAILURE;
+ }
+ rtpdebugaddr.sin_family = AF_INET;
+ memcpy(&rtpdebugaddr.sin_addr, hp->h_addr, sizeof(rtpdebugaddr.sin_addr));
+ rtpdebugaddr.sin_port = htons(port);
+ if (port == 0)
+ ast_cli(a->fd, "RTP Debugging Enabled for IP: %s\n", ast_inet_ntoa(rtpdebugaddr.sin_addr));
+ else
+ ast_cli(a->fd, "RTP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(rtpdebugaddr.sin_addr), port);
+ rtpdebug = 1;
+ return CLI_SUCCESS;
+}
+
+static char *rtcp_do_debug_ip(struct ast_cli_args *a)
+{
+ struct hostent *hp;
+ struct ast_hostent ahp;
+ int port = 0;
+ char *p, *arg;
+
+ arg = a->argv[3];
+ p = strstr(arg, ":");
+ if (p) {
+ *p = '\0';
+ p++;
+ port = atoi(p);
+ }
+ hp = ast_gethostbyname(arg, &ahp);
+ if (hp == NULL) {
+ ast_cli(a->fd, "Lookup failed for '%s'\n", arg);
+ return CLI_FAILURE;
+ }
+ rtcpdebugaddr.sin_family = AF_INET;
+ memcpy(&rtcpdebugaddr.sin_addr, hp->h_addr, sizeof(rtcpdebugaddr.sin_addr));
+ rtcpdebugaddr.sin_port = htons(port);
+ if (port == 0)
+ ast_cli(a->fd, "RTCP Debugging Enabled for IP: %s\n", ast_inet_ntoa(rtcpdebugaddr.sin_addr));
+ else
+ ast_cli(a->fd, "RTCP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(rtcpdebugaddr.sin_addr), port);
+ rtcpdebug = 1;
+ return CLI_SUCCESS;
+}
+
+static char *handle_cli_rtp_set_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
+{
+ switch (cmd) {
+ case CLI_INIT:
+ e->command = "rtp set debug {on|off|ip}";
+ e->usage =
+ "Usage: rtp set debug {on|off|ip host[:port]}\n"
+ " Enable/Disable dumping of all RTP packets. If 'ip' is\n"
+ " specified, limit the dumped packets to those to and from\n"
+ " the specified 'host' with optional port.\n";
+ return NULL;
+ case CLI_GENERATE:
+ return NULL;
+ }
+
+ if (a->argc == e->args) { /* set on or off */
+ if (!strncasecmp(a->argv[e->args-1], "on", 2)) {
+ rtpdebug = 1;
+ memset(&rtpdebugaddr, 0, sizeof(rtpdebugaddr));
+ ast_cli(a->fd, "RTP Debugging Enabled\n");
+ return CLI_SUCCESS;
+ } else if (!strncasecmp(a->argv[e->args-1], "off", 3)) {
+ rtpdebug = 0;
+ ast_cli(a->fd, "RTP Debugging Disabled\n");
+ return CLI_SUCCESS;
+ }
+ } else if (a->argc == e->args +1) { /* ip */
+ return rtp_do_debug_ip(a);
+ }
+
+ return CLI_SHOWUSAGE; /* default, failure */
+}
+
+static char *handle_cli_rtcp_set_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
+{
+ switch (cmd) {
+ case CLI_INIT:
+ e->command = "rtcp set debug {on|off|ip}";
+ e->usage =
+ "Usage: rtcp set debug {on|off|ip host[:port]}\n"
+ " Enable/Disable dumping of all RTCP packets. If 'ip' is\n"
+ " specified, limit the dumped packets to those to and from\n"
+ " the specified 'host' with optional port.\n";
+ return NULL;
+ case CLI_GENERATE:
+ return NULL;
+ }
+
+ if (a->argc == e->args) { /* set on or off */
+ if (!strncasecmp(a->argv[e->args-1], "on", 2)) {
+ rtcpdebug = 1;
+ memset(&rtcpdebugaddr, 0, sizeof(rtcpdebugaddr));
+ ast_cli(a->fd, "RTCP Debugging Enabled\n");
+ return CLI_SUCCESS;
+ } else if (!strncasecmp(a->argv[e->args-1], "off", 3)) {
+ rtcpdebug = 0;
+ ast_cli(a->fd, "RTCP Debugging Disabled\n");
+ return CLI_SUCCESS;
+ }
+ } else if (a->argc == e->args +1) { /* ip */
+ return rtcp_do_debug_ip(a);
+ }
+
+ return CLI_SHOWUSAGE; /* default, failure */
+}
+
+static char *handle_cli_rtcp_set_stats(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
+{
+ switch (cmd) {
+ case CLI_INIT:
+ e->command = "rtcp set stats {on|off}";
+ e->usage =
+ "Usage: rtcp set stats {on|off}\n"
+ " Enable/Disable dumping of RTCP stats.\n";
+ return NULL;
+ case CLI_GENERATE:
+ return NULL;
+ }
+
+ if (a->argc != e->args)
+ return CLI_SHOWUSAGE;
+
+ if (!strncasecmp(a->argv[e->args-1], "on", 2))
+ rtcpstats = 1;
+ else if (!strncasecmp(a->argv[e->args-1], "off", 3))
+ rtcpstats = 0;
+ else
+ return CLI_SHOWUSAGE;
+
+ ast_cli(a->fd, "RTCP Stats %s\n", rtcpstats ? "Enabled" : "Disabled");
+ return CLI_SUCCESS;
+}
+
+static struct ast_cli_entry cli_rtp[] = {
+ AST_CLI_DEFINE(handle_cli_rtp_set_debug, "Enable/Disable RTP debugging"),
+ AST_CLI_DEFINE(handle_cli_rtcp_set_debug, "Enable/Disable RTCP debugging"),
+ AST_CLI_DEFINE(handle_cli_rtcp_set_stats, "Enable/Disable RTCP stats"),
+};
+
+static int rtp_reload(int reload)
+{
+ struct ast_config *cfg;
+ const char *s;
+ struct ast_flags config_flags = { reload ? CONFIG_FLAG_FILEUNCHANGED : 0 };
+
+ cfg = ast_config_load2("rtp.conf", "rtp", config_flags);
+ if (cfg == CONFIG_STATUS_FILEMISSING || cfg == CONFIG_STATUS_FILEUNCHANGED || cfg == CONFIG_STATUS_FILEINVALID) {
+ return 0;
+ }
+
+ rtpstart = DEFAULT_RTP_START;
+ rtpend = DEFAULT_RTP_END;
+ dtmftimeout = DEFAULT_DTMF_TIMEOUT;
+ strictrtp = STRICT_RTP_OPEN;
+ if (cfg) {
+ if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) {
+ rtpstart = atoi(s);
+ if (rtpstart < MINIMUM_RTP_PORT)
+ rtpstart = MINIMUM_RTP_PORT;
+ if (rtpstart > MAXIMUM_RTP_PORT)
+ rtpstart = MAXIMUM_RTP_PORT;
+ }
+ if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) {
+ rtpend = atoi(s);
+ if (rtpend < MINIMUM_RTP_PORT)
+ rtpend = MINIMUM_RTP_PORT;
+ if (rtpend > MAXIMUM_RTP_PORT)
+ rtpend = MAXIMUM_RTP_PORT;
+ }
+ if ((s = ast_variable_retrieve(cfg, "general", "rtcpinterval"))) {
+ rtcpinterval = atoi(s);
+ if (rtcpinterval == 0)
+ rtcpinterval = 0; /* Just so we're clear... it's zero */
+ if (rtcpinterval < RTCP_MIN_INTERVALMS)
+ rtcpinterval = RTCP_MIN_INTERVALMS; /* This catches negative numbers too */
+ if (rtcpinterval > RTCP_MAX_INTERVALMS)
+ rtcpinterval = RTCP_MAX_INTERVALMS;
+ }
+ if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) {
+#ifdef SO_NO_CHECK
+ nochecksums = ast_false(s) ? 1 : 0;
+#else
+ if (ast_false(s))
+ ast_log(LOG_WARNING, "Disabling RTP checksums is not supported on this operating system!\n");
+#endif
+ }
+ if ((s = ast_variable_retrieve(cfg, "general", "dtmftimeout"))) {
+ dtmftimeout = atoi(s);
+ if ((dtmftimeout < 0) || (dtmftimeout > 20000)) {
+ ast_log(LOG_WARNING, "DTMF timeout of '%d' outside range, using default of '%d' instead\n",
+ dtmftimeout, DEFAULT_DTMF_TIMEOUT);
+ dtmftimeout = DEFAULT_DTMF_TIMEOUT;
+ };
+ }
+ if ((s = ast_variable_retrieve(cfg, "general", "strictrtp"))) {
+ strictrtp = ast_true(s);
+ }
+ ast_config_destroy(cfg);
+ }
+ if (rtpstart >= rtpend) {
+ ast_log(LOG_WARNING, "Unreasonable values for RTP start/end port in rtp.conf\n");
+ rtpstart = DEFAULT_RTP_START;
+ rtpend = DEFAULT_RTP_END;
+ }
+ ast_verb(2, "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend);
+ return 0;
+}
+
+static int reload_module(void)
+{
+ rtp_reload(1);
+ return 0;
+}
+
+static int load_module(void)
+{
+ if (ast_rtp_engine_register(&asterisk_rtp_engine)) {
+ return AST_MODULE_LOAD_DECLINE;
+ }
+
+ if (ast_cli_register_multiple(cli_rtp, ARRAY_LEN(cli_rtp))) {
+ ast_rtp_engine_unregister(&asterisk_rtp_engine);
+ return AST_MODULE_LOAD_DECLINE;
+ }
+
+ rtp_reload(0);
+
+ return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+ ast_rtp_engine_unregister(&asterisk_rtp_engine);
+ ast_cli_unregister_multiple(cli_rtp, ARRAY_LEN(cli_rtp));
+
+ return 0;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Asterisk RTP Stack",
+ .load = load_module,
+ .unload = unload_module,
+ .reload = reload_module,
+ );
--
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