From 865daf4858ba8f3a592e08d37f8025d92c02810b Mon Sep 17 00:00:00 2001
From: Terry Wilson <twilson@digium.com>
Date: Wed, 30 Sep 2009 17:52:30 +0000
Subject: [PATCH] Merged revisions 221086 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines
Change the SSRC by default when our media stream changes
Be default, change SSRC when doing an audio stream changes Asterisk doesn't
honor marker bit when reinvited to already-bridged RTP streams,resulting in
far-end stack discarding packets with "old" timestamps that areactually part of
a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a
reinvite, unless the 'constantssrc' is set to true in sip.conf.
The original issue reported to Digium support detailed the following situation:
ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in
fromITSP, Asterisk dials the app server which sends a re-invite back
toAsterisk--not to negotiate to send media directly to the ITSP, but to
indicatethat it's changing the stream it's sending to Asterisk. The app
servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker
bit on the new stream. Asterisk passes through the teimstamp of the new stream,
butdoes not reset the SSRC, sequence numbers, or set the marker bit.
When the timestamp on the new stream is older than the timestamp on the
originalstream, the ITSP (which doesn't know there has been any change) discards
the newframes because it thinks they are too old. This patch addresses this by
changing the SSRC on a stream update unless constantssrc=true is set in
sip.conf.
Review: https://reviewboard.asterisk.org/r/374/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
channels/chan_sip.c | 23 ++++++++++++++++++++++-
configs/sip.conf.sample | 3 +++
include/asterisk/rtp_engine.h | 19 +++++++++++++++++++
main/rtp_engine.c | 7 +++++++
res/res_rtp_asterisk.c | 14 ++++++++++++++
5 files changed, 65 insertions(+), 1 deletion(-)
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b8e53d9157..9c00765544 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1503,6 +1503,7 @@ struct sip_auth {
#define SIP_PAGE2_RTAUTOCLEAR (1 << 2) /*!< GP: Should we clean memory from peers after expiry? */
#define SIP_PAGE2_RPID_UPDATE (1 << 3)
/* Space for addition of other realtime flags in the future */
+#define SIP_PAGE2_CONSTANT_SSRC (1 << 7) /*!< GDP: Don't change SSRC on reinvite */
#define SIP_PAGE2_SYMMETRICRTP (1 << 8) /*!< GDP: Whether symmetric RTP is enabled or not */
#define SIP_PAGE2_STATECHANGEQUEUE (1 << 9) /*!< D: Unsent state pending change exists */
@@ -1540,7 +1541,7 @@ struct sip_auth {
SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | \
SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_FAX_DETECT | \
SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS | SIP_PAGE2_PREFERRED_CODEC | \
- SIP_PAGE2_RPID_IMMEDIATE | SIP_PAGE2_RPID_UPDATE | SIP_PAGE2_SYMMETRICRTP)
+ SIP_PAGE2_RPID_IMMEDIATE | SIP_PAGE2_RPID_UPDATE | SIP_PAGE2_SYMMETRICRTP | SIP_PAGE2_CONSTANT_SSRC)
/*@}*/
@@ -5192,6 +5193,9 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
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);
+ if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
+ ast_rtp_instance_set_constantssrc(dialog->rtp);
+ }
/* Set Frame packetization */
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(dialog->rtp), dialog->rtp, &dialog->prefs);
dialog->autoframing = peer->autoframing;
@@ -5199,6 +5203,9 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
if (dialog->vrtp) { /* Video */
ast_rtp_instance_set_timeout(dialog->vrtp, peer->rtptimeout);
ast_rtp_instance_set_hold_timeout(dialog->vrtp, peer->rtpholdtimeout);
+ if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
+ ast_rtp_instance_set_constantssrc(dialog->vrtp);
+ }
}
if (dialog->trtp) { /* Realtime text */
ast_rtp_instance_set_timeout(dialog->trtp, peer->rtptimeout);
@@ -20437,6 +20444,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return -1;
}
+ ast_queue_control(p->owner, AST_CONTROL_SRCUPDATE);
} else {
p->jointcapability = p->capability;
ast_debug(1, "Hm.... No sdp for the moment\n");
@@ -20487,6 +20495,14 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
ast_debug(1, "No compatible codecs for this SIP call.\n");
return -1;
}
+ if (ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
+ if (p->rtp) {
+ ast_rtp_instance_set_constantssrc(p->rtp);
+ }
+ if (p->vrtp) {
+ ast_rtp_instance_set_constantssrc(p->vrtp);
+ }
+ }
} else { /* No SDP in invite, call control session */
p->jointcapability = p->capability;
ast_debug(2, "No SDP in Invite, third party call control\n");
@@ -23854,6 +23870,9 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
} else if (!strcasecmp(v->name, "t38pt_usertpsource")) {
ast_set_flag(&mask[1], SIP_PAGE2_UDPTL_DESTINATION);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_UDPTL_DESTINATION);
+ } else if (!strcasecmp(v->name, "constantssrc")) {
+ ast_set_flag(&mask[1], SIP_PAGE2_CONSTANT_SSRC);
+ ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_CONSTANT_SSRC);
} else
res = 0;
@@ -25357,6 +25376,8 @@ static int reload_config(enum channelreloadreason reason)
} else if (!strcasecmp(v->name, "disallowed_methods")) {
char *disallow = ast_strdupa(v->value);
mark_parsed_methods(&sip_cfg.disallowed_methods, disallow);
+ } else if (!strcasecmp(v->name, "constantssrc")) {
+ ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_CONSTANT_SSRC);
}
}
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index ba95e6355c..bdd356c299 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -730,6 +730,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
; This field MUST NOT contain spaces
+;constantssrc=yes ; Don't change the RTP SSRC when our media stream changes
+
;----------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
; please read realtime.txt and extconfig.txt in the /doc directory of the
@@ -935,6 +937,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; timerb
; qualifyfreq
; t38pt_usertpsource
+; constantssrc
; contactpermit ; Limit what a host may register as (a neat trick
; contactdeny ; is to register at the same IP as a SIP provider,
; ; then call oneself, and get redirected to that
diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h
index 5d5ae3f7b1..29070d0c7e 100644
--- a/include/asterisk/rtp_engine.h
+++ b/include/asterisk/rtp_engine.h
@@ -317,6 +317,8 @@ struct ast_rtp_engine {
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 to tell new_source not to change SSRC */
+ void (*constant_ssrc_set)(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 */
@@ -1182,6 +1184,23 @@ int ast_rtp_instance_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_r
*/
enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *instance);
+/*!
+ * \brief Mark an RTP instance not to update SSRC on a new source
+ *
+ * \param instance Instance to update
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_set_constantssrc(instance);
+ * \endcode
+ *
+ * This sets the indicated instance to not update the RTP SSRC when new_source
+ * is called.
+ *
+ * \since 1.6.3
+ */
+void ast_rtp_instance_set_constantssrc(struct ast_rtp_instance *instance);
/*!
* \brief Indicate a new source of audio has dropped in
*
diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index cf6d2c6f2d..53ed892b26 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -726,6 +726,13 @@ enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *i
return instance->dtmf_mode;
}
+void ast_rtp_instance_set_constantssrc(struct ast_rtp_instance *instance)
+{
+ if (instance->engine->constant_ssrc_set) {
+ instance->engine->constant_ssrc_set(instance);
+ }
+}
+
void ast_rtp_instance_new_source(struct ast_rtp_instance *instance)
{
if (instance->engine->new_source) {
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 3abb6c686a..42cce37868 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -103,6 +103,7 @@ enum strict_rtp_state {
#define FLAG_NAT_INACTIVE_NOWARN (1 << 1)
#define FLAG_NEED_MARKER_BIT (1 << 3)
#define FLAG_DTMF_COMPENSATE (1 << 4)
+#define FLAG_CONSTANT_SSRC (1 << 5)
/*! \brief RTP session description */
struct ast_rtp {
@@ -253,6 +254,7 @@ 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 void ast_rtp_set_constantssrc(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);
@@ -275,6 +277,7 @@ static struct ast_rtp_engine asterisk_rtp_engine = {
.dtmf_begin = ast_rtp_dtmf_begin,
.dtmf_end = ast_rtp_dtmf_end,
.new_source = ast_rtp_new_source,
+ .constant_ssrc_set = ast_rtp_set_constantssrc,
.write = ast_rtp_write,
.read = ast_rtp_read,
.prop_set = ast_rtp_prop_set,
@@ -653,6 +656,13 @@ static int ast_rtp_dtmf_end(struct ast_rtp_instance *instance, char digit)
return 0;
}
+void ast_rtp_set_constantssrc(struct ast_rtp_instance *instance)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+
+ ast_set_flag(rtp, FLAG_CONSTANT_SSRC);
+}
+
static void ast_rtp_new_source(struct ast_rtp_instance *instance)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
@@ -660,6 +670,10 @@ static void ast_rtp_new_source(struct ast_rtp_instance *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);
+ if (!ast_test_flag(rtp, FLAG_CONSTANT_SSRC)) {
+ rtp->ssrc = ast_random();
+ }
+
return;
}
--
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