From d3418aa14a83615a488508df8dcf60e86c0504c9 Mon Sep 17 00:00:00 2001
From: Mark Spencer <markster@digium.com>
Date: Fri, 7 Jan 2005 07:11:40 +0000
Subject: [PATCH] Support CNG transmission when on hold (bug #2904)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4704 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
channels/chan_sip.c | 32 ++++++++++++++++++++++++++++++--
include/asterisk/rtp.h | 2 ++
rtp.c | 40 ++++++++++++++++++++++++++++++++++++++++
3 files changed, 72 insertions(+), 2 deletions(-)
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index fa0f63214f..e2a3c77783 100755
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -151,6 +151,8 @@ static int global_rtptimeout = 0;
static int global_rtpholdtimeout = 0;
+static int global_rtpkeepalive = 0;
+
static int global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT;
/* Object counters */
@@ -379,8 +381,10 @@ static struct sip_pvt {
int initid; /* Auto-congest ID if appropriate */
int autokillid; /* Auto-kill ID */
time_t lastrtprx; /* Last RTP received */
+ time_t lastrtptx; /* Last RTP sent */
int rtptimeout; /* RTP timeout time */
int rtpholdtimeout; /* RTP timeout when on hold */
+ int rtpkeepalive; /* Send RTP packets for keepalive */
int subscribed; /* Is this call a subscription? */
int stateid;
@@ -468,6 +472,7 @@ struct sip_peer {
int capability; /* Codec capability */
int rtptimeout;
int rtpholdtimeout;
+ int rtpkeepalive; /* Send RTP packets for keepalive */
unsigned int callgroup; /* Call group */
unsigned int pickupgroup; /* Pickup group */
struct sockaddr_in addr; /* IP address of peer */
@@ -1821,6 +1826,7 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, 0);
ast_set_flag(p, SIP_PROGRESS_SENT);
}
+ time(&p->lastrtptx);
res = ast_rtp_write(p->rtp, frame);
}
ast_mutex_unlock(&p->lock);
@@ -1833,6 +1839,7 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, 0);
ast_set_flag(p, SIP_PROGRESS_SENT);
}
+ time(&p->lastrtptx);
res = ast_rtp_write(p->vrtp, frame);
}
ast_mutex_unlock(&p->lock);
@@ -2346,6 +2353,7 @@ static struct sip_pvt *sip_alloc(char *callid, struct sockaddr_in *sin, int useg
strncpy(p->musicclass, global_musicclass, sizeof(p->musicclass) - 1);
p->rtptimeout = global_rtptimeout;
p->rtpholdtimeout = global_rtpholdtimeout;
+ p->rtpkeepalive = global_rtpkeepalive;
p->capability = global_capability;
if (ast_test_flag(p, SIP_DTMF) == SIP_DTMF_RFC2833)
p->noncodeccapability |= AST_RTP_DTMF;
@@ -2637,6 +2645,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
/* Update our last rtprx when we receive an SDP, too */
time(&p->lastrtprx);
+ time(&p->lastrtptx);
/* Get codec and RTP info from SDP */
if (strcasecmp(get_header(req, "Content-Type"), "application/sdp")) {
@@ -3550,6 +3559,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
}
/* Update lastrtprx when we send our SDP */
time(&p->lastrtprx);
+ time(&p->lastrtptx);
return 0;
}
@@ -8192,8 +8202,13 @@ restartsearch:
sip = iflist;
while(sip) {
ast_mutex_lock(&sip->lock);
- if (sip->rtp && sip->owner && (sip->owner->_state == AST_STATE_UP) && sip->lastrtprx && (sip->rtptimeout || sip->rtpholdtimeout) && !sip->redirip.sin_addr.s_addr) {
- if (t > sip->lastrtprx + sip->rtptimeout) {
+ if (sip->rtp && sip->owner && (sip->owner->_state == AST_STATE_UP) && !sip->redirip.sin_addr.s_addr) {
+ if (sip->lastrtptx && sip->rtpkeepalive && t > sip->lastrtptx + sip->rtpkeepalive) {
+ /* Need to send an empty RTP packet */
+ time(&sip->lastrtptx);
+ ast_rtp_sendcng(sip->rtp, 0);
+ }
+ if (sip->lastrtprx && (sip->rtptimeout || sip->rtpholdtimeout) && t > sip->lastrtprx + sip->rtptimeout) {
/* Might be a timeout now -- see if we're on hold */
struct sockaddr_in sin;
ast_rtp_get_peer(sip->rtp, &sin);
@@ -8718,6 +8733,7 @@ static struct sip_peer *temp_peer(char *name)
peer->capability = global_capability;
peer->rtptimeout = global_rtptimeout;
peer->rtpholdtimeout = global_rtpholdtimeout;
+ peer->rtpkeepalive = global_rtpkeepalive;
ast_set_flag(peer, SIP_SELFDESTRUCT);
ast_set_flag(peer, SIP_DYNAMIC);
peer->prefs = prefs;
@@ -8883,6 +8899,11 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, int
ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno);
peer->rtpholdtimeout = global_rtpholdtimeout;
}
+ } else if (!strcasecmp(v->name, "rtpkeepalive")) {
+ if ((sscanf(v->value, "%d", &peer->rtpkeepalive) != 1) || (peer->rtpkeepalive < 0)) {
+ ast_log(LOG_WARNING, "'%s' is not a valid RTP keepalive time at line %d. Using default.\n", v->value, v->lineno);
+ peer->rtpkeepalive = global_rtpkeepalive;
+ }
} else if (!strcasecmp(v->name, "qualify")) {
if (!strcasecmp(v->value, "no")) {
peer->maxms = 0;
@@ -8968,6 +8989,7 @@ static int reload_config(void)
ourport = DEFAULT_SIP_PORT;
global_rtptimeout = 0;
global_rtpholdtimeout = 0;
+ global_rtpkeepalive = 0;
pedanticsipchecking = 0;
ast_clear_flag(&global_flags, AST_FLAGS_ALL);
ast_set_flag(&global_flags, SIP_DTMF_RFC2833);
@@ -9018,6 +9040,11 @@ static int reload_config(void)
ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno);
global_rtpholdtimeout = 0;
}
+ } else if (!strcasecmp(v->name, "rtpkeepalive")) {
+ if ((sscanf(v->value, "%d", &global_rtpkeepalive) != 1) || (global_rtpkeepalive < 0)) {
+ ast_log(LOG_WARNING, "'%s' is not a valid RTP keepalive time at line %d. Using default.\n", v->value, v->lineno);
+ global_rtpkeepalive = 0;
+ }
} else if (!strcasecmp(v->name, "videosupport")) {
videosupport = ast_true(v->value);
} else if (!strcasecmp(v->name, "compactheaders")) {
@@ -9276,6 +9303,7 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
}
/* Reset lastrtprx timer */
time(&p->lastrtprx);
+ time(&p->lastrtptx);
ast_mutex_unlock(&p->lock);
return 0;
}
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index 6b2df7c02f..0fc653e59b 100755
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -78,6 +78,8 @@ int ast_rtcp_fd(struct ast_rtp *rtp);
int ast_rtp_senddigit(struct ast_rtp *rtp, char digit);
+int ast_rtp_sendcng(struct ast_rtp *rtp, int level);
+
int ast_rtp_settos(struct ast_rtp *rtp, int tos);
// Setting RTP payload types from lines in a SDP description:
diff --git a/rtp.c b/rtp.c
index acf67d081e..c213dcd592 100755
--- a/rtp.c
+++ b/rtp.c
@@ -1129,6 +1129,46 @@ int ast_rtp_senddigit(struct ast_rtp *rtp, char digit)
return 0;
}
+int ast_rtp_sendcng(struct ast_rtp *rtp, int level)
+{
+ unsigned int *rtpheader;
+ int hdrlen = 12;
+ int res;
+ int payload;
+ char data[256];
+ char iabuf[INET_ADDRSTRLEN];
+ 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;
+
+ gettimeofday(&rtp->dtmfmute, NULL);
+ rtp->dtmfmute.tv_usec += (500 * 1000);
+ if (rtp->dtmfmute.tv_usec > 1000000) {
+ rtp->dtmfmute.tv_usec -= 1000000;
+ rtp->dtmfmute.tv_sec += 1;
+ }
+
+ /* 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(iabuf, sizeof(iabuf), 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:%d (type %d, seq %d, ts %d, len %d)\n"
+ , ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastts,res - hdrlen);
+
+ }
+ return 0;
+}
+
#ifdef SOLARIS
static void put_uint32(unsigned char *buf, int i)
{
--
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