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authortwilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b>2009-09-30 14:49:11 +0000
committertwilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b>2009-09-30 14:49:11 +0000
commit4f67fa12535deb67bc00f833d168270544281077 (patch)
treef65c97c3a71a2daada0530012670d648d3c34478 /channels
parentec161909b6471d89ba0301ba2a67cac094eee718 (diff)
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: http://svn.digium.com/svn/asterisk/branches/1.4@221086 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
-rw-r--r--channels/chan_sip.c24
1 files changed, 23 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 15984fb0e..0e8c24a01 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -807,10 +807,12 @@ struct sip_auth {
#define SIP_PAGE2_UDPTL_DESTINATION (1 << 28) /*!< 28: Use source IP of RTP as destination if NAT is enabled */
#define SIP_PAGE2_DIALOG_ESTABLISHED (1 << 29) /*!< 29: Has a dialog been established? */
#define SIP_PAGE2_RPORT_PRESENT (1 << 30) /*!< 30: Was rport received in the Via header? */
+#define SIP_PAGE2_CONSTANT_SSRC (1 << 31) /*!< 31: Don't change SSRC on reinvite */
#define SIP_PAGE2_FLAGS_TO_COPY \
(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | \
- SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_UDPTL_DESTINATION)
+ SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | SIP_PAGE2_BUGGY_MWI | \
+ SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_CONSTANT_SSRC)
/* SIP packet flags */
#define SIP_PKT_DEBUG (1 << 0) /*!< Debug this packet */
@@ -2939,6 +2941,9 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
ast_rtp_set_rtptimeout(dialog->rtp, peer->rtptimeout);
ast_rtp_set_rtpholdtimeout(dialog->rtp, peer->rtpholdtimeout);
ast_rtp_set_rtpkeepalive(dialog->rtp, peer->rtpkeepalive);
+ if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
+ ast_rtp_set_constantssrc(dialog->rtp);
+ }
/* Set Frame packetization */
ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
dialog->autoframing = peer->autoframing;
@@ -2949,6 +2954,9 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
ast_rtp_set_rtptimeout(dialog->vrtp, peer->rtptimeout);
ast_rtp_set_rtpholdtimeout(dialog->vrtp, peer->rtpholdtimeout);
ast_rtp_set_rtpkeepalive(dialog->vrtp, peer->rtpkeepalive);
+ if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
+ ast_rtp_set_constantssrc(dialog->vrtp);
+ }
}
ast_string_field_set(dialog, peername, peer->name);
@@ -14854,6 +14862,7 @@ 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;
if (option_debug > 2)
@@ -14908,6 +14917,14 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
ast_log(LOG_DEBUG, "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_set_constantssrc(p->rtp);
+ }
+ if (p->vrtp) {
+ ast_rtp_set_constantssrc(p->vrtp);
+ }
+ }
} else { /* No SDP in invite, call control session */
p->jointcapability = p->capability;
if (option_debug > 1)
@@ -17366,6 +17383,9 @@ 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;
@@ -18422,6 +18442,8 @@ static int reload_config(enum channelreloadreason reason)
default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE;
} else if (!strcasecmp(v->name, "matchexterniplocally")) {
global_matchexterniplocally = ast_true(v->value);
+ } else if (!strcasecmp(v->name, "constantssrc")) {
+ ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_CONSTANT_SSRC);
}
}