[asterisk-commits] twilson: branch 1.6.0 r221301 - in /branches/1.6.0: ./ channels/ configs/ inc...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Sep 30 13:50:54 CDT 2009
Author: twilson
Date: Wed Sep 30 13:50:50 2009
New Revision: 221301
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=221301
Log:
Merged revisions 221266 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009) | 32 lines
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/
........
................
Modified:
branches/1.6.0/ (props changed)
branches/1.6.0/channels/chan_sip.c
branches/1.6.0/configs/sip.conf.sample
branches/1.6.0/include/asterisk/rtp.h
branches/1.6.0/main/rtp.c
Propchange: branches/1.6.0/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.
Modified: branches/1.6.0/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=221301&r1=221300&r2=221301
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Wed Sep 30 13:50:50 2009
@@ -1003,11 +1003,13 @@
#define SIP_PAGE2_DIALOG_ESTABLISHED (1 << 27) /*!< 29: Has a dialog been established? */
#define SIP_PAGE2_REGISTERTRYING (1 << 29) /*!< DP: Send 100 Trying on REGISTER attempts */
#define SIP_PAGE2_UDPTL_DESTINATION (1 << 30) /*!< DP: Use source IP of RTP as destination if NAT is enabled */
+#define SIP_PAGE2_CONSTANT_SSRC (1 << 31) /*!< GDP: Don't change SSRC on reinvite */
#define SIP_PAGE2_FLAGS_TO_COPY \
(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_IGNORESDPVERSION | \
SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | \
- SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_UDPTL_DESTINATION)
+ SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_UDPTL_DESTINATION | \
+ SIP_PAGE2_CONSTANT_SSRC)
/*@}*/
@@ -4300,6 +4302,9 @@
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;
@@ -4310,6 +4315,9 @@
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);
+ }
}
if (dialog->trtp) { /* Realtime text */
ast_rtp_setdtmf(dialog->trtp, 0);
@@ -17648,6 +17656,7 @@
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");
@@ -17695,6 +17704,14 @@
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
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_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;
@@ -20833,6 +20850,9 @@
} 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;
@@ -22365,6 +22385,8 @@
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);
} else if (!strcasecmp(v->name, "session-timers")) {
int i = (int) str2stmode(v->value);
if (i < 0) {
Modified: branches/1.6.0/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.0/configs/sip.conf.sample?view=diff&rev=221301&r1=221300&r2=221301
==============================================================================
--- branches/1.6.0/configs/sip.conf.sample (original)
+++ branches/1.6.0/configs/sip.conf.sample Wed Sep 30 13:50:50 2009
@@ -587,6 +587,8 @@
; for devices that send us non standard SDP packets
; (observed with Microsoft OCS). By default this option is
; off.
+
+;constantssrc=yes ; Don't change the RTP SSRC when our media stream changes
;----------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
Modified: branches/1.6.0/include/asterisk/rtp.h
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.0/include/asterisk/rtp.h?view=diff&rev=221301&r1=221300&r2=221301
==============================================================================
--- branches/1.6.0/include/asterisk/rtp.h (original)
+++ branches/1.6.0/include/asterisk/rtp.h Wed Sep 30 13:50:50 2009
@@ -187,6 +187,9 @@
int ast_rtp_setqos(struct ast_rtp *rtp, int tos, int cos, char *desc);
+/*! \brief When changing sources, don't generate a new SSRC */
+void ast_rtp_set_constantssrc(struct ast_rtp *rtp);
+
void ast_rtp_new_source(struct ast_rtp *rtp);
/*! \brief Setting RTP payload types from lines in a SDP description: */
Modified: branches/1.6.0/main/rtp.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.0/main/rtp.c?view=diff&rev=221301&r1=221300&r2=221301
==============================================================================
--- branches/1.6.0/main/rtp.c (original)
+++ branches/1.6.0/main/rtp.c Wed Sep 30 13:50:50 2009
@@ -179,6 +179,7 @@
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 */
+ unsigned int constantssrc:1;
};
/* Forward declarations */
@@ -2389,12 +2390,19 @@
return ast_netsock_set_qos(rtp->s, tos, cos, desc);
}
+void ast_rtp_set_constantssrc(struct ast_rtp *rtp)
+{
+ rtp->constantssrc = 1;
+}
+
void ast_rtp_new_source(struct ast_rtp *rtp)
{
if (rtp) {
rtp->set_marker_bit = 1;
- }
- return;
+ if (!rtp->constantssrc) {
+ rtp->ssrc = ast_random();
+ }
+ }
}
void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
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