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/
  ........
................


git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221301 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 23 insertions, 1 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index f14dc16fa..23fee3496 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1003,11 +1003,13 @@ struct sip_auth {
 #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 @@ 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;
@@ -4310,6 +4315,9 @@ 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);
+		}
 	}
 	if (dialog->trtp) { /* Realtime text */
 		ast_rtp_setdtmf(dialog->trtp, 0);
@@ -17648,6 +17656,7 @@ 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");
@@ -17696,6 +17705,14 @@ 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_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;
 			ast_debug(2, "No SDP in Invite, third party call control\n");
@@ -20833,6 +20850,9 @@ 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;
 
@@ -22365,6 +22385,8 @@ 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);
 		} else if (!strcasecmp(v->name, "session-timers")) {
 			int i = (int) str2stmode(v->value); 
 			if (i < 0) {