Merged revisions 252089 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

........
  r252089 | twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines
  
  Only change the RTP ssrc when we see that it has changed
  
  This change basically reverts the change reviewed in
  https://reviewboard.asterisk.org/r/374/ and instead limits the
  updating of the RTP synchronization source to only those times when we
  detect that the other side of the conversation has changed the ssrc.
  
  The problem is that SRCUPDATE control frames are sent many times where
  we don't want a new ssrc, including whenever Asterisk has to send DTMF
  in a normal bridge. This is also not the first time that this mistake
  has been made. The initial implementation of the ast_rtp_new_source
  function also changed the ssrc--and then it was removed because of
  this same issue. Then, we put it back in again to fix a different
  issue. This patch attempts to only change the ssrc when we see that
  the other side of the conversation has changed the ssrc.
  
  It also renames some functions to make their purpose more clear.
  
  Review: https://reviewboard.asterisk.org/r/540/
........


git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@252134 f38db490-d61c-443f-a65b-d21fe96a405b

4 files changed, 22 insertions, 29 deletions

diff --git a/channels/chan_h323.c b/channels/chan_h323.c
index 697b90d30..640836731 100644
--- a/channels/chan_h323.c
+++ b/channels/chan_h323.c
@@ -919,7 +919,11 @@ static int oh323_indicate(struct ast_channel *c, int condition, const void *data
 		res = 0;
 		break;
 	case AST_CONTROL_SRCUPDATE:
-		ast_rtp_new_source(pvt->rtp);
+		ast_rtp_update_source(pvt->rtp);
+		res = 0;
+		break;
+	case AST_CONTROL_SRCCHANGE:
+		ast_rtp_change_source(pvt->rtp);
 		res = 0;
 		break;
 	case AST_CONTROL_PROCEEDING:
diff --git a/channels/chan_mgcp.c b/channels/chan_mgcp.c
index 07cf590bb..1e02db100 100644
--- a/channels/chan_mgcp.c
+++ b/channels/chan_mgcp.c
@@ -1477,7 +1477,10 @@ static int mgcp_indicate(struct ast_channel *ast, int ind, const void *data, siz
 		ast_moh_stop(ast);
 		break;
 	case AST_CONTROL_SRCUPDATE:
-		ast_rtp_new_source(sub->rtp);
+		ast_rtp_update_source(sub->rtp);
+		break;
+	case AST_CONTROL_SRCCHANGE:
+		ast_rtp_change_source(sub->rtp);
 		break;
 	case -1:
 		transmit_notify_request(sub, "");
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 9e797e7d6..e861f127b 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1008,13 +1008,12 @@ struct sip_auth {
 #define SIP_PAGE2_FAX_DETECT		(1 << 28)		/*!< DP: Fax Detection support */
 #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_CONSTANT_SSRC | SIP_PAGE2_FAX_DETECT)
+	 SIP_PAGE2_FAX_DETECT)
 
 /*@}*/ 
 
@@ -4333,9 +4332,6 @@ 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;
@@ -4346,9 +4342,6 @@ 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);
@@ -5322,7 +5315,7 @@ static int sip_answer(struct ast_channel *ast)
 
 		ast_setstate(ast, AST_STATE_UP);
 		ast_debug(1, "SIP answering channel: %s\n", ast->name);
-		ast_rtp_new_source(p->rtp);
+		ast_rtp_update_source(p->rtp);
 		ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
 		res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE);
 	}
@@ -5357,7 +5350,7 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
 				if ((ast->_state != AST_STATE_UP) &&
 				    !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
 				    !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
-					ast_rtp_new_source(p->rtp);
+					ast_rtp_update_source(p->rtp);
 					if (!global_prematuremediafilter) {
 						p->invitestate = INV_EARLY_MEDIA;
 						transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
@@ -5677,11 +5670,11 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
 		res = -1;
 		break;
 	case AST_CONTROL_HOLD:
-		ast_rtp_new_source(p->rtp);
+		ast_rtp_update_source(p->rtp);
 		ast_moh_start(ast, data, p->mohinterpret);
 		break;
 	case AST_CONTROL_UNHOLD:
-		ast_rtp_new_source(p->rtp);
+		ast_rtp_update_source(p->rtp);
 		ast_moh_stop(ast);
 		break;
 	case AST_CONTROL_VIDUPDATE:	/* Request a video frame update */
@@ -5700,7 +5693,10 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
 		}
 		break;
 	case AST_CONTROL_SRCUPDATE:
-		ast_rtp_new_source(p->rtp);
+		ast_rtp_update_source(p->rtp);
+		break;
+	case AST_CONTROL_SRCCHANGE:
+		ast_rtp_change_source(p->rtp);
 		break;
 	case -1:
 		res = -1;
@@ -18168,14 +18164,6 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
 				res = -1;
 				goto request_invite_cleanup;
 			}
-			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");
@@ -21367,9 +21355,6 @@ 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 if (!strcasecmp(v->name, "faxdetect")) {
                 ast_set_flag(&mask[1], SIP_PAGE2_FAX_DETECT);
                 ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_FAX_DETECT);
@@ -22878,8 +22863,6 @@ 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) {
diff --git a/channels/chan_skinny.c b/channels/chan_skinny.c
index a04a56493..12533fbfc 100644
--- a/channels/chan_skinny.c
+++ b/channels/chan_skinny.c
@@ -3769,7 +3769,10 @@ static int skinny_indicate(struct ast_channel *ast, int ind, const void *data, s
 	case AST_CONTROL_PROCEEDING:
 		break;
 	case AST_CONTROL_SRCUPDATE:
-		ast_rtp_new_source(sub->rtp);
+		ast_rtp_update_source(sub->rtp);
+		break;
+	case AST_CONTROL_SRCCHANGE:
+		ast_rtp_change_source(sub->rtp);
 		break;
 	default:
 		ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", ind);