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/1.4@252175 f38db490-d61c-443f-a65b-d21fe96a405b

4 files changed, 22 insertions, 28 deletions

diff --git a/channels/chan_h323.c b/channels/chan_h323.c
index 700fdd174..f8a929d6e 100644
--- a/channels/chan_h323.c
+++ b/channels/chan_h323.c
@@ -918,7 +918,11 @@ 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 159657b85..3f8b49e0d 100644
--- a/channels/chan_mgcp.c
+++ b/channels/chan_mgcp.c
@@ -1442,7 +1442,10 @@ 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 618fdd7ca..6d80d466f 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -811,12 +811,11 @@ 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_CONSTANT_SSRC)
+	SIP_PAGE2_UDPTL_DESTINATION)
 
 /* SIP packet flags */
 #define SIP_PKT_DEBUG		(1 << 0)	/*!< Debug this packet */
@@ -2952,9 +2951,6 @@ 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;
@@ -2965,9 +2961,6 @@ 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);
-		}
 	}
 
 	ast_string_field_set(dialog, peername, peer->name);
@@ -3871,6 +3864,7 @@ static int sip_answer(struct ast_channel *ast)
 		if (option_debug)
 			ast_log(LOG_DEBUG, "SIP answering channel: %s\n", ast->name);
 
+		ast_rtp_update_source(p->rtp);
 		res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
 		ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
 	}
@@ -3905,7 +3899,7 @@ 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, 1);
@@ -4153,11 +4147,11 @@ 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 */
@@ -4168,7 +4162,10 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
 			res = -1;
 		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;
@@ -15085,14 +15082,6 @@ 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;
 			if (option_debug > 1)
@@ -17574,9 +17563,6 @@ 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
 		res = 0;
 
@@ -18650,8 +18636,6 @@ 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, "shrinkcallerid")) {
 			if (ast_true(v->value)) {
 				global_shrinkcallerid = 1;
diff --git a/channels/chan_skinny.c b/channels/chan_skinny.c
index 812aa16fb..48be77b9c 100644
--- a/channels/chan_skinny.c
+++ b/channels/chan_skinny.c
@@ -2868,7 +2868,10 @@ 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);