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

2 files changed, 43 insertions, 14 deletions

diff --git a/main/channel.c b/main/channel.c
index 1ac9f3d5d..c80ad1cf0 100644
--- a/main/channel.c
+++ b/main/channel.c
@@ -2461,6 +2461,7 @@ int ast_waitfordigit_full(struct ast_channel *c, int ms, int audiofd, int cmdfd)
 				case AST_CONTROL_RINGING:
 				case AST_CONTROL_ANSWER:
 				case AST_CONTROL_SRCUPDATE:
+				case AST_CONTROL_SRCCHANGE:
 					/* Unimportant */
 					break;
 				default:
@@ -3102,6 +3103,7 @@ static int attribute_const is_visible_indication(enum ast_control_frame_type con
 	case AST_CONTROL_PROCEEDING:
 	case AST_CONTROL_VIDUPDATE:
 	case AST_CONTROL_SRCUPDATE:
+	case AST_CONTROL_SRCCHANGE:
 	case AST_CONTROL_RADIO_KEY:
 	case AST_CONTROL_RADIO_UNKEY:
 	case AST_CONTROL_OPTION:
@@ -3207,6 +3209,7 @@ int ast_indicate_data(struct ast_channel *chan, int _condition,
 	case AST_CONTROL_PROCEEDING:
 	case AST_CONTROL_VIDUPDATE:
 	case AST_CONTROL_SRCUPDATE:
+	case AST_CONTROL_SRCCHANGE:
 	case AST_CONTROL_RADIO_KEY:
 	case AST_CONTROL_RADIO_UNKEY:
 	case AST_CONTROL_OPTION:
@@ -3911,6 +3914,7 @@ struct ast_channel *__ast_request_and_dial(const char *type, int format, void *d
 				case AST_CONTROL_UNHOLD:
 				case AST_CONTROL_VIDUPDATE:
 				case AST_CONTROL_SRCUPDATE:
+				case AST_CONTROL_SRCCHANGE:
 				case -1:			/* Ignore -- just stopping indications */
 					break;
 
@@ -4865,6 +4869,7 @@ static enum ast_bridge_result ast_generic_bridge(struct ast_channel *c0, struct
 			case AST_CONTROL_UNHOLD:
 			case AST_CONTROL_VIDUPDATE:
 			case AST_CONTROL_SRCUPDATE:
+			case AST_CONTROL_SRCCHANGE:
 			case AST_CONTROL_T38_PARAMETERS:
 				ast_indicate_data(other, f->subclass, f->data, f->datalen);
 				if (jb_in_use) {
diff --git a/main/rtp.c b/main/rtp.c
index 393a9b137..5342a3e6e 100644
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -1434,6 +1434,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
 	unsigned int *rtpheader;
 	struct rtpPayloadType rtpPT;
 	struct ast_rtp *bridged = NULL;
+	AST_LIST_HEAD_NOLOCK(, ast_frame) frames;
 	
 	/* If time is up, kill it */
 	if (rtp->sending_digit)
@@ -1533,10 +1534,22 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
 	timestamp = ntohl(rtpheader[1]);
 	ssrc = ntohl(rtpheader[2]);
 	
-	if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) {
-		if (option_debug || rtpdebug)
-			ast_debug(0, "Forcing Marker bit, because SSRC has changed\n");
-		mark = 1;
+	AST_LIST_HEAD_INIT_NOLOCK(&frames);
+	/* Force a marker bit and change SSRC if the SSRC changes */
+	if (rtp->rxssrc && rtp->rxssrc != ssrc) {
+		struct ast_frame *f, srcupdate = {
+			AST_FRAME_CONTROL,
+			.subclass = AST_CONTROL_SRCCHANGE,
+		};
+
+		if (!mark) {
+			if (option_debug || rtpdebug) {
+				ast_debug(0, "Forcing Marker bit, because SSRC has changed\n");
+			}
+			mark = 1;
+		}
+		f = ast_frisolate(&srcupdate);
+		AST_LIST_INSERT_TAIL(&frames, f, frame_list);
 	}
 
 	rtp->rxssrc = ssrc;
@@ -1567,7 +1580,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
 
 	if (res < hdrlen) {
 		ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen);
-		return &ast_null_frame;
+		return AST_LIST_FIRST(&frames) ? AST_LIST_FIRST(&frames) : &ast_null_frame;
 	}
 
 	rtp->rxcount++; /* Only count reasonably valid packets, this'll make the rtcp stats more accurate */
@@ -1629,7 +1642,11 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
 		} else {
 			ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(rtp->them.sin_addr));
 		}
-		return f ? f : &ast_null_frame;
+		if (f) {
+			AST_LIST_INSERT_TAIL(&frames, f, frame_list);
+			return AST_LIST_FIRST(&frames);
+		}
+		return &ast_null_frame;
 	}
 	rtp->lastrxformat = rtp->f.subclass = rtpPT.code;
 	rtp->f.frametype = (rtp->f.subclass & AST_FORMAT_AUDIO_MASK) ? AST_FRAME_VOICE : (rtp->f.subclass & AST_FORMAT_VIDEO_MASK) ? AST_FRAME_VIDEO : AST_FRAME_TEXT;
@@ -1645,7 +1662,8 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
 			f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass)), ast_tv(0, 0));
 			rtp->resp = 0;
 			rtp->dtmf_timeout = rtp->dtmf_duration = 0;
-			return f;
+			AST_LIST_INSERT_TAIL(&frames, f, frame_list);
+			return AST_LIST_FIRST(&frames);
 		}
 	}
 
@@ -1691,7 +1709,9 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
 		rtp->f.delivery.tv_usec = 0;
 	}
 	rtp->f.src = "RTP";
-	return &rtp->f;
+
+	AST_LIST_INSERT_TAIL(&frames, &rtp->f, frame_list);
+	return AST_LIST_FIRST(&frames);
 }
 
 /* The following array defines the MIME Media type (and subtype) for each
@@ -2382,18 +2402,22 @@ int ast_rtp_setqos(struct ast_rtp *rtp, int tos, int cos, char *desc)
 	return ast_netsock_set_qos(rtp->s, tos, cos, desc);
 }
 
-void ast_rtp_set_constantssrc(struct ast_rtp *rtp)
+void ast_rtp_update_source(struct ast_rtp *rtp)
 {
-	rtp->constantssrc = 1;
+	if (rtp) {
+		rtp->set_marker_bit = 1;
+		ast_debug(3, "Setting the marker bit due to a source update\n");
+	}
 }
 
-void ast_rtp_new_source(struct ast_rtp *rtp)
+void ast_rtp_change_source(struct ast_rtp *rtp)
 {
 	if (rtp) {
+		unsigned int ssrc = ast_random();
+
 		rtp->set_marker_bit = 1;
-		if (!rtp->constantssrc) {
-			rtp->ssrc = ast_random();
-		}
+		ast_debug(3, "Changing ssrc from %u to %u due to a source change\n", rtp->ssrc, ssrc);
+		rtp->ssrc = ssrc;
 	}
 }