When a new source of audio comes in (such as music on hold) make sure the marker bit gets set.

(closes issue #10355)
Reported by: wdecarne
Patches:
      10355.diff uploaded by file (license 11)
(closes issue #11491)
Reported by: kanderson


git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@105674 f38db490-d61c-443f-a65b-d21fe96a405b

3 files changed, 22 insertions, 1 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5873fe07b..994a4c964 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -3661,8 +3661,10 @@ static int sip_answer(struct ast_channel *ast)
 			if (option_debug > 1)
 				ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", p->t38.state, ast->name);
 			res = transmit_response_with_t38_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
-		} else 
+		} else {
+			ast_rtp_new_source(p->rtp);
 			res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
+		}
 	}
 	ast_mutex_unlock(&p->lock);
 	return res;
@@ -3695,6 +3697,7 @@ 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);
 					transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
 					ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);	
 				}
@@ -3920,9 +3923,11 @@ 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_moh_start(ast, data, p->mohinterpret);
 		break;
 	case AST_CONTROL_UNHOLD:
+		ast_rtp_new_source(p->rtp);
 		ast_moh_stop(ast);
 		break;
 	case AST_CONTROL_VIDUPDATE:	/* Request a video frame update */
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index 1ffa04d0d..870529e5d 100644
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -162,6 +162,8 @@ int ast_rtp_sendcng(struct ast_rtp *rtp, int level);
 
 int ast_rtp_settos(struct ast_rtp *rtp, int tos);
 
+void ast_rtp_new_source(struct ast_rtp *rtp);
+
 /*! \brief  Setting RTP payload types from lines in a SDP description: */
 void ast_rtp_pt_clear(struct ast_rtp* rtp);
 /*! \brief Set payload types to defaults */
diff --git a/main/rtp.c b/main/rtp.c
index a1aa2985f..d75fe4852 100644
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -172,6 +172,7 @@ struct ast_rtp {
 	struct ast_rtcp *rtcp;
 	struct ast_codec_pref pref;
 	struct ast_rtp *bridged;        /*!< Who we are Packet bridged to */
+	int set_marker_bit:1;           /*!< Whether to set the marker bit or not */
 };
 
 /* Forward declarations */
@@ -1994,6 +1995,12 @@ int ast_rtp_settos(struct ast_rtp *rtp, int tos)
 	return res;
 }
 
+void ast_rtp_new_source(struct ast_rtp *rtp)
+{
+	rtp->set_marker_bit = 1;
+	return;
+}
+
 void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
 {
 	rtp->them.sin_port = them->sin_port;
@@ -2641,6 +2648,13 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
 			}
 		}
 	}
+
+	/* If we have been explicitly told to set the marker bit do so */
+	if (rtp->set_marker_bit) {
+		mark = 1;
+		rtp->set_marker_bit = 0;
+	}
+
 	/* If the timestamp for non-digit packets has moved beyond the timestamp
 	   for digits, update the digit timestamp.
 	*/