1 files changed, 38 insertions, 14 deletions

diff --git a/main/rtp.c b/main/rtp.c
index d463873f7..c8e89d9a0 100644
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -1578,6 +1578,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
 	struct rtpPayloadType rtpPT;
 	struct ast_rtp *bridged = NULL;
 	int prev_seqno;
+	AST_LIST_HEAD_NOLOCK(, ast_frame) frames;
 	
 	/* If time is up, kill it */
 	if (rtp->sending_digit)
@@ -1679,10 +1680,22 @@ 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;
@@ -1713,7 +1726,7 @@ 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 */
@@ -1777,7 +1790,11 @@ 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;
@@ -1793,7 +1810,8 @@ 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);
 		}
 	}
 
@@ -1897,7 +1915,9 @@ 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
@@ -2638,18 +2658,22 @@ int ast_rtp_setqos(struct ast_rtp *rtp, int type_of_service, int class_of_servic
 	return ast_netsock_set_qos(rtp->s, type_of_service, class_of_service, 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;
 	}
 }