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/trunk@252089 f38db490-d61c-443f-a65b-d21fe96a405b

12 files changed, 122 insertions, 59 deletions

diff --git a/addons/chan_ooh323.c b/addons/chan_ooh323.c
index d7a4dfc88..7724c21d4 100644
--- a/addons/chan_ooh323.c
+++ b/addons/chan_ooh323.c
@@ -1206,10 +1206,12 @@ static int ooh323_indicate(struct ast_channel *ast, int condition, const void *d
 	    		ooManualRingback(callToken);
 	    }
 	 break;
-      case AST_CONTROL_SRCUPDATE:
-		ast_rtp_instance_new_source(p->rtp);
+	case AST_CONTROL_SRCUPDATE:
+		ast_rtp_instance_update_source(p->rtp);
+		break;
+	case AST_CONTROL_SRCCHANGE:
+		ast_rtp_instance_change_source(p->rtp);
 		break;
-
       case AST_CONTROL_CONNECTED_LINE:
 		if (gH323Debug)
 			ast_log(LOG_DEBUG, "Sending connected line info for %s (%s)\n",
diff --git a/channels/chan_h323.c b/channels/chan_h323.c
index 145a3f5bb..f2a6bc511 100644
--- a/channels/chan_h323.c
+++ b/channels/chan_h323.c
@@ -914,7 +914,11 @@ static int oh323_indicate(struct ast_channel *c, int condition, const void *data
 		res = 0;
 		break;
 	case AST_CONTROL_SRCUPDATE:
-		ast_rtp_instance_new_source(pvt->rtp);
+		ast_rtp_instance_update_source(pvt->rtp);
+		res = 0;
+		break;
+	case AST_CONTROL_SRCCHANGE:
+		ast_rtp_instance_change_source(pvt->rtp);
 		res = 0;
 		break;
 	case AST_CONTROL_PROCEEDING:
diff --git a/channels/chan_mgcp.c b/channels/chan_mgcp.c
index 5dfafb84a..16195cb2e 100644
--- a/channels/chan_mgcp.c
+++ b/channels/chan_mgcp.c
@@ -1456,7 +1456,10 @@ 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_instance_new_source(sub->rtp);
+		ast_rtp_instance_update_source(sub->rtp);
+		break;
+	case AST_CONTROL_SRCCHANGE:
+		ast_rtp_instance_change_source(sub->rtp);
 		break;
 	case AST_CONTROL_PROGRESS:
 	case AST_CONTROL_PROCEEDING:
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b8f93b583..ae1cbb3a2 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -3903,7 +3903,6 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
 	if (dialog->rtp) { /* Audio */
 		ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
 		ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
-		ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_CONSTANT_SSRC, ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC));
 		ast_rtp_instance_set_timeout(dialog->rtp, peer->rtptimeout);
 		ast_rtp_instance_set_hold_timeout(dialog->rtp, peer->rtpholdtimeout);
 		/* Set Frame packetization */
@@ -3913,7 +3912,6 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
 	if (dialog->vrtp) { /* Video */
 		ast_rtp_instance_set_timeout(dialog->vrtp, peer->rtptimeout);
 		ast_rtp_instance_set_hold_timeout(dialog->vrtp, peer->rtpholdtimeout);
-		ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_CONSTANT_SSRC, ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC));
 	}
 	if (dialog->trtp) { /* Realtime text */
 		ast_rtp_instance_set_timeout(dialog->trtp, peer->rtptimeout);
@@ -4970,7 +4968,7 @@ 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_instance_new_source(p->rtp);
+		ast_rtp_instance_update_source(p->rtp);
 		res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE, TRUE);
 		ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
 	}
@@ -5002,7 +5000,7 @@ 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_instance_new_source(p->rtp);
+					ast_rtp_instance_update_source(p->rtp);
 					if (!global_prematuremediafilter) {
 						p->invitestate = INV_EARLY_MEDIA;
 						transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
@@ -5333,11 +5331,11 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
 		res = -1;
 		break;
 	case AST_CONTROL_HOLD:
-		ast_rtp_instance_new_source(p->rtp);
+		ast_rtp_instance_update_source(p->rtp);
 		ast_moh_start(ast, data, p->mohinterpret);
 		break;
 	case AST_CONTROL_UNHOLD:
-		ast_rtp_instance_new_source(p->rtp);
+		ast_rtp_instance_update_source(p->rtp);
 		ast_moh_stop(ast);
 		break;
 	case AST_CONTROL_VIDUPDATE:	/* Request a video frame update */
@@ -5356,7 +5354,10 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
 		}
 		break;
 	case AST_CONTROL_SRCUPDATE:
-		ast_rtp_instance_new_source(p->rtp);
+		ast_rtp_instance_update_source(p->rtp);
+		break;
+	case AST_CONTROL_SRCCHANGE:
+		ast_rtp_instance_change_source(p->rtp);
 		break;
 	case AST_CONTROL_CONNECTED_LINE:
 		update_connectedline(p, data, datalen);
@@ -19232,12 +19233,6 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
 				res = -1;
 				goto request_invite_cleanup;
 			}
-			if (p->rtp) {
-				ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_CONSTANT_SSRC, ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC));
-			}
-			if (p->vrtp) {
-				ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_CONSTANT_SSRC, ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC));
-			}
 		} else {	/* No SDP in invite, call control session */
 			p->jointcapability = p->capability;
 			ast_debug(2, "No SDP in Invite, third party call control\n");
@@ -22595,9 +22590,6 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
 	} else if (!strcasecmp(v->name, "buggymwi")) {
 		ast_set_flag(&mask[1], SIP_PAGE2_BUGGY_MWI);
 		ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_BUGGY_MWI);
-	} 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;
 
diff --git a/channels/chan_skinny.c b/channels/chan_skinny.c
index 136e2699c..a19fdc15f 100644
--- a/channels/chan_skinny.c
+++ b/channels/chan_skinny.c
@@ -4304,7 +4304,10 @@ 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_instance_new_source(sub->rtp);
+		ast_rtp_instance_update_source(sub->rtp);
+		break;
+	case AST_CONTROL_SRCCHANGE:
+		ast_rtp_instance_change_source(sub->rtp);
 		break;
 	case AST_CONTROL_CONNECTED_LINE:
 		update_connectedline(sub, data, datalen);
diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h
index 6ae1ce069..57fac84ca 100644
--- a/channels/sip/include/sip.h
+++ b/channels/sip/include/sip.h
@@ -308,7 +308,6 @@
 #define SIP_PAGE2_Q850_REASON           (1 << 4)    /*!< DP: Get/send cause code via Reason header */
 
 /* Space for addition of other realtime flags in the future */
-#define SIP_PAGE2_CONSTANT_SSRC         (1 << 7)    /*!< GDP: Don't change SSRC on reinvite */
 #define SIP_PAGE2_SYMMETRICRTP          (1 << 8)    /*!< GDP: Whether symmetric RTP is enabled or not */
 #define SIP_PAGE2_STATECHANGEQUEUE      (1 << 9)    /*!< D: Unsent state pending change exists */
 
@@ -346,7 +345,7 @@
 	SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | \
 	SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_FAX_DETECT | \
 	SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS | SIP_PAGE2_PREFERRED_CODEC | \
-	SIP_PAGE2_RPID_IMMEDIATE | SIP_PAGE2_RPID_UPDATE | SIP_PAGE2_SYMMETRICRTP | SIP_PAGE2_CONSTANT_SSRC |\
+	SIP_PAGE2_RPID_IMMEDIATE | SIP_PAGE2_RPID_UPDATE | SIP_PAGE2_SYMMETRICRTP |\
 	SIP_PAGE2_Q850_REASON)
 
 /*@}*/
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index c4c4b83ea..a2d25f261 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -789,8 +789,6 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;sdpowner=root                  ; Allows you to change the username field in the SDP owner string, (o=)
                                 ; This field MUST NOT contain spaces
 
-;constantssrc=yes               ; Don't change the RTP SSRC when our media stream changes
-
 ;----------------------------------------- REALTIME SUPPORT ------------------------
 ; For additional information on ARA, the Asterisk Realtime Architecture,
 ; please read realtime.txt and extconfig.txt in the /doc directory of the
@@ -1003,7 +1001,6 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ; timerb
 ; qualifyfreq
 ; t38pt_usertpsource
-; constantssrc
 ; contactpermit         ; Limit what a host may register as (a neat trick
 ; contactdeny           ; is to register at the same IP as a SIP provider,
 ;                       ; then call oneself, and get redirected to that
diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index f9a9a4f0a..6f45386cd 100644
--- a/include/asterisk/frame.h
+++ b/include/asterisk/frame.h
@@ -85,7 +85,8 @@ struct ast_codec_pref {
  * \arg \b HOLD            Call is placed on hold
  * \arg \b UNHOLD          Call is back from hold
  * \arg \b VIDUPDATE       Video update requested
- * \arg \b SRCUPDATE       The source of media has changed
+ * \arg \b SRCUPDATE       The source of media has changed (RTP marker bit must change)
+ * \arg \b SRCCHANGE       Media source has changed (RTP marker bit and SSRC must change)
  * \arg \b CONNECTED_LINE  Connected line has changed
  * \arg \b REDIRECTING     Call redirecting information has changed.
  */
@@ -323,6 +324,7 @@ enum ast_control_frame_type {
 	AST_CONTROL_CONNECTED_LINE = 22,/*!< Indicate connected line has changed */
 	AST_CONTROL_REDIRECTING = 23,    /*!< Indicate redirecting id has changed */
 	AST_CONTROL_T38_PARAMETERS = 24, /*! T38 state change request/notification with parameters */
+	AST_CONTROL_SRCCHANGE = 25,  /*!< Media source has changed and requires a new RTP SSRC */
 };
 
 enum ast_control_t38 {
diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h
index 86f7c27e7..770f4d2f5 100644
--- a/include/asterisk/rtp_engine.h
+++ b/include/asterisk/rtp_engine.h
@@ -92,8 +92,6 @@ enum ast_rtp_property {
 	AST_RTP_PROPERTY_STUN,
 	/*! Enable RTCP support */
 	AST_RTP_PROPERTY_RTCP,
-	/*! Don't force a new SSRC on new source */
-	AST_RTP_PROPERTY_CONSTANT_SSRC,
 
 	/*!
 	 * \brief Maximum number of RTP properties supported
@@ -322,10 +320,10 @@ struct ast_rtp_engine {
 	int (*dtmf_begin)(struct ast_rtp_instance *instance, char digit);
 	/*! Callback for stopping RFC2833 DTMF transmission */
 	int (*dtmf_end)(struct ast_rtp_instance *instance, char digit);
-	/*! Callback to indicate that a new source of media has come in */
-	void (*new_source)(struct ast_rtp_instance *instance);
-	/*! Callback to tell new_source not to change SSRC */
-	void (*constant_ssrc_set)(struct ast_rtp_instance *instance);
+	/*! Callback to indicate that we should update the marker bit */
+	void (*update_source)(struct ast_rtp_instance *instance);
+	/*! Callback to indicate that we should update the marker bit and ssrc */
+	void (*change_source)(struct ast_rtp_instance *instance);
 	/*! Callback for setting an extended RTP property */
 	int (*extended_prop_set)(struct ast_rtp_instance *instance, int property, void *value);
 	/*! Callback for getting an extended RTP property */
@@ -1192,22 +1190,40 @@ int ast_rtp_instance_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_r
 enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *instance);
 
 /*!
- * \brief Indicate a new source of audio has dropped in
+ * \brief Indicate that the RTP marker bit should be set on an RTP stream
  *
  * \param instance Instance that the new media source is feeding into
  *
  * Example usage:
  *
  * \code
- * ast_rtp_instance_new_source(instance);
+ * ast_rtp_instance_update_source(instance);
  * \endcode
  *
- * This indicates that a new source of media is feeding the instance pointed to by
- * instance.
+ * This indicates that the source of media that is feeding the instance pointed to by
+ * instance has been updated and that the marker bit should be set.
  *
  * \since 1.8
  */
-void ast_rtp_instance_new_source(struct ast_rtp_instance *instance);
+void ast_rtp_instance_update_source(struct ast_rtp_instance *instance);
+
+/*!
+ * \brief Indicate a new source of audio has dropped in and the ssrc should change
+ *
+ * \param instance Instance that the new media source is feeding into
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_change_source(instance);
+ * \endcode
+ *
+ * This indicates that the source of media that is feeding the instance pointed to by
+ * instance has changed and that the marker bit should be set and the SSRC updated.
+ *
+ * \since 1.8
+ */
+void ast_rtp_instance_change_source(struct ast_rtp_instance *instance);
 
 /*!
  * \brief Set QoS parameters on an RTP session
diff --git a/main/channel.c b/main/channel.c
index 833de79b3..016453c24 100644
--- a/main/channel.c
+++ b/main/channel.c
@@ -2964,6 +2964,7 @@ 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:
 				case AST_CONTROL_CONNECTED_LINE:
 				case AST_CONTROL_REDIRECTING:
 					/* Unimportant */
@@ -3599,6 +3600,7 @@ 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:
@@ -3739,6 +3741,7 @@ 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:
@@ -4474,6 +4477,7 @@ struct ast_channel *__ast_request_and_dial(const char *type, format_t format, co
 				case AST_CONTROL_UNHOLD:
 				case AST_CONTROL_VIDUPDATE:
 				case AST_CONTROL_SRCUPDATE:
+				case AST_CONTROL_SRCCHANGE:
 				case AST_CONTROL_CONNECTED_LINE:
 				case AST_CONTROL_REDIRECTING:
 				case -1:			/* Ignore -- just stopping indications */
@@ -5724,6 +5728,7 @@ 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:
 				ast_indicate_data(other, f->subclass.integer, f->data.ptr, f->datalen);
 				if (jb_in_use) {
 					ast_jb_empty_and_reset(c0, c1);
diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index 8085e9d9e..88f534614 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -728,10 +728,17 @@ enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *i
 	return instance->dtmf_mode;
 }
 
-void ast_rtp_instance_new_source(struct ast_rtp_instance *instance)
+void ast_rtp_instance_update_source(struct ast_rtp_instance *instance)
 {
-	if (instance->engine->new_source) {
-		instance->engine->new_source(instance);
+	if (instance->engine->update_source) {
+		instance->engine->update_source(instance);
+	}
+}
+
+void ast_rtp_instance_change_source(struct ast_rtp_instance *instance)
+{
+	if (instance->engine->change_source) {
+		instance->engine->change_source(instance);
 	}
 }
 
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index d8a279199..fb060c75f 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -254,7 +254,8 @@ static int ast_rtp_new(struct ast_rtp_instance *instance, struct sched_context *
 static int ast_rtp_destroy(struct ast_rtp_instance *instance);
 static int ast_rtp_dtmf_begin(struct ast_rtp_instance *instance, char digit);
 static int ast_rtp_dtmf_end(struct ast_rtp_instance *instance, char digit);
-static void ast_rtp_new_source(struct ast_rtp_instance *instance);
+static void ast_rtp_update_source(struct ast_rtp_instance *instance);
+static void ast_rtp_change_source(struct ast_rtp_instance *instance);
 static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame);
 static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtcp);
 static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value);
@@ -276,7 +277,8 @@ static struct ast_rtp_engine asterisk_rtp_engine = {
 	.destroy = ast_rtp_destroy,
 	.dtmf_begin = ast_rtp_dtmf_begin,
 	.dtmf_end = ast_rtp_dtmf_end,
-	.new_source = ast_rtp_new_source,
+	.update_source = ast_rtp_update_source,
+	.change_source = ast_rtp_change_source,
 	.write = ast_rtp_write,
 	.read = ast_rtp_read,
 	.prop_set = ast_rtp_prop_set,
@@ -655,16 +657,27 @@ static int ast_rtp_dtmf_end(struct ast_rtp_instance *instance, char digit)
 	return 0;
 }
 
-static void ast_rtp_new_source(struct ast_rtp_instance *instance)
+static void ast_rtp_update_source(struct ast_rtp_instance *instance)
 {
 	struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 
 	/* We simply set this bit so that the next packet sent will have the marker bit turned on */
 	ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
+	ast_debug(3, "Setting the marker bit due to a source update\n");
 
-	if (!ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_CONSTANT_SSRC)) {
-		rtp->ssrc = ast_random();
-	}
+	return;
+}
+
+static void ast_rtp_change_source(struct ast_rtp_instance *instance)
+{
+	struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+	unsigned int ssrc = ast_random();
+
+	/* We simply set this bit so that the next packet sent will have the marker bit turned on */
+	ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
+
+	ast_debug(3, "Changing ssrc from %u to %u due to a source change\n", rtp->ssrc, ssrc);
+	rtp->ssrc = ssrc;
 
 	return;
 }
@@ -1854,6 +1867,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 	unsigned int *rtpheader = (unsigned int*)(rtp->rawdata + AST_FRIENDLY_OFFSET), seqno, ssrc, timestamp;
 	struct ast_rtp_payload_type payload;
 	struct sockaddr_in remote_address = { 0, };
+	AST_LIST_HEAD_NOLOCK(, ast_frame) frames;
 
 	/* If this is actually RTCP let's hop on over and handle it */
 	if (rtcp) {
@@ -1951,14 +1965,27 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 	timestamp = ntohl(rtpheader[1]);
 	ssrc = ntohl(rtpheader[2]);
 
-	/* Force a marker bit if the SSRC changes */
-	if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) {
-		if (option_debug || rtpdebug) {
-			ast_debug(1, "Forcing Marker bit, because SSRC has changed\n");
+	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.integer = AST_CONTROL_SRCCHANGE,
+		};
+
+		if (!mark) {
+			if (option_debug || rtpdebug) {
+				ast_debug(1, "Forcing Marker bit, because SSRC has changed\n");
+			}
+			mark = 1;
 		}
-		mark = 1;
+
+		f = ast_frisolate(&srcupdate);
+		AST_LIST_INSERT_TAIL(&frames, f, frame_list);
 	}
 
+	rtp->rxssrc = ssrc;
+
 	/* Remove any padding bytes that may be present */
 	if (padding) {
 		res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1];
@@ -1986,7 +2013,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 	/* Make sure after we potentially mucked with the header length that it is once again valid */
 	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++;
@@ -2029,7 +2056,11 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 			ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(remote_address.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.codec = payload.code;
@@ -2046,7 +2077,8 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 			f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass.codec)), 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);
 		}
 	}
 
@@ -2081,7 +2113,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 		rtp->f.subclass.codec = AST_FORMAT_T140;
 		header_end = memchr(data, ((*data) & 0x7f), rtp->f.datalen);
 		if (header_end == NULL) {
-			return &ast_null_frame;
+			return AST_LIST_FIRST(&frames) ? AST_LIST_FIRST(&frames) : &ast_null_frame;
 		}
 		header_end++;
 
@@ -2094,7 +2126,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 				len += data[x * 4 + 3];
 
 			if (!(rtp->f.datalen - len))
-				return &ast_null_frame;
+				return AST_LIST_FIRST(&frames) ? AST_LIST_FIRST(&frames) : &ast_null_frame;
 
 			rtp->f.data.ptr += len;
 			rtp->f.datalen -= len;
@@ -2150,7 +2182,8 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 		rtp->f.delivery.tv_usec = 0;
 	}
 
-	return &rtp->f;
+	AST_LIST_INSERT_TAIL(&frames, &rtp->f, frame_list);
+	return AST_LIST_FIRST(&frames);
 }
 
 static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value)