Merge "res_rtp_asterisk / res_pjsip: Add support for BUNDLE."

channels/chan_pjsip.c

@@ -792,8 +792,6 @@ static struct ast_frame *chan_pjsip_read_stream(struct ast_channel *ast)
 		return f;
 	}
 
-	f->stream_num = callback_state->session->stream_num;
-
 	if (f->frametype != AST_FRAME_VOICE ||
 		callback_state->session != session->active_media_state->default_session[callback_state->session->type]) {
 		return f;

include/asterisk/res_pjsip.h

@@ -688,6 +688,8 @@ struct ast_sip_endpoint_media_configuration {
 	unsigned int max_audio_streams;
 	/*! Maximum number of video streams to offer/accept */
 	unsigned int max_video_streams;
+	/*! Use BUNDLE */
+	unsigned int bundle;
 };
 
 /*!

include/asterisk/res_pjsip_session.h

@@ -99,6 +99,12 @@ struct ast_sip_session_media {
 	ast_sip_session_media_write_cb write_callback;
 	/*! \brief The stream number to place into any resulting frames */
 	int stream_num;
+	/*! \brief Media identifier for this stream (may be shared across multiple streams) */
+	char *mid;
+	/*! \brief The bundle group the stream belongs to */
+	int bundle_group;
+	/*! \brief Whether this stream is currently bundled or not */
+	unsigned int bundled;
 };
 
 /*!
@@ -833,6 +839,19 @@ int ast_sip_session_media_add_read_callback(struct ast_sip_session *session, str
 int ast_sip_session_media_set_write_callback(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
 	ast_sip_session_media_write_cb callback);
 
+/*!
+ * \brief Retrieve the underlying media session that is acting as transport for a media session
+ * \since 15.0.0
+ *
+ * \param session The session
+ * \param session_media The media session to retrieve the transport for
+ *
+ * \note This operates on the pending media state
+ *
+ * \note This function is guaranteed to return non-NULL
+ */
+struct ast_sip_session_media *ast_sip_session_media_get_transport(struct ast_sip_session *session, struct ast_sip_session_media *session_media);
+
 /*! \brief Determines whether the res_pjsip_session module is loaded */
 #define CHECK_PJSIP_SESSION_MODULE_LOADED()				\
 	do {								\

include/asterisk/rtp_engine.h

@@ -603,6 +603,12 @@ struct ast_rtp_engine {
 	unsigned int (*ssrc_get)(struct ast_rtp_instance *instance);
 	/*! Callback to retrieve RTCP SDES CNAME */
 	const char *(*cname_get)(struct ast_rtp_instance *instance);
+	/*! Callback to bundle an RTP instance to another */
+	int (*bundle)(struct ast_rtp_instance *child, struct ast_rtp_instance *parent);
+	/*! Callback to set remote SSRC information */
+	void (*set_remote_ssrc)(struct ast_rtp_instance *instance, unsigned int ssrc);
+	/*! Callback to set the stream identifier */
+	void (*set_stream_num)(struct ast_rtp_instance *instance, int stream_num);
 	/*! Callback to pointer for optional ICE support */
 	struct ast_rtp_engine_ice *ice;
 	/*! Callback to pointer for optional DTLS SRTP support */
@@ -1511,6 +1517,20 @@ void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, struct ast_fo
  */
 int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, int asterisk_format, struct ast_format *format, int code);
 
+/*!
+ * \brief Set a payload code for use with a specific Asterisk format
+ *
+ * \param codecs Codecs structure to manipulate
+ * \param code The payload code
+ * \param format Asterisk format
+ *
+ * \retval 0 Payload was set to the given format
+ * \retval -1 Payload was in use or could not be set
+ *
+ * \since 15.0.0
+ */
+int ast_rtp_codecs_payload_set_rx(struct ast_rtp_codecs *codecs, int code, struct ast_format *format);
+
 /*!
  * \brief Retrieve a tx mapped payload type based on whether it is an Asterisk format and the code
  * \since 14.0.0
@@ -2271,6 +2291,8 @@ int ast_rtp_instance_sendcng(struct ast_rtp_instance *instance, int level);
  *
  * \retval 0 Success
  * \retval non-zero Failure
+ *
+ * \note If no remote policy is provided any existing SRTP policies are left and the new local policy is added
  */
 int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct ast_srtp_policy* remote_policy, struct ast_srtp_policy *local_policy, int rtcp);
 
@@ -2416,6 +2438,36 @@ unsigned int ast_rtp_instance_get_ssrc(struct ast_rtp_instance *rtp);
  */
 const char *ast_rtp_instance_get_cname(struct ast_rtp_instance *rtp);
 
+/*!
+ * \brief Request that an RTP instance be bundled with another
+ * \since 15.0.0
+ *
+ * \param child The child RTP instance
+ * \param parent The parent RTP instance the child should be bundled with
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ */
+int ast_rtp_instance_bundle(struct ast_rtp_instance *child, struct ast_rtp_instance *parent);
+
+/*!
+ * \brief Set the remote SSRC for an RTP instance
+ * \since 15.0.0
+ *
+ * \param rtp The RTP instance
+ * \param ssrc The remote SSRC
+ */
+void ast_rtp_instance_set_remote_ssrc(struct ast_rtp_instance *rtp, unsigned int ssrc);
+
+/*!
+ * \brief Set the stream number for an RTP instance
+ * \since 15.0.0
+ *
+ * \param rtp The RTP instance
+ * \param stream_num The stream identifier number
+ */
+void ast_rtp_instance_set_stream_num(struct ast_rtp_instance *instance, int stream_num);
+
 /*! \addtogroup StasisTopicsAndMessages
  * @{
  */

main/rtp_engine.c

@@ -1495,22 +1495,25 @@ static int rtp_codecs_find_non_primary_dynamic_rx(struct ast_rtp_codecs *codecs)
  * \param asterisk_format Non-zero if the given Asterisk format is present
  * \param format Asterisk format to look for
  * \param code The format to look for
+ * \param explicit Require the provided code to be explicitly used
  *
  * \note It is assumed that static_RTP_PT_lock is at least read locked before calling.
  *
  * \retval Numerical payload type
  * \retval -1 if could not assign.
  */
-static int rtp_codecs_assign_payload_code_rx(struct ast_rtp_codecs *codecs, int asterisk_format, struct ast_format *format, int code)
+static int rtp_codecs_assign_payload_code_rx(struct ast_rtp_codecs *codecs, int asterisk_format, struct ast_format *format, int code, int explicit)
 {
-	int payload;
+	int payload = code;
 	struct ast_rtp_payload_type *new_type;
 
+	if (!explicit) {
 		payload = find_static_payload_type(asterisk_format, format, code);
 
 		if (payload < 0 && (!asterisk_format || ast_option_rtpusedynamic)) {
 			return payload;
 		}
+	}
 
 	new_type = rtp_payload_type_alloc(format, payload, code, 1);
 	if (!new_type) {
@@ -1526,8 +1529,8 @@ static int rtp_codecs_assign_payload_code_rx(struct ast_rtp_codecs *codecs, int
 		 * or our default dynamic position is available.
 		 */
 		rtp_codecs_payload_replace_rx(codecs, payload, new_type);
-	} else if (-1 < (payload = find_unused_payload(codecs))
-		|| -1 < (payload = rtp_codecs_find_non_primary_dynamic_rx(codecs))) {
+	} else if (!explicit && (-1 < (payload = find_unused_payload(codecs))
+		|| -1 < (payload = rtp_codecs_find_non_primary_dynamic_rx(codecs)))) {
 		/*
 		 * We found the first available empty dynamic position
 		 * or we found a mapping that should no longer be
@@ -1535,6 +1538,11 @@ static int rtp_codecs_assign_payload_code_rx(struct ast_rtp_codecs *codecs, int
 		 */
 		new_type->payload = payload;
 		rtp_codecs_payload_replace_rx(codecs, payload, new_type);
+	} else if (explicit) {
+		/*
+		* They explicitly requested this payload number be used but it couldn't be
+		*/
+		payload = -1;
 	} else {
 		/*
 		 * There are no empty or non-primary dynamic positions
@@ -1595,13 +1603,18 @@ int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, int asterisk_form
 
 	if (payload < 0) {
 		payload = rtp_codecs_assign_payload_code_rx(codecs, asterisk_format, format,
-			code);
+			code, 0);
 	}
 	ast_rwlock_unlock(&static_RTP_PT_lock);
 
 	return payload;
 }
 
+int ast_rtp_codecs_payload_set_rx(struct ast_rtp_codecs *codecs, int code, struct ast_format *format)
+{
+	return rtp_codecs_assign_payload_code_rx(codecs, 1, format, code, 1);
+}
+
 int ast_rtp_codecs_payload_code_tx(struct ast_rtp_codecs *codecs, int asterisk_format, const struct ast_format *format, int code)
 {
 	struct ast_rtp_payload_type *type;
@@ -2424,7 +2437,7 @@ int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct a
 
 	if (!*srtp) {
 		res = res_srtp->create(srtp, instance, remote_policy);
-	} else {
+	} else if (remote_policy) {
 		res = res_srtp->replace(srtp, instance, remote_policy);
 	}
 	if (!res) {
@@ -3366,3 +3379,38 @@ const char *ast_rtp_instance_get_cname(struct ast_rtp_instance *rtp)
 
 	return cname;
 }
+
+int ast_rtp_instance_bundle(struct ast_rtp_instance *child, struct ast_rtp_instance *parent)
+{
+	int res = -1;
+
+	if (child->engine != parent->engine) {
+		return -1;
+	}
+
+	ao2_lock(child);
+	if (child->engine->bundle) {
+		res = child->engine->bundle(child, parent);
+	}
+	ao2_unlock(child);
+
+	return res;
+}
+
+void ast_rtp_instance_set_remote_ssrc(struct ast_rtp_instance *rtp, unsigned int ssrc)
+{
+	ao2_lock(rtp);
+	if (rtp->engine->set_remote_ssrc) {
+		rtp->engine->set_remote_ssrc(rtp, ssrc);
+	}
+	ao2_unlock(rtp);
+}
+
+void ast_rtp_instance_set_stream_num(struct ast_rtp_instance *rtp, int stream_num)
+{
+	ao2_lock(rtp);
+	if (rtp->engine->set_stream_num) {
+		rtp->engine->set_stream_num(rtp, stream_num);
+	}
+	ao2_unlock(rtp);
+}

res/res_pjsip.c

@@ -1002,6 +1002,14 @@
 						streams allowed for the endpoint.
 					</para></description>
 				</configOption>
+				<configOption name="bundle" default="no">
+					<synopsis>Enable RTP bundling</synopsis>
+					<description><para>
+						With this option enabled, Asterisk will attempt to negotiate the use of bundle.
+						If negotiated this will result in multiple RTP streams being carried over the same
+						underlying transport. Note that enabling bundle will also enable the rtcp_mux option.
+					</para></description>
+				</configOption>
 			</configObject>
 			<configObject name="auth">
 				<synopsis>Authentication type</synopsis>

res/res_pjsip/pjsip_configuration.c

@@ -1363,6 +1363,10 @@ static int sip_endpoint_apply_handler(const struct ast_sorcery *sorcery, void *o
 		return -1;
 	}
 
+	if (endpoint->media.bundle) {
+		endpoint->media.rtcp_mux = 1;
+	}
+
 	return 0;
 }
 
@@ -1985,6 +1989,7 @@ int ast_res_pjsip_initialize_configuration(void)
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "notify_early_inuse_ringing", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, notify_early_inuse_ringing));
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "max_audio_streams", "1", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, media.max_audio_streams));
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "max_video_streams", "1", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, media.max_video_streams));
+	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "bundle", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.bundle));
 
 	if (ast_sip_initialize_sorcery_transport()) {
 		ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n");

res/res_pjsip_sdp_rtp.c

@@ -317,6 +317,7 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
 
 static int set_caps(struct ast_sip_session *session,
 	struct ast_sip_session_media *session_media,
+	struct ast_sip_session_media *session_media_transport,
 	const struct pjmedia_sdp_media *stream,
 	int is_offer, struct ast_stream *asterisk_stream)
 {
@@ -376,6 +377,24 @@ static int set_caps(struct ast_sip_session *session,
 
 	ast_stream_set_formats(asterisk_stream, joint);
 
+	/* If this is a bundled stream then apply the payloads to RTP instance acting as transport to prevent conflicts */
+	if (session_media_transport != session_media && session_media->bundled) {
+		int index;
+
+		for (index = 0; index < ast_format_cap_count(joint); ++index) {
+			struct ast_format *format = ast_format_cap_get_format(joint, index);
+			int rtp_code;
+
+			/* Ensure this payload is in the bundle group transport codecs, this purposely doesn't check the return value for
+			 * things as the format is guaranteed to have a payload already.
+			 */
+			rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media->rtp), 1, format, 0);
+			ast_rtp_codecs_payload_set_rx(ast_rtp_instance_get_codecs(session_media_transport->rtp), rtp_code, format);
+
+			ao2_ref(format, -1);
+		}
+	}
+
 	if (session->channel && ast_sip_session_is_pending_stream_default(session, asterisk_stream)) {
 		ast_channel_lock(session->channel);
 		ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_UNKNOWN);
@@ -496,7 +515,8 @@ static pjmedia_sdp_attr* generate_fmtp_attr(pj_pool_t *pool, struct ast_format *
 }
 
 /*! \brief Function which adds ICE attributes to a media stream */
-static void add_ice_to_stream(struct ast_sip_session *session, struct ast_sip_session_media *session_media, pj_pool_t *pool, pjmedia_sdp_media *media)
+static void add_ice_to_stream(struct ast_sip_session *session, struct ast_sip_session_media *session_media, pj_pool_t *pool, pjmedia_sdp_media *media,
+	unsigned int include_candidates)
 {
 	struct ast_rtp_engine_ice *ice;
 	struct ao2_container *candidates;
@@ -506,8 +526,7 @@ static void add_ice_to_stream(struct ast_sip_session *session, struct ast_sip_se
 	struct ao2_iterator it_candidates;
 	struct ast_rtp_engine_ice_candidate *candidate;
 
-	if (!session->endpoint->media.rtp.ice_support || !(ice = ast_rtp_instance_get_ice(session_media->rtp)) ||
-		!(candidates = ice->get_local_candidates(session_media->rtp))) {
+	if (!session->endpoint->media.rtp.ice_support || !(ice = ast_rtp_instance_get_ice(session_media->rtp))) {
 		return;
 	}
 
@@ -521,6 +540,15 @@ static void add_ice_to_stream(struct ast_sip_session *session, struct ast_sip_se
 		media->attr[media->attr_count++] = attr;
 	}
 
+	if (!include_candidates) {
+		return;
+	}
+
+	candidates = ice->get_local_candidates(session_media->rtp);
+	if (!candidates) {
+		return;
+	}
+
 	it_candidates = ao2_iterator_init(candidates, 0);
 	for (; (candidate = ao2_iterator_next(&it_candidates)); ao2_ref(candidate, -1)) {
 		struct ast_str *attr_candidate = ast_str_create(128);
@@ -940,6 +968,63 @@ static void set_ice_components(struct ast_sip_session *session, struct ast_sip_s
 	}
 }
 
+/*! \brief Function which adds ssrc attributes to a media stream */
+static void add_ssrc_to_stream(struct ast_sip_session *session, struct ast_sip_session_media *session_media, pj_pool_t *pool, pjmedia_sdp_media *media)
+{
+	pj_str_t stmp;
+	pjmedia_sdp_attr *attr;
+	char tmp[128];
+
+	if (!session->endpoint->media.bundle || session_media->bundle_group == -1) {
+		return;
+	}
+
+	snprintf(tmp, sizeof(tmp), "%u cname:%s", ast_rtp_instance_get_ssrc(session_media->rtp), ast_rtp_instance_get_cname(session_media->rtp));
+	attr = pjmedia_sdp_attr_create(pool, "ssrc", pj_cstr(&stmp, tmp));
+	media->attr[media->attr_count++] = attr;
+}
+
+/*! \brief Function which processes ssrc attributes in a stream */
+static void process_ssrc_attributes(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
+				   const struct pjmedia_sdp_media *remote_stream)
+{
+	int index;
+
+	if (!session->endpoint->media.bundle) {
+		return;
+	}
+
+	for (index = 0; index < remote_stream->attr_count; ++index) {
+		pjmedia_sdp_attr *attr = remote_stream->attr[index];
+		char attr_value[pj_strlen(&attr->value) + 1];
+		char *ssrc_attribute_name, *ssrc_attribute_value = NULL;
+		unsigned int ssrc;
+
+		/* We only care about ssrc attributes */
+		if (pj_strcmp2(&attr->name, "ssrc")) {
+			continue;
+		}
+
+		ast_copy_pj_str(attr_value, &attr->value, sizeof(attr_value));
+
+		if ((ssrc_attribute_name = strchr(attr_value, ' '))) {
+			/* This has an actual attribute */
+			*ssrc_attribute_name++ = '\0';
+			ssrc_attribute_value = strchr(ssrc_attribute_name, ':');
+			if (ssrc_attribute_value) {
+				/* Values are actually optional according to the spec */
+				*ssrc_attribute_value++ = '\0';
+			}
+		}
+
+		if (sscanf(attr_value, "%30u", &ssrc) < 1) {
+			continue;
+		}
+
+		ast_rtp_instance_set_remote_ssrc(session_media->rtp, ssrc);
+	}
+}
+
 /*! \brief Function which negotiates an incoming media stream */
 static int negotiate_incoming_sdp_stream(struct ast_sip_session *session,
 	struct ast_sip_session_media *session_media, const pjmedia_sdp_session *sdp,
@@ -948,6 +1033,7 @@ static int negotiate_incoming_sdp_stream(struct ast_sip_session *session,
 	char host[NI_MAXHOST];
 	RAII_VAR(struct ast_sockaddr *, addrs, NULL, ast_free);
 	pjmedia_sdp_media *stream = sdp->media[index];
+	struct ast_sip_session_media *session_media_transport;
 	enum ast_media_type media_type = session_media->type;
 	enum ast_sip_session_media_encryption encryption = AST_SIP_MEDIA_ENCRYPT_NONE;
 	int res;
@@ -981,6 +1067,12 @@ static int negotiate_incoming_sdp_stream(struct ast_sip_session *session,
 		return -1;
 	}
 
+	process_ssrc_attributes(session, session_media, stream);
+
+	session_media_transport = ast_sip_session_media_get_transport(session, session_media);
+
+	if (session_media_transport == session_media || !session_media->bundled) {
+		/* If this media session is carrying actual traffic then set up those aspects */
 		session_media->remote_rtcp_mux = (pjmedia_sdp_media_find_attr2(stream, "rtcp-mux", NULL) != NULL);
 		set_ice_components(session, session_media);
 
@@ -1009,10 +1101,17 @@ static int negotiate_incoming_sdp_stream(struct ast_sip_session *session,
 			((encryption == AST_SIP_MEDIA_ENCRYPT_SDES) && !res)) {
 			pj_strdup(session->inv_session->pool, &session_media->transport, &stream->desc.transport);
 		}
+	} else {
+		/* This is bundled with another session, so mark it as such */
+		ast_rtp_instance_bundle(session_media->rtp, session_media_transport->rtp);
+
+		enable_rtcp(session, session_media, stream);
+	}
 
-	if (set_caps(session, session_media, stream, 1, asterisk_stream)) {
+	if (set_caps(session, session_media, session_media_transport, stream, 1, asterisk_stream)) {
 		return 0;
 	}
+
 	return 1;
 }
 
@@ -1032,6 +1131,7 @@ static int add_crypto_to_stream(struct ast_sip_session *session,
 	static const pj_str_t STR_PASSIVE = { "passive", 7 };
 	static const pj_str_t STR_ACTPASS = { "actpass", 7 };
 	static const pj_str_t STR_HOLDCONN = { "holdconn", 8 };
+	enum ast_rtp_dtls_setup setup;
 
 	switch (session_media->encryption) {
 	case AST_SIP_MEDIA_ENCRYPT_NONE:
@@ -1085,7 +1185,16 @@ static int add_crypto_to_stream(struct ast_sip_session *session,
 			break;
 		}
 
-		switch (dtls->get_setup(session_media->rtp)) {
+		/* If this is an answer we need to use our current state, if it's an offer we need to use
+		 * the configured value.
+		 */
+		if (pjmedia_sdp_neg_get_state(session->inv_session->neg) != PJMEDIA_SDP_NEG_STATE_DONE) {
+			setup = dtls->get_setup(session_media->rtp);
+		} else {
+			setup = session->endpoint->media.rtp.dtls_cfg.default_setup;
+		}
+
+		switch (setup) {
 		case AST_RTP_DTLS_SETUP_ACTIVE:
 			attr = pjmedia_sdp_attr_create(pool, "setup", &STR_ACTIVE);
 			media->attr[media->attr_count++] = attr;
@@ -1100,7 +1209,6 @@ static int add_crypto_to_stream(struct ast_sip_session *session,
 			break;
 		case AST_RTP_DTLS_SETUP_HOLDCONN:
 			attr = pjmedia_sdp_attr_create(pool, "setup", &STR_HOLDCONN);
-			media->attr[media->attr_count++] = attr;
 			break;
 		default:
 			break;
@@ -1152,6 +1260,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
 	int rtp_code;
 	RAII_VAR(struct ast_format_cap *, caps, NULL, ao2_cleanup);
 	enum ast_media_type media_type = session_media->type;
+	struct ast_sip_session_media *session_media_transport;
 
 	int direct_media_enabled = !ast_sockaddr_isnull(&session_media->direct_media_addr) &&
 		ast_format_cap_count(session->direct_media_cap);
@@ -1195,6 +1304,28 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
 		return -1;
 	}
 
+	/* If this stream has not been bundled already it is new and we need to ensure there is no SSRC conflict */
+	if (session_media->bundle_group != -1 && !session_media->bundled) {
+		for (index = 0; index < sdp->media_count; ++index) {
+			struct ast_sip_session_media *other_session_media;
+
+			other_session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, index);
+			if (!other_session_media->rtp || other_session_media->bundle_group != session_media->bundle_group) {
+				continue;
+			}
+
+			if (ast_rtp_instance_get_ssrc(session_media->rtp) == ast_rtp_instance_get_ssrc(other_session_media->rtp)) {
+				ast_rtp_instance_change_source(session_media->rtp);
+				/* Start the conflict check over again */
+				index = -1;
+				continue;
+			}
+		}
+	}
+
+	session_media_transport = ast_sip_session_media_get_transport(session, session_media);
+
+	if (session_media_transport == session_media || !session_media->bundled) {
 		set_ice_components(session, session_media);
 		enable_rtcp(session, session_media, NULL);
 
@@ -1252,11 +1383,27 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
 		}
 
 		/* Add ICE attributes and candidates */
-	add_ice_to_stream(session, session_media, pool, media);
+		add_ice_to_stream(session, session_media, pool, media, 1);
 
 		ast_rtp_instance_get_local_address(session_media->rtp, &addr);
 		media->desc.port = direct_media_enabled ? ast_sockaddr_port(&session_media->direct_media_addr) : (pj_uint16_t) ast_sockaddr_port(&addr);
 		media->desc.port_count = 1;
+	} else {
+		pjmedia_sdp_media *bundle_group_stream = sdp->media[session_media_transport->stream_num];
+
+		/* As this is in a bundle group it shares the same details as the group instance */
+		media->desc.transport = bundle_group_stream->desc.transport;
+		media->conn = bundle_group_stream->conn;
+		media->desc.port = bundle_group_stream->desc.port;
+
+		if (add_crypto_to_stream(session, session_media_transport, pool, media)) {
+			return -1;
+		}
+
+		add_ice_to_stream(session, session_media_transport, pool, media, 0);
+
+		enable_rtcp(session, session_media, NULL);
+	}
 
 	if (!(caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
 		ast_log(LOG_ERROR, "Failed to allocate %s capabilities\n",
@@ -1278,11 +1425,24 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
 			continue;
 		}
 
+		/* If this stream is not a transport we need to use the transport codecs structure for payload management to prevent
+		 * conflicts.
+		 */
+		if (session_media_transport != session_media) {
+			if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media_transport->rtp), 1, format, 0)) == -1) {
+				ast_log(LOG_WARNING,"Unable to get rtp codec payload code for %s\n", ast_format_get_name(format));
+				ao2_ref(format, -1);
+				continue;
+			}
+			/* Our instance has to match the payload number though */
+			ast_rtp_codecs_payload_set_rx(ast_rtp_instance_get_codecs(session_media->rtp), rtp_code, format);
+		} else {
 			if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media->rtp), 1, format, 0)) == -1) {
 				ast_log(LOG_WARNING,"Unable to get rtp codec payload code for %s\n", ast_format_get_name(format));
 				ao2_ref(format, -1);
 				continue;
 			}
+		}
 
 		if ((attr = generate_rtpmap_attr(session, media, pool, rtp_code, 1, format, 0))) {
 			media->attr[media->attr_count++] = attr;
@@ -1332,6 +1492,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
 		}
 	}
 
+
 	/* If no formats were actually added to the media stream don't add it to the SDP */
 	if (!media->desc.fmt_count) {
 		return 1;
@@ -1365,6 +1526,8 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
 		pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
 	}
 
+	add_ssrc_to_stream(session, session_media, pool, media);
+
 	/* Add the media stream to the SDP */
 	sdp->media[sdp->media_count++] = media;
 
@@ -1425,6 +1588,7 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session,
 	enum ast_media_type media_type = session_media->type;
 	char host[NI_MAXHOST];
 	int res;
+	struct ast_sip_session_media *session_media_transport;
 
 	if (!session->channel) {
 		return 1;
@@ -1441,6 +1605,11 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session,
 		return -1;
 	}
 
+	process_ssrc_attributes(session, session_media, remote_stream);
+
+	session_media_transport = ast_sip_session_media_get_transport(session, session_media);
+
+	if (session_media_transport == session_media || !session_media->bundled) {
 		session_media->remote_rtcp_mux = (pjmedia_sdp_media_find_attr2(remote_stream, "rtcp-mux", NULL) != NULL);
 		set_ice_components(session, session_media);
 
@@ -1469,9 +1638,6 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session,
 		/* Apply connection information to the RTP instance */
 		ast_sockaddr_set_port(addrs, remote_stream->desc.port);
 		ast_rtp_instance_set_remote_address(session_media->rtp, addrs);
-	if (set_caps(session, session_media, remote_stream, 0, asterisk_stream)) {
-		return 1;
-	}
 
 		ast_sip_session_media_set_write_callback(session, session_media, media_session_rtp_write_callback);
 		ast_sip_session_media_add_read_callback(session, session_media, ast_rtp_instance_fd(session_media->rtp, 0),
@@ -1483,6 +1649,16 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session,
 
 		/* If ICE support is enabled find all the needed attributes */
 		process_ice_attributes(session, session_media, remote, remote_stream);
+	} else {
+		/* This is bundled with another session, so mark it as such */
+		ast_rtp_instance_bundle(session_media->rtp, session_media_transport->rtp);
+		ast_sip_session_media_set_write_callback(session, session_media, media_session_rtp_write_callback);
+		enable_rtcp(session, session_media, remote_stream);
+	}
+
+	if (set_caps(session, session_media, session_media_transport, remote_stream, 0, asterisk_stream)) {
+		return 1;
+	}
 
 	/* Set the channel uniqueid on the RTP instance now that it is becoming active */
 	ast_channel_lock(session->channel);
@@ -1490,6 +1666,7 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session,
 	ast_channel_unlock(session->channel);
 
 	/* Ensure the RTP instance is active */
+	ast_rtp_instance_set_stream_num(session_media->rtp, ast_stream_get_position(asterisk_stream));
 	ast_rtp_instance_activate(session_media->rtp);
 
 	/* audio stream handles music on hold */

res/res_pjsip_session.c

@@ -324,6 +324,28 @@ int ast_sip_session_media_set_write_callback(struct ast_sip_session *session, st
 	return 0;
 }
 
+struct ast_sip_session_media *ast_sip_session_media_get_transport(struct ast_sip_session *session, struct ast_sip_session_media *session_media)
+{
+	int index;
+
+	if (!session->endpoint->media.bundle || ast_strlen_zero(session_media->mid)) {
+		return session_media;
+	}
+
+	for (index = 0; index < AST_VECTOR_SIZE(&session->pending_media_state->sessions); ++index) {
+		struct ast_sip_session_media *bundle_group_session_media;
+
+		bundle_group_session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, index);
+
+		/* The first session which is in the bundle group is considered the authoritative session for transport */
+		if (bundle_group_session_media->bundle_group == session_media->bundle_group) {
+			return bundle_group_session_media;
+		}
+	}
+
+	return session_media;
+}
+
 /*!
  * \brief Set an SDP stream handler for a corresponding session media.
  *
@@ -371,6 +393,8 @@ static void session_media_dtor(void *obj)
 	if (session_media->srtp) {
 		ast_sdp_srtp_destroy(session_media->srtp);
 	}
+
+	ast_free(session_media->mid);
 }
 
 struct ast_sip_session_media *ast_sip_session_media_state_add(struct ast_sip_session *session,
@@ -408,13 +432,25 @@ struct ast_sip_session_media *ast_sip_session_media_state_add(struct ast_sip_ses
 		session_media->timeout_sched_id = -1;
 		session_media->type = type;
 		session_media->stream_num = position;
+
+		if (session->endpoint->media.bundle) {
+			/* This is a new stream so create a new mid based on media type and position, which makes it unique.
+			 * If this is the result of an offer the mid will just end up getting replaced.
+			 */
+			if (ast_asprintf(&session_media->mid, "%s-%d", ast_codec_media_type2str(type), position) < 0) {
+				ao2_ref(session_media, -1);
+				return NULL;
+			}
+			session_media->bundle_group = 0;
+		} else {
+			session_media->bundle_group = -1;
+		}
 	}
 
 	AST_VECTOR_REPLACE(&media_state->sessions, position, session_media);
 
 	/* If this stream will be active in some way and it is the first of this type then consider this the default media session to match */
-	if (!media_state->default_session[type] &&
-		ast_stream_get_state(ast_stream_topology_get_stream(media_state->topology, position)) != AST_STREAM_STATE_REMOVED) {
+	if (!media_state->default_session[type] && ast_stream_get_state(ast_stream_topology_get_stream(media_state->topology, position)) != AST_STREAM_STATE_REMOVED) {
 		media_state->default_session[type] = session_media;
 	}
 
@@ -441,6 +477,78 @@ static int is_stream_limitation_reached(enum ast_media_type type, const struct a
 	}
 }
 
+static int get_mid_bundle_group(const pjmedia_sdp_session *sdp, const char *mid)
+{
+	int bundle_group = 0;
+	int index;
+
+	for (index = 0; index < sdp->attr_count; ++index) {
+		pjmedia_sdp_attr *attr = sdp->attr[index];
+		char value[pj_strlen(&attr->value) + 1], *mids = value, *attr_mid;
+
+		if (pj_strcmp2(&attr->name, "group") || pj_strncmp2(&attr->value, "BUNDLE", 6)) {
+			continue;
+		}
+
+		ast_copy_pj_str(value, &attr->value, sizeof(value));
+
+		/* Skip the BUNDLE at the front */
+		mids += 7;
+
+		while ((attr_mid = strsep(&mids, " "))) {
+			if (!strcmp(attr_mid, mid)) {
+				/* The ordering of attributes determines our internal identification of the bundle group based on number,
+				 * with -1 being not in a bundle group. Since this is only exposed internally for response purposes it's
+				 * actually even fine if things move around.
+				 */
+				return bundle_group;
+			}
+		}
+
+		bundle_group++;
+	}
+
+	return -1;
+}
+
+static int set_mid_and_bundle_group(struct ast_sip_session *session,
+	struct ast_sip_session_media *session_media,
+	const pjmedia_sdp_session *sdp,
+	const struct pjmedia_sdp_media *stream)
+{
+	pjmedia_sdp_attr *attr;
+
+	if (!session->endpoint->media.bundle) {
+		return 0;
+	}
+
+	/* By default on an incoming negotiation we assume no mid and bundle group is present */
+	ast_free(session_media->mid);
+	session_media->mid = NULL;
+	session_media->bundle_group = -1;
+	session_media->bundled = 0;
+
+	/* Grab the media identifier for the stream */
+	attr = pjmedia_sdp_media_find_attr2(stream, "mid", NULL);
+	if (!attr) {
+		return 0;
+	}
+
+	session_media->mid = ast_calloc(1, attr->value.slen + 1);
+	if (!session_media->mid) {
+		return 0;
+	}
+	ast_copy_pj_str(session_media->mid, &attr->value, attr->value.slen + 1);
+
+	/* Determine what bundle group this is part of */
+	session_media->bundle_group = get_mid_bundle_group(sdp, session_media->mid);
+
+	/* If this is actually part of a bundle group then the other side requested or accepted the bundle request */
+	session_media->bundled = session_media->bundle_group != -1;
+
+	return 0;
+}
+
 static int handle_incoming_sdp(struct ast_sip_session *session, const pjmedia_sdp_session *sdp)
 {
 	int i;
@@ -497,9 +605,13 @@ static int handle_incoming_sdp(struct ast_sip_session *session, const pjmedia_sd
 			ast_debug(1, "Declining incoming SDP media stream '%s' at position '%d'\n",
 				ast_codec_media_type2str(type), i);
 			ast_stream_set_state(stream, AST_STREAM_STATE_REMOVED);
+			session_media->bundle_group = -1;
+			session_media->bundled = 0;
 			continue;
 		}
 
+		set_mid_and_bundle_group(session, session_media, sdp, remote_stream);
+
 		if (session_media->handler) {
 			handler = session_media->handler;
 			ast_debug(1, "Negotiating incoming SDP media stream '%s' using %s SDP handler\n",
@@ -589,6 +701,8 @@ static int handle_negotiated_sdp_session_media(struct ast_sip_session_media *ses
 	/* We need a null-terminated version of the media string */
 	ast_copy_pj_str(media, &local->media[index]->desc.media, sizeof(media));
 
+	set_mid_and_bundle_group(session, session_media, remote, remote->media[index]);
+
 	handler = session_media->handler;
 	if (handler) {
 		ast_debug(1, "Applying negotiated SDP media stream '%s' using %s SDP handler\n",
@@ -3443,6 +3557,82 @@ static int add_sdp_streams(struct ast_sip_session_media *session_media,
 	return 0;
 }
 
+/*! \brief Bundle group building structure */
+struct sip_session_media_bundle_group {
+	/*! \brief The media identifiers in this bundle group */
+	char *mids[PJMEDIA_MAX_SDP_MEDIA];
+	/*! \brief SDP attribute string */
+	struct ast_str *attr_string;
+};
+
+static int add_bundle_groups(struct ast_sip_session *session, pj_pool_t *pool, pjmedia_sdp_session *answer)
+{
+	pj_str_t stmp;
+	pjmedia_sdp_attr *attr;
+	struct sip_session_media_bundle_group bundle_groups[PJMEDIA_MAX_SDP_MEDIA];
+	int index, mid_id;
+	struct sip_session_media_bundle_group *bundle_group;
+
+	if (!session->endpoint->media.bundle) {
+		return 0;
+	}
+
+	memset(bundle_groups, 0, sizeof(bundle_groups));
+
+	attr = pjmedia_sdp_attr_create(pool, "msid-semantic", pj_cstr(&stmp, "WMS *"));
+	pjmedia_sdp_attr_add(&answer->attr_count, answer->attr, attr);
+
+	/* Build the bundle group layout so we can then add it to the SDP */
+	for (index = 0; index < AST_VECTOR_SIZE(&session->pending_media_state->sessions); ++index) {
+		struct ast_sip_session_media *session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, index);
+
+		/* If this stream is not part of a bundle group we can't add it */
+		if (session_media->bundle_group == -1) {
+			continue;
+		}
+
+		bundle_group = &bundle_groups[session_media->bundle_group];
+
+		/* If this is the first mid then we need to allocate the attribute string and place BUNDLE in front */
+		if (!bundle_group->mids[0]) {
+			bundle_group->mids[0] = session_media->mid;
+			bundle_group->attr_string = ast_str_create(64);
+			if (!bundle_group->attr_string) {
+				continue;
+			}
+
+			ast_str_set(&bundle_group->attr_string, -1, "BUNDLE %s", session_media->mid);
+			continue;
+		}
+
+		for (mid_id = 1; mid_id < PJMEDIA_MAX_SDP_MEDIA; ++mid_id) {
+			if (!bundle_group->mids[mid_id]) {
+				bundle_group->mids[mid_id] = session_media->mid;
+				ast_str_append(&bundle_group->attr_string, -1, " %s", session_media->mid);
+				break;
+			} else if (!strcmp(bundle_group->mids[mid_id], session_media->mid)) {
+				break;
+			}
+		}
+	}
+
+	/* Add all bundle groups that have mids to the SDP */
+	for (index = 0; index < PJMEDIA_MAX_SDP_MEDIA; ++index) {
+		bundle_group = &bundle_groups[index];
+
+		if (!bundle_group->attr_string) {
+			continue;
+		}
+
+		attr = pjmedia_sdp_attr_create(pool, "group", pj_cstr(&stmp, ast_str_buffer(bundle_group->attr_string)));
+		pjmedia_sdp_attr_add(&answer->attr_count, answer->attr, attr);
+
+		ast_free(bundle_group->attr_string);
+	}
+
+	return 0;
+}
+
 static struct pjmedia_sdp_session *create_local_sdp(pjsip_inv_session *inv, struct ast_sip_session *session, const pjmedia_sdp_session *offer)
 {
 	static const pj_str_t STR_IN = { "IN", 2 };
@@ -3485,6 +3675,7 @@ static struct pjmedia_sdp_session *create_local_sdp(pjsip_inv_session *inv, stru
 	for (i = 0; i < ast_stream_topology_get_count(session->pending_media_state->topology); ++i) {
 		struct ast_sip_session_media *session_media;
 		struct ast_stream *stream;
+		unsigned int streams = local->media_count;
 
 		/* This code does not enforce any maximum stream count limitations as that is done on either
 		 * the handling of an incoming SDP offer or on the handling of a session refresh.
@@ -3501,12 +3692,30 @@ static struct pjmedia_sdp_session *create_local_sdp(pjsip_inv_session *inv, stru
 			return NULL;
 		}
 
+		/* If a stream was actually added then add any additional details */
+		if (streams != local->media_count) {
+			pjmedia_sdp_media *media = local->media[streams];
+			pj_str_t stmp;
+			pjmedia_sdp_attr *attr;
+
+			/* Add the media identifier if present */
+			if (!ast_strlen_zero(session_media->mid)) {
+				attr = pjmedia_sdp_attr_create(inv->pool_prov, "mid", pj_cstr(&stmp, session_media->mid));
+				pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
+			}
+		}
+
 		/* Ensure that we never exceed the maximum number of streams PJMEDIA will allow. */
 		if (local->media_count == PJMEDIA_MAX_SDP_MEDIA) {
 			break;
 		}
 	}
 
+	/* Add any bundle groups that are present on the media state */
+	if (add_bundle_groups(session, inv->pool_prov, local)) {
+		return NULL;
+	}
+
 	/* Use the connection details of an available media if possible for SDP level */
 	for (stream = 0; stream < local->media_count; stream++) {
 		if (!local->media[stream]->conn) {

res/res_pjsip_t38.c

@@ -880,11 +880,20 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
 
 static struct ast_frame *media_session_udptl_read_callback(struct ast_sip_session *session, struct ast_sip_session_media *session_media)
 {
+	struct ast_frame *frame;
+
 	if (!session_media->udptl) {
 		return &ast_null_frame;
 	}
 
-	return ast_udptl_read(session_media->udptl);
+	frame = ast_udptl_read(session_media->udptl);
+	if (!frame) {
+		return NULL;
+	}
+
+	frame->stream_num = session_media->stream_num;
+
+	return frame;
 }
 
 static int media_session_udptl_write_callback(struct ast_sip_session *session, struct ast_sip_session_media *session_media, struct ast_frame *frame)