res_rtp_asterisk / res_pjsip: Add support for BUNDLE.

BUNDLE is a specification used in WebRTC to allow multiple
streams to use the same underlying transport. This reduces
the number of ICE and DTLS negotiations that has to occur
to 1 normally.

This change implements this by adding support for it to
the RTP SDP module in PJSIP. BUNDLE can be turned on using
the "bundle" option and on an offer we will offer to
bundle streams together. On an answer we will accept any
bundle groups provided. Once accepted each stream is bundled
to another RTP instance for transport.

For the res_rtp_asterisk changes the ability to bundle
an RTP instance to another based on the SSRC received
from the remote side has been added. For outgoing traffic
if an RTP instance is bundled to another we will use the
other RTP instance for any transport related things. For
incoming traffic received from the transport instance we
look up the correct instance based on the SSRC and use it
for any non-transport related data.

ASTERISK-27118

Change-Id: I96c0920b9f9aca7382256484765a239017973c11

1 files changed, 58 insertions, 10 deletions

diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index 9cfae09f4..abd4b1fcf 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -1495,21 +1495,24 @@ 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;
 
-	payload = find_static_payload_type(asterisk_format, format, code);
+	if (!explicit) {
+		payload = find_static_payload_type(asterisk_format, format, code);
 
-	if (payload < 0 && (!asterisk_format || ast_option_rtpusedynamic)) {
-		return payload;
+		if (payload < 0 && (!asterisk_format || ast_option_rtpusedynamic)) {
+			return payload;
+		}
 	}
 
 	new_type = rtp_payload_type_alloc(format, payload, code, 1);
@@ -1525,9 +1528,9 @@ static int rtp_codecs_assign_payload_code_rx(struct ast_rtp_codecs *codecs, int
 		 * The payload type is a static assignment
 		 * 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))) {
+		rtp_codecs_payload_replace_rx(codecs, payload, new_type);
+	} 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);
+}
\ No newline at end of file