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, 52 insertions, 0 deletions

diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h
index 5f439163f..20ae959e9 100644
--- a/include/asterisk/rtp_engine.h
+++ b/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 */
@@ -1507,6 +1513,20 @@ 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
  *
@@ -2266,6 +2286,8 @@ 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);
 
@@ -2411,6 +2433,36 @@ 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
  * @{
  */