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, 5 insertions, 0 deletions

diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c
index 372b01bc8..d56ff5d23 100644
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -1332,6 +1332,10 @@ 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;
 }
 
@@ -1954,6 +1958,7 @@ 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");