[Asterisk-code-review] res pjsip/rtp: No joint capabilities between streams. (asterisk[15])

[Jenkins2]

Jenkins2 has submitted this change and it was merged. ( https://gerrit.asterisk.org/9834 )

Change subject: res_pjsip/rtp: No joint capabilities between streams.
......................................................................

res_pjsip/rtp: No joint capabilities between streams.

When a conference contained a mixture of audio/video and audio-only
users, a NOTICE message would pop up stating there are no joint
capabilities between streams. This happens because streams can never be
removed, but they can be in a REMOVED state. If we have the scenario
where user A joins with audio/video, user B joins with audio-only, and
user C joins with audio/video, then user A leaves, the message would
be triggered. That removed stream is still in the SDP, but Asterisk
would pass it through, causing it to be seen as a ulaw stream. A check
has been added for removed streams, setting their status to REMOVED when
handling negotiated SDPs.

Also addressed an issue where user A joins, then user B joins but does
not receive video until much later. Full frames were not being sent,
causing some PLI from the browser. Because the video was flowing in one
direction, the browser sets the SSRC to 1, but Asterisk was dropping the
PLI because of that. Added a check to see if the SSRC is 1 or not, which
sends full frames and allows video to flow between user A and user B.
This should only happen when dealing with PSFB or FUR, and in the case
of PSFB, only for PLI.

ASTERISK-27398

Change-Id: I26e7c6f101bc119549eeca406b5bcd25ad8ebc5e
---
M res/res_pjsip_session.c
M res/res_rtp_asterisk.c
2 files changed, 23 insertions(+), 5 deletions(-)

Approvals:
  Joshua Colp: Looks good to me, but someone else must approve
  George Joseph: Looks good to me, approved
  Jenkins2: Approved for Submit



diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c
index 5796117..81e7a55 100644
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -931,10 +931,18 @@
 		session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, i);
 		stream = ast_stream_topology_get_stream(session->pending_media_state->topology, i);
 
-		/* The stream state will have already been set to removed when either we
-		 * negotiate the incoming SDP stream or when we produce our own local SDP.
-		 * This can occur if an internal thing has requested it to be removed, or if
-		 * we remove it as a result of the stream limit being reached.
+		/* Make sure that this stream is in the correct state. If we need to change
+		 * the state to REMOVED, then our work here is done, so go ahead and move on
+		 * to the next stream.
+		 */
+		if (!remote->media[i]->desc.port) {
+			ast_stream_set_state(stream, AST_STREAM_STATE_REMOVED);
+			continue;
+		}
+
+		/* If the stream state is REMOVED, nothing needs to be done, so move on to the
+		 * next stream. This can occur if an internal thing has requested it to be
+		 * removed, or if we remove it as a result of the stream limit being reached.
 		 */
 		if (ast_stream_get_state(stream) == AST_STREAM_STATE_REMOVED) {
 			/*
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 1327d71..1d1d66e 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -5802,7 +5802,17 @@
 		}
 
 		if (ssrc_valid && rtp->themssrc_valid) {
-			if (ssrc != rtp->themssrc && use_packet_source) {
+			/*
+			 * If the SSRC is 1, we still need to handle RTCP since this could be a
+			 * special case. For example, if we have a unidirectional video stream, the
+			 * SSRC may be set to 1 by the browser (in the case of chromium), and requests
+			 * will still need to be processed so that video can flow as expected. This
+			 * should only be done for PLI and FUR, since there is not a way to get the
+			 * appropriate rtp instance when the SSRC is 1.
+			 */
+			int exception = (ssrc == 1 && !((pt == RTCP_PT_PSFB && rc == AST_RTP_RTCP_FMT_PLI) || pt == RTCP_PT_FUR));
+			if ((ssrc != rtp->themssrc && use_packet_source && ssrc != 1)
+					|| exception) {
 				/*
 				 * Skip over this RTCP record as it does not contain the
 				 * correct SSRC.  We should not act upon RTCP records

-- 
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Gerrit-Project: asterisk
Gerrit-Branch: 15
Gerrit-MessageType: merged
Gerrit-Change-Id: I26e7c6f101bc119549eeca406b5bcd25ad8ebc5e
Gerrit-Change-Number: 9834
Gerrit-PatchSet: 3
Gerrit-Owner: Benjamin Keith Ford <bford at digium.com>
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
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