[Asterisk-code-review] res pjsip session/BUNDLE: Handle no audio codecs on endpoint (asterisk[master])

[Joshua Colp]

Joshua Colp has submitted this change and it was merged. ( https://gerrit.asterisk.org/6557 )

Change subject: res_pjsip_session/BUNDLE:  Handle no audio codecs on endpoint
......................................................................

res_pjsip_session/BUNDLE:  Handle no audio codecs on endpoint

When an INVITE came in with both audio and video streams but there
were no audio codecs defined for the endpoint, we weren't declining
the audio stream.  Since it's usually the first/transport stream,
when the video stream was processed and tried to use the transport,
it was empty and caused a crash.  We now decline the the stream if
there are no matching codecs so when the video stream is processed,
it's now the first/transport stream and processes normally.

Change-Id: Ic854eda54c95031e66b076ecfae3041d34daa692
---
M res/res_pjsip_session.c
1 file changed, 22 insertions(+), 5 deletions(-)

Approvals:
  Joshua Colp: Looks good to me, but someone else must approve; Approved for Submit
  Kevin Harwell: Looks good to me, approved



diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c
index 163c2b8..d50937f 100644
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -549,6 +549,16 @@
 	return 0;
 }
 
+static void remove_stream_from_bundle(struct ast_sip_session_media *session_media,
+	struct ast_stream *stream)
+{
+	ast_stream_set_state(stream, AST_STREAM_STATE_REMOVED);
+	ast_free(session_media->mid);
+	session_media->mid = NULL;
+	session_media->bundle_group = -1;
+	session_media->bundled = 0;
+}
+
 static int handle_incoming_sdp(struct ast_sip_session *session, const pjmedia_sdp_session *sdp)
 {
 	int i;
@@ -611,9 +621,7 @@
 		if (!remote_stream->desc.port || is_stream_limitation_reached(type, session->endpoint, type_streams)) {
 			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;
+			remove_stream_from_bundle(session_media, stream);
 			continue;
 		}
 
@@ -628,6 +636,11 @@
 			if (res < 0) {
 				/* Catastrophic failure. Abort! */
 				return -1;
+			} else if (res == 0) {
+				ast_debug(1, "Declining incoming SDP media stream '%s' at position '%d'\n",
+					ast_codec_media_type2str(type), i);
+				remove_stream_from_bundle(session_media, stream);
+				continue;
 			} else if (res > 0) {
 				ast_debug(1, "Media stream '%s' handled by %s\n",
 					ast_codec_media_type2str(session_media->type),
@@ -655,8 +668,12 @@
 			if (res < 0) {
 				/* Catastrophic failure. Abort! */
 				return -1;
-			}
-			if (res > 0) {
+			} else if (res == 0) {
+				ast_debug(1, "Declining incoming SDP media stream '%s' at position '%d'\n",
+					ast_codec_media_type2str(type), i);
+				remove_stream_from_bundle(session_media, stream);
+				continue;
+			} else if (res > 0) {
 				ast_debug(1, "Media stream '%s' handled by %s\n",
 					ast_codec_media_type2str(session_media->type),
 					handler->id);

-- 
To view, visit https://gerrit.asterisk.org/6557
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Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-MessageType: merged
Gerrit-Change-Id: Ic854eda54c95031e66b076ecfae3041d34daa692
Gerrit-Change-Number: 6557
Gerrit-PatchSet: 3
Gerrit-Owner: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
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