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Merge "res_pjsip/rtp: No joint capabilities between streams." into 16

This commit is contained in:
George Joseph 2018-08-15 09:44:57 -05:00 committed by Gerrit Code Review
parent c03ab4c4b1 a46fcaca7b
commit 100ffc6866
2 changed files with 23 additions and 5 deletions
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16
res/res_pjsip_session.c
View File

@ -930,10 +930,18 @@ static int handle_negotiated_sdp(struct ast_sip_session *session, const pjmedia_
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) {
/*

12
res/res_rtp_asterisk.c
View File

@ -5802,7 +5802,17 @@ static struct ast_frame *ast_rtcp_interpret(struct ast_rtp_instance *instance, s
}
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