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authortwilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b>2009-09-30 14:49:11 +0000
committertwilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b>2009-09-30 14:49:11 +0000
commit4f67fa12535deb67bc00f833d168270544281077 (patch)
treef65c97c3a71a2daada0530012670d648d3c34478 /include
parentec161909b6471d89ba0301ba2a67cac094eee718 (diff)
Change the SSRC by default when our media stream changes
Be default, change SSRC when doing an audio stream changes Asterisk doesn't honor marker bit when reinvited to already-bridged RTP streams,resulting in far-end stack discarding packets with "old" timestamps that areactually part of a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a reinvite, unless the 'constantssrc' is set to true in sip.conf. The original issue reported to Digium support detailed the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in fromITSP, Asterisk dials the app server which sends a re-invite back toAsterisk--not to negotiate to send media directly to the ITSP, but to indicatethat it's changing the stream it's sending to Asterisk. The app servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker bit on the new stream. Asterisk passes through the teimstamp of the new stream, butdoes not reset the SSRC, sequence numbers, or set the marker bit. When the timestamp on the new stream is older than the timestamp on the originalstream, the ITSP (which doesn't know there has been any change) discards the newframes because it thinks they are too old. This patch addresses this by changing the SSRC on a stream update unless constantssrc=true is set in sip.conf. Review: https://reviewboard.asterisk.org/r/374/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@221086 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'include')
-rw-r--r--include/asterisk/rtp.h3
1 files changed, 3 insertions, 0 deletions
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index f9da4bdac..fada0bcb4 100644
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -179,6 +179,9 @@ int ast_rtp_sendcng(struct ast_rtp *rtp, int level);
int ast_rtp_settos(struct ast_rtp *rtp, int tos);
+/*! \brief When changing sources, don't generate a new SSRC */
+void ast_rtp_set_constantssrc(struct ast_rtp *rtp);
+
void ast_rtp_new_source(struct ast_rtp *rtp);
/*! \brief Setting RTP payload types from lines in a SDP description: */