diff options
| author | twilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-09-30 19:15:06 +0000 |
|---|---|---|
| committer | twilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-09-30 19:15:06 +0000 |
| commit | 0dde50dfc3e37a0f382957b3a3b1b9e90a7ebf0f (patch) | |
| tree | 0d3c42e4e0c049616db3655b8aee0456400afe56 /configs | |
| parent | 0462e2f9adb65ec3f8b5a79f9ec4c843619c9900 (diff) | |
Merged revisions 221266 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009) | 32 lines
Merged revisions 221086 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines
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/
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@221304 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs')
| -rw-r--r-- | configs/sip.conf.sample | 3 |
1 files changed, 3 insertions, 0 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index a77745f0c..29c43affd 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -662,6 +662,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; (observed with Microsoft OCS). By default this option is ; off. +;constantssrc=yes ; Don't change the RTP SSRC when our media stream changes + ;----------------------------------------- REALTIME SUPPORT ------------------------ ; For additional information on ARA, the Asterisk Realtime Architecture, ; please read realtime.txt and extconfig.txt in the /doc directory of the @@ -867,6 +869,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; timerb ; qualifyfreq ; t38pt_usertpsource +; constantssrc ; contactpermit ; Limit what a host may register as (a neat trick ; contactdeny ; is to register at the same IP as a SIP provider, ; ; then call oneself, and get redirected to that |