Merged revisions 221266 via svnmerge from

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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@221302 65c4cc65-6c06-0410-ace0-fbb531ad65f3