[asterisk-dev] [Code Review] Change SSRC when a peer sends a re-invite

[Terry Wilson]

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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/374/
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(Updated 2009-09-23 13:33:03.990229)


Review request for Asterisk Developers and Russell Bryant.


Summary
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Asterisk doesn't honor marker bit when reinvited to already-bridged RTP streams, resulting in far-end stack discarding packets with "old" timestamps that are actually 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.


Diffs
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  /branches/1.4/channels/chan_sip.c 219719 
  /branches/1.4/include/asterisk/rtp.h 219719 
  /branches/1.4/main/rtp.c 219719 

Diff: https://reviewboard.asterisk.org/r/374/diff


Testing
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Verified that the control message is queued on reinvite (via hold on a polycom phone), except when constantssrc=true in either global or peer config.


Thanks,

Terry