[asterisk-dev] [Code Review] 2967: chan_sip: Fix RTCP port for SRFLX ICE candidates

[opticron]

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(Updated Nov. 1, 2013, 7:41 a.m.)


Status
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This change has been marked as submitted.


Review request for Asterisk Developers.


Bugs: ASTERISK-21383
    https://issues.asterisk.org/jira/browse/ASTERISK-21383


Repository: Asterisk


Description
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This corrects one-way audio between Asterisk and Chrome/jssip as a result of Asterisk inserting the incorrect RTCP port into RTCP SRFLX ICE candidates.


Diffs
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  branches/11/channels/chan_sip.c 402109 
  branches/11/include/asterisk/rtp_engine.h 402109 
  branches/11/res/res_rtp_asterisk.c 402109 

Diff: https://reviewboard.asterisk.org/r/2967/diff/


Testing
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Thanks,

opticron

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