[asterisk-dev] [Code Review] 3596: PJSIP: Recalculate translation paths on raw format changes

[opticron]

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https://reviewboard.asterisk.org/r/3596/
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Review request for Asterisk Developers.


Bugs: AFS-63
    https://issues.asterisk.org/jira/browse/AFS-63


Repository: Asterisk


Description
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Currently, there are situations that can occur when using chan_pjsip and certain dialplan applications (notably ChanSpy()) that can cause the pjsip spyer channel to get no audio with scrolling warnings about format mismatches. This is caused by a failure to update translation paths on a mid-call raw format update since the requested read and write formats have not changed. This change introduces two new helper functions for setting a channel's raw formats that allow translation paths to be recalculated if necessary.


Diffs
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  branches/12/res/res_pjsip_sdp_rtp.c 415334 
  branches/12/main/channel.c 415334 
  branches/12/include/asterisk/channel.h 415334 

Diff: https://reviewboard.asterisk.org/r/3596/diff/


Testing
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Basic call tests.


Thanks,

opticron

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