[asterisk-dev] [Code Review] Chan_sip: Voice frame dropped for every early media audio call

[Olle E Johansson]

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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1186/
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(Updated 2011-04-18 05:55:30.718934)


Review request for Asterisk Developers.


Changes
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Typos. Monday.


Summary (updated)
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The code for checking T.38 in sip_write accidentally drops one frame in situations where an audio frame forces early media . If you compare with the video code below this part of the code, the frame is not dropped even though we add an 183 message. It's not a big issue, but nevertheless, frames are frames and should be treated with love and care.

Moved the T.38 check out of the lock - maybe that's wrong?


This addresses bug 19312.
    https://issues.asterisk.org/view.php?id=19312


Diffs
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  /branches/1.4/channels/chan_sip.c 313187 

Diff: https://reviewboard.asterisk.org/r/1186/diff


Testing
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Can't test with T.38 - only with audio. It works.


Thanks,

Olle E

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