[asterisk-dev] [Code Review] 4000: res_pjsip_sdp_rtp.c: Fix native formats containing formats that were not negotiated.

Mark Michelson reviewboard at asterisk.org
Thu Sep 18 17:36:31 CDT 2014


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Ship it!


Ship It!

- Mark Michelson


On Sept. 18, 2014, 6:26 p.m., rmudgett wrote:
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> https://reviewboard.asterisk.org/r/4000/
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> (Updated Sept. 18, 2014, 6:26 p.m.)
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> 
> Review request for Asterisk Developers.
> 
> 
> Bugs: AFS-162
>     https://issues.asterisk.org/jira/browse/AFS-162
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> Repository: Asterisk
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> Description
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> Outgoing PJSIP calls can result in non-negotiated formats listed in the channel's native formats if video formats are listed in the endpoint's configuration.  The resulting call could then use a non-negotiated format resulting in one way audio.
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> * Simplified the update of session->req_caps in set_caps().  Why do something in five steps when only one is needed?
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> 
> Diffs
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>   /branches/13/res/res_pjsip_sdp_rtp.c 423446 
>   /branches/13/channels/chan_pjsip.c 423446 
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> Diff: https://reviewboard.asterisk.org/r/4000/diff/
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> Testing
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> Configured PJSIP endpoints with: allow=!all,h264,g722,h263,ulaw,h263p,alaw
> Called from D40 with g722 among other formats enabled to a Polycom that negotiates ulaw.
> Before the patch, Asterisk would send g722 frames to the Polycom.  The resulting call had one way audio because the Polycom does not understand g722.
> After the patch, Asterisk sends ulaw frames to the Polycom.
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> 
> Thanks,
> 
> rmudgett
> 
>

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