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

[rmudgett]

> On Sept. 18, 2014, 4:35 p.m., Mark Michelson wrote:
> > /branches/13/res/res_pjsip_sdp_rtp.c, line 259
> > <https://reviewboard.asterisk.org/r/4000/diff/1/?file=67396#file67396line259>
> >
> >     Since joint only has formats of type media_type, would specifying media_type instead of AST_MEDIA_TYPE_UNKNOWN make more sense here?

It's a case of six of one and half a dozen of another.  It won't make any difference in this case since all formats will be appended anyway.  It's a little more efficient to use the constant since the function has to test if it isn't UNKNOWN and then test to see if it is the specified type.

I'll leave it as is unless there is a stronger argument for changing it.


- rmudgett


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On Sept. 18, 2014, 1:26 p.m., rmudgett wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4000/
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> 
> (Updated Sept. 18, 2014, 1:26 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Bugs: AFS-162
>     https://issues.asterisk.org/jira/browse/AFS-162
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> 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.
> 
> * Simplified the update of session->req_caps in set_caps().  Why do something in five steps when only one is needed?
> 
> 
> Diffs
> -----
> 
>   /branches/13/res/res_pjsip_sdp_rtp.c 423446 
>   /branches/13/channels/chan_pjsip.c 423446 
> 
> Diff: https://reviewboard.asterisk.org/r/4000/diff/
> 
> 
> Testing
> -------
> 
> 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.
> 
> 
> Thanks,
> 
> rmudgett
> 
>

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