[asterisk-dev] [Code Review] 2728: Allow the SIP_CODEC family of variables to specify more than one codec

wdoekes reviewboard at asterisk.org
Fri Aug 16 02:51:03 CDT 2013


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With nitpick mode on, I found a few issues. All minor.


/trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/2728/#comment18486>

    Why remove the docs? Updating it is better =)



/trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/2728/#comment18487>

    Call this original_jointcaps. We see that it gets copied. The reason why is more important.



/trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/2728/#comment18484>

    I know this was here before you came, but:
    
    according to the docs, you should check the return value (NULL), not fmt.id.
    
    (And we can drop the ast_format_clear() above.)



/trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/2728/#comment18483>

    I know this was here before you came, but:
    
    we read a couple of different variables, but claim that it was "${SIP_CODEC}". Perhaps replace with "${SIP_CODEC*}" to avoid confusing the reader.



/trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/2728/#comment18482>

    Sounds more like a debug message to me. Especially if we're doing a NOTICE message for every codec added.


- wdoekes


On Aug. 1, 2013, 12:42 a.m., Matt Jordan wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2728/
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> 
> (Updated Aug. 1, 2013, 12:42 a.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Bugs: ASTERISK-21976
>     https://issues.asterisk.org/jira/browse/ASTERISK-21976
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> {quote}
> For video calls, we would like to set the codecs in the dialplan using 
> SIP_CODEC. However, if SIP_CODEC is set, all codecs except the ONE set are disallowed and thus either audio or video is available.
> Attached is a patch for 11.4 that allows SIP_CODEC to contain a list of codecs , e.g. "gsm,h264".
> {quote}
> 
> As an aside, chan_pjsip has an analogous dialplan function "PJSIP_MEDIA_OFFER". While this doesn't allow for setting multiple codecs, it does handle multiple media types, as you can specify both video or audio for the codec you want to apply - hence I didn't port this patch/feature over to chan_pjsip. If we think it needs it, it would be reasonably easy to do so.
> 
> 
> Diffs
> -----
> 
>   /trunk/channels/chan_sip.c 395907 
> 
> Diff: https://reviewboard.asterisk.org/r/2728/diff/
> 
> 
> Testing
> -------
> 
> 
> Thanks,
> 
> Matt Jordan
> 
>

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