[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
-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2728/#review9418
-----------------------------------------------------------
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:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2728/
> -----------------------------------------------------------
>
> (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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20130816/2eddd4ce/attachment-0001.htm>
More information about the asterisk-dev
mailing list