[asterisk-dev] [Code Review] 2728: Allow the SIP_CODEC family of variables to specify more than one codec
Matt Jordan
reviewboard at asterisk.org
Mon Aug 19 18:17:05 CDT 2013
> On Aug. 16, 2013, 7:51 a.m., wdoekes wrote:
> > With nitpick mode on, I found a few issues. All minor.
Nothing wrong with nitpick mode :-)
> On Aug. 16, 2013, 7:51 a.m., wdoekes wrote:
> > /trunk/channels/chan_sip.c, line 7391
> > <https://reviewboard.asterisk.org/r/2728/diff/1/?file=43439#file43439line7391>
> >
> > Sounds more like a debug message to me. Especially if we're doing a NOTICE message for every codec added.
The "success" message variant was a NOTICE message before. I agree that it doesn't justify a NOTICE message (hey! Look! Dialplan worked!) - how about a VERBOSE level 4?
- Matt
-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2728/#review9418
-----------------------------------------------------------
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/20130819/973a414a/attachment-0001.htm>
More information about the asterisk-dev
mailing list