[asterisk-dev] [Code Review] 4093: Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
Joshua Colp
reviewboard at asterisk.org
Thu Nov 20 11:32:59 CST 2014
-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/4093/#review13831
-----------------------------------------------------------
Ship it!
Ship It!
- Joshua Colp
On Nov. 17, 2014, 2:51 a.m., Frankie Chin wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4093/
> -----------------------------------------------------------
>
> (Updated Nov. 17, 2014, 2:51 a.m.)
>
>
> Review request for Asterisk Developers.
>
>
> Bugs: ASTERISK-24274
> https://issues.asterisk.org/jira/browse/ASTERISK-24274
>
>
> Repository: Asterisk
>
>
> Description
> -------
>
> Currently SLIN12, SLIN24, SLIN32, SLIN44, SLIN48, SLIN96 and SLIN192 are found not working with SIP. The following error will be thrown if one of those codecs is used: chan_sip.c:10718 process_sdp: No compatible codecs, not accepting this offer!
>
> What I think the issue is that the codec format isn't being included in the SDP media attributes when one of those codecs is used. Please refer to ASTERISK-24274 for more details. This change updates the main/rtp_engine.c and main/frame.c to ensure all these codecs are supported.
>
> Note: SLIN and SLIN16 are working fine.
>
>
> Diffs
> -----
>
> /tags/12.4.0/main/rtp_engine.c 425756
> /tags/12.4.0/main/frame.c 425756
>
> Diff: https://reviewboard.asterisk.org/r/4093/diff/
>
>
> Testing
> -------
>
> Specified SLIN48 codec in sip.conf of two Asterisk servers. Used AMI to originate a call from Server A to Server B and then put Server B into a conference hosted in Server A. The above mentioned error was no longer reported and the conference was working as expected.
>
>
> Thanks,
>
> Frankie Chin
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20141120/6460df45/attachment.html>
More information about the asterisk-dev
mailing list