[asterisk-dev] [Code Review] 4093: Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used

[Joshua Colp]

> On Nov. 5, 2014, 12:49 p.m., Joshua Colp wrote:
> > /tags/12.4.0/main/rtp_engine.c, lines 2012-2018
> > <https://reviewboard.asterisk.org/r/4093/diff/1/?file=68394#file68394line2012>
> >
> >     This is not compliant to the way L16 is supposed to be declared within SDP. The payload name is supposed to remain the same (L16) but the clock rate differs.
> >     
> >     Did you try that and it did not work, or did you just go for different payload names just because?
> 
> Frankie Chin wrote:
>     Hi Joshua, in my testing I did use the same payload name L16 for 48kHz and it worked. The reason I chose to use different payload names was because of a comment from Matt Jordan in ASTERISK-24274 on 03/Sep/14 9:25 AM, i.e. "You most likely wouldn't want to use the same MIME sub-type as 16 khz SLIN ("L16"). That would probably cause conflicts (or other badness) if Asterisk or another SIP endpoint offered L16."
>     
>     Is there a standard or RFC for defining the payload names? Please advise what I need to do next. Thanks.
>     
>

L16 is the correct name to use for it. It's 16-bit audio, just at a different rate. https://tools.ietf.org/html/rfc3551#section-4.5.11 Depends that name.


- Joshua


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On Oct. 31, 2014, 1:32 a.m., Frankie Chin wrote:
> 
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> 
> (Updated Oct. 31, 2014, 1:32 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
> 
>

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