> 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?
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.
- Frankie
-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/4093/#review13684
-----------------------------------------------------------
On Oct. 31, 2014, 1:32 a.m., Frankie Chin wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4093/
> -----------------------------------------------------------
>
> (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
>
>
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev
