> On Nov. 5, 2014, 6:49 a.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. > > > > Joshua Colp wrote: > 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.
Mea culpa. I will say that I'm still not sure how useful this patch is. Transmitting 192kHz SLIN audio over the wire is ... probably not a good idea. What is the actual use case for this patch? - Matt ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4093/#review13684 ----------------------------------------------------------- On Oct. 30, 2014, 8:32 p.m., Frankie Chin wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/4093/ > ----------------------------------------------------------- > > (Updated Oct. 30, 2014, 8:32 p.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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