[asterisk-users] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled

[Daniel Pocock]

On 31/03/13 23:43, Joshua Colp wrote:
> Daniel Pocock wrote:
>> I'm trying to call from DruCall to Asterisk and I get this error:
>>
>> WARNING[11021]: chan_sip.c:8687 process_sdp: Error in codec string 'F
>> 103 104 111 0 8 107 106 105 13 126'
>>    == Problem setting up ssl connection:
>> error:00000000:lib(0):func(0):reason(0)
>>
>>
>> I'm guessing my Asterisk is too old (it is 1.8 from Debian).  Can you
>> confirm which version is needed to parse a media descriptor with SAVPF?
>>   Do I need to upgrade all the way to v11 with WebRTC support, or was
>> avpf support added in some intermediate version?
> 
> Asterisk 1.8 does not have any knowledge of AVPF, and since it's a new
> feature it was only added to Asterisk 11. You could try to backport the
> changes but chan_sip has changed quite a bit, so it could be rough.


Thanks for the fast reply.  I agree backporting full support for AVPF
would not be justified for many use cases (including my own).  What I
was more curious about is whether the F can be tolerated (in other
words, ignored or silently removed), as described here:

http://www.ietf.org/mail-archive/web/rtcweb/current/msg01145.html
"1) RTCWEB end-point will always signal AVPF or SAVPF. I signalling
gateway to legacy will change that by removing the F to AVP or SAVP."

and whether such behavior is possible even without setting avpf=yes on a
per-peer basis?