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? -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
