Guys, just tried asterisk13 and added seanbrights' patch for opus.
incoming INVITE has fmtp ------> maxplaybackrate=8000;sprop-maxcapturerate=8000 but INVITE to my registered peer is ----------> maxplaybackrate=48000;sprop-maxcapturerate=48000 it should not even have to load up the opus patch because it is just a passthrough. have you changed anything to chan_sip.c to make this work? Kelvin Chua On Sat, Jun 27, 2015 at 7:26 AM, Kelvin Chua <kel...@gmail.com> wrote: > i verified parse_sdp is doing its job correctly and stores it in struct. > but after going back to chan_sip somehow somewhere everything resets before > generate_sdp. maybe because i am working on ast12, i'm going to try 13 > On Jun 26, 2015 5:49 PM, "Joshua Colp" <jc...@digium.com> wrote: > >> Kelvin Chua wrote: >> >>> Just an experiment I am doing, correct me if I am wrong >>> >>> If I receive an INVITE with fmtp from a peer, it won't be used to build >>> the INVITE to the egress right? >>> >>> What will happen is, codecs.conf will be checked for the parameters and >>> use that to build the INVITE. >>> >>> Is there any function I can use to get away from this behavior and act >>> like a proxy and just copy the fmtp from the ingress? >>> >> >> As Alexander mentioned there has to be a specific handler for each codec >> in order to parse/store/create the specific attributes internally. It's not >> done for every codec. Asterisk also has to be aware of the codec. This is a >> bit easier in 13+, but may be possible in earlier versions depending on the >> amount of storage required. >> >> Cheers, >> >> -- >> Joshua Colp >> Digium, Inc. | Senior Software Developer >> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US >> Check us out at: www.digium.com & www.asterisk.org >> >> -- >> _____________________________________________________________________ >> -- 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 >> >
-- _____________________________________________________________________ -- 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