On Thu, Jun 8, 2017, at 11:35 AM, Stephen Davies wrote: > Hi, > > Reviewing rtp_engine.c it appears that we only support telephone-event > rtp > with a sample rate of 8000? > > JSSIP using Opus offers Opus as "opus/48000/2" and then (by necessity, I > think), telephone-event/48000. >
<snip> > > In rtp_engine.c there is only: > > set_next_mime_type(NULL, AST_RTP_DTMF, "audio", "telephone-event", 8000); > > Has this come up before? Not that I can remember. > Can any other developer point me as to where I'd need to look to try to > add > 48000 too? Things really aren't written at all to handle this case. You'd like need to change code in rtp_engine.c to add it as 48000, and then also in the channel driver (or res_pjsip_sdp_rtp in the case of PJSIP) to handle it. You'd also need to add logic to determine when exactly to use it. It's uncharted territory for the DTMF support in RTP in Asterisk. -- 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
