oh that figures. i am working on ast12 and never bothered trying 13. i'm playing around with opus and so frustrated that it does not adapt to the incoming sdp. even if after parse_sdp, it just resets everything on generate_sdp. i'll try 13 later, thanks for the tip On Jun 26, 2015 5:30 PM, "Alexander Traud" <[email protected]> wrote:
> > If I receive an INVITE with fmtp from a peer, it won't be used to build > the > > INVITE to the egress right? > > With Asterisk 13/chan_sip, it is possible to copy over the fmtp - even 1:1 > - > I do this here with AMR-WB. I created a res/res_format_attr_ and adopted > format_parse_sdp_fmtp/format_generate_sdp_fmtp, just like the Opus sample. > Thanks to Asterisk 13, the selected codec of the ingress gets the first > priority for the egress. > > > Is there any function [to] act like a proxy? > > Mhm. I was not able to do that because here, Asterisk removed unknown > codecs > and adds its allowed ones after the one selected for the ingress. So the > lines m=, a/v/t=, and their order are going to be different. But as stated, > copying over (one!) fmtp per format is possible. By the way, which > codec/format are you about? > > > > -- > _____________________________________________________________________ > -- 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 >
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