Ok, Thanks Joshua. I'll try and make it work. On Wed, 28 Aug 2019 at 17:17, Joshua C. Colp <jc...@digium.com> wrote:
> On Wed, Aug 28, 2019, at 8:36 AM, Mohit Dhiman wrote: > > Ok, now I am a little confused here because when Asterisk initiate a > > SIP transaction (INVITE) > > and it generates SDP, then without any prior knowledge how > > ref_format_attr_xyz.c can set the > > SDP parameters which were read by codec_opus.so from codecs.conf file. > > Ah yes, I forgot about that functionality. The way it works is that it > constructs an fmtp string which is passed to the ast_format_parse_sdp_fmtp > function, which creates the appropriate format. This is then cached. It > also stores the configuration on the format itself for later retrieval if > need be. > > If you are doing something similar and it isn't working then I'd suggest > following the path of things and identifying where it goes wrong. > > -- > Joshua C. Colp > Digium - A Sangoma Company | 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
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