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
