On Fri, Mar 24, 2017 at 7:43 AM, Jean Aunis <jean.au...@prescom.fr> wrote: > Hello, > > I need to add support for a new audio format in Asterisk. It will be > actually a very limited support : I just need to take the audio from one > side and transmit it to the other side. No decoding / transcoding will be > involved. One side will be a DAHDI endpoint, the other one a SIP endpoint.
What's the new audio format, if you don't mind me asking? (/me wonders if it's CLEARMODE) > Let's put aside for the moment the modifications which may have to be done > on the channels drivers to handle the codec negotiation. If I focus on > Asterisk's core and RTP handling, it seems I could achieve this simply by > registering a new format in codec_builtin.c. Two questions then : > > - firstly, is this really the only thing to do, or am I missing something ? I forget every time I've had to do this, and end up relearning it every time. Here are some other places that you'll need to modify: main/rtp_engine.c: Some work here to work with SDP based protocols, like SIP. include/asterisk/format_cache.h: To declare the new format in a header file. main/format_cache.c: Also need to define the format here. channels/chan_dahdi.c: It needs modified for support for any new codecs that it needs to work with. > - secondly, is there a more "pluggable" way to do this ? Maybe with a shared > object which would be loaded on startup ? Good question. I don't think that there's a great way to add a completely new codec (not previously described in Asterisk at all) without patching Asterisk. Perhaps Josh Colp or Kevin Harwell can correct me if I'm wrong :-) -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _____________________________________________________________________ -- 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
