[asterisk-dev] Adding an audio format in Asterisk

[Alexander Traud]

> adding a new line "CODEC_REGISTER_AND_CACHE" would magically do
> everything required, but now I see there is some more work to do

At <https://github.com/traud?tab=repositories&q=asterisk> you find several examples. If you do not need transcoding and just pass-through, the patch file for GSM-EFR might be the right starting point. If your audio codec is variable bit-rate (VBR) or you want to support just 20ms, simply go for ".smooth = 0". In case of constant bit-rate (CBR), you change the two "31" into the amount of bytes you see in Wireshark.

This does the trick for the channel drivers based on SIP (chan_sip and res_pjsip). Please, drop the mailing list your feedback what else was required for chan_dahdi.