[asterisk-dev] Transcoding: Codec 2, iLBC 20, SILK, GSM-EFR, AMR(-WB)

[Alexander Traud]

> If I get a "codec2" stream, which rate (and/or other parameters) are used?

Faced the same question, when I started with Codec 2. I am glad, somebody is
interested. I am going to add this to the Read Me in my GitHub repository:

Currently, FreeSWITCH and CSipSimple support only the first Codec 2 release,
which was 2400 only. Wrappers for Asterisk 1.8 and Asterisk 11 exist, which
do the same. Those wrapper are included (but unmaintained) in the Codec 2
sources. I used those wrappers as starting point and ported it over to
Asterisk 13. Consequently, still with mode 2400.

Because there is neither an IANA MIME media-type registration nor a IETF RFC
for the negotiation within the Session Description Protocol (SDP) yet,
because of this, I would have to come up with my own SDP negotiation to
support those other Codec 2 modes/rates. Any suggestions are warmly welcome.

I am curious to try mode 3200 and especially mode 1600, because the latter
uses a 40 msec packetization time as default. And how that compares with AMR
(Mode 0 = MR475 = 4750 bit/s) in sRTP.