> 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. -- _____________________________________________________________________ -- 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