Question about VoIP codecs

This relates to Asterisk.

I have three views of the codec negotiation protocol (see below) and I can't reconcile them. The first one, from "sip debug peer" lists numbers such as '96' and '97'. How does that match "sip show codec 96" or "sip show codec 97"? Even stranger is codec 0, since "show codec 0" gives an error. The last entry (Cisco) includes a given codec with varying lengths(?).

Other than reading the RFCs, is there any documentation on this? Perhaps a kind soul can explain this?

TIA,

-Ramon F Herrera

--------------- sip debug peer: v=0 o=- 2649501 2649501 IN IP4 71.41.206.193 s=- c=IN IP4 71.41.206.193 t=0 0 m=audio 36388 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv

----------------------

cowboy*CLI> show codec 96

32 (1
Reply to
Ramon F Herrera
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Sorry, for asking, do you use a software transcoder, because here I can see G723 and G726 and G729 codecs. I realy don't know if the Asterisk software does transcoding.

Reply to
Stela

Yes, Asterisk will transcode from one flavour to another.

Reply to
Rod Dorman

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