Question about VoIP codecs
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Posted by Ramon F Herrera on February 20, 2007, 3:57 pm
Please log in for more thread options 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 << 5) ADPCM
64 (1 << 6) 16 bit Signed Linear PCM ----------------------- cowboy*CLI> show codec 97
1 (1 << 0) G.723.1
32 (1 << 5) ADPCM 64 (1 << 6) 16 bit Signed Linear PCM ------------------------- cowboy*CLI> show codec 0
Codec 0 not found
-------------------------- >From a Cisco:
voice class codec 1 codec preference 1 g729r8 bytes 40 codec preference 2 gsmfr bytes 132 codec preference 3 g723r63 bytes 96 codec preference 4 g726r16 bytes 80 | ||||||||||||||||
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Posted by Stela on February 22, 2007, 9:37 am
Please log in for more thread options > 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 << 5) ADPCM > 64 (1 << 6) 16 bit Signed Linear PCM > > ----------------------- > > cowboy*CLI> show codec 97 > 1 (1 << 0) G.723.1 > 32 (1 << 5) ADPCM > 64 (1 << 6) 16 bit Signed Linear PCM > > ------------------------- > > cowboy*CLI> show codec 0 > Codec 0 not found > > -------------------------- > > >From a Cisco:
>
> voice class codec 1 > codec preference 1 g729r8 bytes 40 > codec preference 2 gsmfr bytes 132 > codec preference 3 g723r63 bytes 96 > codec preference 4 g726r16 bytes 80 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. | ||||||||||||||||
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Posted by Rod Dorman on February 22, 2007, 2:16 pm
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>> This relates to Asterisk.
>> ... >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. Yes, Asterisk will transcode from one flavour to another. -- -- Rod -- rodd(at)polylogics(dot)com | ||||||||||||||||
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