[Asterisk-Dev] Frame.c/frame.h RTP payload formats
Martin Pycko
martinp at digium.com
Tue Feb 10 09:51:07 MST 2004
Only the printing of these codecs is wrong. Otherwise it's ok.
The function that prints tries to find the codec based on asterisk codec
numbering that's why you see UNKN and GSM .... I think there is a right
function now to display the right IANA codecs based in rtp.c
regards
Martin
On Tue, 10 Feb 2004, Olle E. Johansson wrote:
> I'm on thin ice here, so I want help to confirm if this is a bug.
>
> Check this output from SIP debug:
> ---------------
> m=audio 10020 RTP/AVP 0 8 3 18 101^M
> a=rtpmap:0 pcmu/8000^M
> a=rtpmap:8 pcma/8000^M
> a=rtpmap:3 gsm/8000^M
> a=rtpmap:18 g729/8000^M
> a=rtpmap:101 telephone-event/8000^M
> a=fmtp:101 0-15^M
> a=sendrecv^M
> -*-
> - 16 headers, 13 lines
> * Using latest SIP request as basis request
> * Found audio format UNKN
> * Found audio format ALAW
> * Found audio format UNKN
> * Found audio format UNKN
> * Found audio format UNKN
> * Found description format pcmu
> * Found description format pcma
> * Found description format gsm
> * Found description format g729
> * Found description format telephone-event
> --------
> Asterisk parses the number list wrong (m=audio) and the rtpmap right. The numbers in frame.h
> doesn't map to IANAs list on
> http://www.iana.org/assignments/rtp-parameters
>
> It's so different, so I don't know if they're the same. The calling phone (Snom 200) follows
> IANAs list. Is there a reason why the codec list in frame.h is so different. I would like
> to either correct that list or use another table for SDP parsing. Changing frame.h may
> cause other problems elsewhere in Asterisk...
>
> In this case, Asterisk finds GSM codec from Grandstream phone, but the phone doesn't
> indicate GSM support - or?
> -----
> m=audio 5004 RTP/AVP 0 8 4 18 2 15^M
> a=rtpmap:0 PCMU/8000^M
> a=rtpmap:8 PCMA/8000^M
> a=rtpmap:4 G723/8000^M
> a=rtpmap:18 G729/8000^M
> a=rtpmap:2 G726-32/8000^M
> a=rtpmap:15 G728/8000^M
> a=ptime:20^M
> -*-
> - 13 headers, 13 lines
> * Using latest SIP request as basis request
> * Found audio format UNKN
> * Found audio format ALAW
> * Found audio format ULAW
> * Found audio format UNKN
> * Found audio format GSM
> * Found audio format UNKN
> * Found description format PCMU
> * Found description format PCMA
> * Found description format G723
> * Found description format G729
> * Found description format G726-32
> * Found description format G728
>
> --------
> (Side comment: RTP Payload 2 is "reserved" on IANAs site, but GS uses that for G726-32)
>
> Comments, please?
>
> /O
>
>
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
More information about the asterisk-dev
mailing list