hi,
When calling from Sipura to softphone (either SJphone or the sip client in my e61 nokia phone), with the Sipura set to Prefered codec: g723 Use prefered codec only: no and the softphone only supporting g711, I get this reply from the softphone:
U xxxx:11474 -> xxxxx:5060 SIP/2.0 488 Not Acceptable Here. Via: SIP/2.0/UDP xxxx;branch=z9hG4bK23e.248c6cd1.1,SIP/2.0/UDP xxxx:10001;branch=z9hG4bK-3e8ce192;rport=10001. To: sip:1@test.com;tag=ujk6mpqhbhhc6kj68irr. From: sipura line1 sip:2@test.com;tag=d0404120874d710eo0. Call-ID: e5052808-575f7746@192.168.1.52. CSeq: 102 INVITE. Warning: 304 192.168.1.3 Media type not available. Content-Length: 0.
the invite was:
U xxxx:5060 -> xxxx:11474 INVITE sip:1@test.co SIP/2.0. Record-Route: sip:xxxx;lr=on;ftag=d0404120874d710eo0. Via: SIP/2.0/UDP xxxx:branch=z9hG4bK23e.248c6cd1.1. Via: SIP/2.0/UDP xxxx:10001;branch=z9hG4bK-3e8ce192;rport=10001. From: sipura line1 sip:2@test.com;tag=d0404120874d710eo0. To: sip:1@test.com. Call-ID: e5052808-575f7746@192.168.1.52. CSeq: 102 INVITE. Max-Forwards: 69. Contact: sipura line1 sip:2@xxx:10001. Expires: 240. User-Agent: Sipura/SPA2002-3.1.5. Content-Length: 419. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 36143 36143 IN IP4 xxxx. s=-. c=IN IP4 xxxxxx. t=0 0. m=audio 62238 RTP/AVP 4 0 2 8 18 96 97 98 100 101. a=rtpmap:4 G723/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. 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=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv.
If I set prefered codec to g711 on the Sipura it works normally. What is the best way to handle this on the sip proxy?
Thanks,
Richard
This is not handled at the sip proxy...it's how codec negotiation works within SIP world: it's done at the end-points. You would need a media gateway (such as *) acting as a bridge transcoding the RTP streams from one codec to another.
Samuel.
2006/10/2, Richard Bennett richard.bennett@skynet.be:
hi,
When calling from Sipura to softphone (either SJphone or the sip client in my e61 nokia phone), with the Sipura set to Prefered codec: g723 Use prefered codec only: no and the softphone only supporting g711, I get this reply from the softphone:
U xxxx:11474 -> xxxxx:5060 SIP/2.0 488 Not Acceptable Here. Via: SIP/2.0/UDP xxxx;branch=z9hG4bK23e.248c6cd1.1,SIP/2.0/UDP xxxx:10001;branch=z9hG4bK-3e8ce192;rport=10001. To: sip:1@test.com;tag=ujk6mpqhbhhc6kj68irr. From: sipura line1 sip:2@test.com;tag=d0404120874d710eo0. Call-ID: e5052808-575f7746@192.168.1.52. CSeq: 102 INVITE. Warning: 304 192.168.1.3 Media type not available. Content-Length: 0.
the invite was:
U xxxx:5060 -> xxxx:11474 INVITE sip:1@test.co SIP/2.0. Record-Route: sip:xxxx;lr=on;ftag=d0404120874d710eo0. Via: SIP/2.0/UDP xxxx:branch=z9hG4bK23e.248c6cd1.1. Via: SIP/2.0/UDP xxxx:10001;branch=z9hG4bK-3e8ce192;rport=10001. From: sipura line1 sip:2@test.com;tag=d0404120874d710eo0. To: sip:1@test.com. Call-ID: e5052808-575f7746@192.168.1.52. CSeq: 102 INVITE. Max-Forwards: 69. Contact: sipura line1 sip:2@xxx:10001. Expires: 240. User-Agent: Sipura/SPA2002-3.1.5. Content-Length: 419. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 36143 36143 IN IP4 xxxx. s=-. c=IN IP4 xxxxxx. t=0 0. m=audio 62238 RTP/AVP 4 0 2 8 18 96 97 98 100 101. a=rtpmap:4 G723/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. 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=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv.
If I set prefered codec to g711 on the Sipura it works normally. What is the best way to handle this on the sip proxy?
Thanks,
Richard
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
On Monday 02 October 2006 10:11, samuel wrote:
This is not handled at the sip proxy...it's how codec negotiation works within SIP world: it's done at the end-points. You would need a media gateway (such as *) acting as a bridge transcoding the RTP streams from one codec to another.
Ok, thanks for the info. I thought it might be possible to send a re-invite to initiate another codec, or to re-write the invite on 488 failure, changing the order of the supported codecs to put g711 in the first place. Frankly I'm surprised the softphones fail at all, as g711 is in the list of supported codecs sent by the sipura's invite, just not in the first place... I'm finding out I still have a lot to learn here though...
Thanks,
Richard
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Richard Bennett -
samuel