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
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