[Users] how to handle "488 Not Acceptable Here"

[samuel]

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 at 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 at test.com>;tag=ujk6mpqhbhhc6kj68irr.
> From: sipura line1 <sip:2 at test.com>;tag=d0404120874d710eo0.
> Call-ID: e5052808-575f7746 at 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 at 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 at test.com>;tag=d0404120874d710eo0.
> To: <sip:1 at test.com>.
> Call-ID: e5052808-575f7746 at 192.168.1.52.
> CSeq: 102 INVITE.
> Max-Forwards: 69.
> Contact: sipura  line1 <sip:2 at 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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