Hello: I'm using a NATed UA to terminate a call in the PSTN using a Cisco 5300. I want to use the RTP relay capability. In the script, I wrote:
...
force_rtp_proxy();
rewritehost("200.60.XXX.XXX");
forward("200.60.XXX.XXX", 5060);
...
Where 200.60.XXX.XXX is the GTW IP.
I get an error in the log:
"/usr/sbin/ser[7039]: ERROR: extract_mediaip: no 'c=' in SDP"
The SIP Message sent to de GTW has a wrong content-length header, so the GTW rejects it. Here I copy the message:
INVITE sip:2345674284618@200.60.30.24 SIP/2.0
Record-Route: sip:2345674284618@200.32.43.55;ftag=673501644;lr=on
Via: SIP/2.0/UDP 200.32.43.55;branch=0
Via: SIP/2.0/UDP 200.68.54.46:5060;rport=25127;branch=z9hG4bKFE5D22D6B6C4407884
2396149F64C3DE
From: 1001 sip:1001@200.32.43.55;tag=673501644
To: sip:2345674284618@200.32.43.55
Contact: sip:1001@200.68.54.46:25127
Call-ID: 107D459F-7402-480D-988A-FC87455E1E5D@192.168.0.15
CSeq: 31862 INVITE
Max-Forwards: 69
Content-Type: application/sdp
voip-gtw04#
User-Agent: X-Lite build 1101
Content-Length: 315296
v=0
o=1001 23178458 23178458 IN IP4 200.68.54.46
s=X-Lite
c=IN IP4 200.32.43.55
t=0 0
m=audio 35052 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=direction:active
I would appreciate any help about this.
Best Regards,
What SER version are you using? (Unless you use stable CVS branch, that's a very likely explanation for the problem you experience.)
-jiri
At 08:19 PM 1/22/2004, Federico R. SANCHEZ Y NAPAL wrote:
Hello: I'm using a NATed UA to terminate a call in the PSTN using a Cisco 5300. I want to use the RTP relay capability. In the script, I wrote:
... force_rtp_proxy(); rewritehost("200.60.XXX.XXX"); forward("200.60.XXX.XXX", 5060); ...
Where 200.60.XXX.XXX is the GTW IP.
I get an error in the log: "/usr/sbin/ser[7039]: ERROR: extract_mediaip: no 'c=' in SDP"
The SIP Message sent to de GTW has a wrong content-length header, so the GTW rejects it. Here I copy the message:
INVITE sip:2345674284618@200.60.30.24 SIP/2.0 Record-Route: sip:2345674284618@200.32.43.55;ftag=673501644;lr=on Via: SIP/2.0/UDP 200.32.43.55;branch=0 Via: SIP/2.0/UDP 200.68.54.46:5060;rport=25127;branch=z9hG4bKFE5D22D6B6C4407884 2396149F64C3DE From: 1001 sip:1001@200.32.43.55;tag=673501644 To: sip:2345674284618@200.32.43.55 Contact: sip:1001@200.68.54.46:25127 Call-ID: 107D459F-7402-480D-988A-FC87455E1E5D@192.168.0.15 CSeq: 31862 INVITE Max-Forwards: 69 Content-Type: application/sdp voip-gtw04# User-Agent: X-Lite build 1101 Content-Length: 315296 v=0 o=1001 23178458 23178458 IN IP4 200.68.54.46 s=X-Lite c=IN IP4 200.32.43.55 t=0 0 m=audio 35052 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=direction:active I would appreciate any help about this. Best Regards,
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