23 Oct
2007
23 Oct
'07
12:36 p.m.
Hello all,
I am noticing that OpenSER relays an INVITE and the packet that is sent
out is clipped. You can see in the following packets that the forwarded
packet is missing a few bytes from the SDP part. Counting the bytes
shows that the outgoing packet is always 1512 bytes. Is this something
that has do with OpenSER (module parameter, compile options) or is it OS
related?
thank you for any help
George
U 2007/10/23 13:27:46.165232 213.5.1.6:57665 -> 213.5.43.4:5060
INVITE sip:1012118204501@213.5.43.4:5060 SIP/2.0.
Via: SIP/2.0/UDP
213.5.1.6:5060;x-route-tag="cid:CID-ACN-3@213.5.1.6";branch=z9hG4bK1594C
1AB9.
Remote-Party-ID: "GEOP Papadopoul"
<sip:2116872933@213.5.1.6>;party=calling;screen=no;privacy=off.
From: "GEOP Papadopoul" <sip:2116872933@213.5.1.6>;tag=237E8130-1821.
To: <sip:1012118204501@213.5.43.4>.
Date: Tue, 23 Oct 2007 10:27:46 GMT.
Call-ID: 6E2DF805-808911DC-AA258232-F1669B65(a)213.5.1.6.
Supported: 100rel,timer,resource-priority,replaces.
Min-SE: 1800.
Cisco-Guid: 1848010793-2156466652-3218142496-4228086245.
User-Agent: Cisco-SIPGateway/IOS-12.x.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY, INFO, REGISTER.
CSeq: 101 INVITE.
Max-Forwards: 70.
Timestamp: 1193135266.
Contact: <sip:2116872933@213.5.1.6:5060>.
Call-Info:
<sip:213.5.1.6:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
.
Expires: 180.
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Disposition: session;handling=required.
Content-Length: 290.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 5211 453 IN IP4 213.5.1.6.
s=SIP Call.
c=IN IP4 213.5.1.6.
t=0 0.
m=audio 16454 RTP/AVP 18 0 8 100.
c=IN IP4 213.5.1.6.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=yes.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:100 X-NSE/8000.
a=fmtp:100 192-194.
U 2007/10/23 13:27:46.173826 213.5.43.4:5060 -> 213.5.1.6:57665
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP
213.5.1.6:5060;x-route-tag="cid:CID-ACN-3@213.5.1.6";branch=z9hG4bK1594C
1AB9;rport=57665.
From: "GEOP Papadopoul" <sip:2116872933@213.5.1.6>;tag=237E8130-1821.
To: <sip:1012118204501@213.5.43.4>.
Call-ID: 6E2DF805-808911DC-AA258232-F1669B65(a)213.5.1.6.
CSeq: 101 INVITE.
Server: Altec Telecoms SIP Proxy.
Content-Length: 0.
.
U 2007/10/23 13:27:46.174241 213.5.43.4:5060 -> 213.5.17.226:5060
INVITE sip:demo1@192.168.1.39:5060 SIP/2.0.
Record-Route: <sip:213.5.43.4;lr=on;ftag=237E8130-1821>.
Via: SIP/2.0/UDP 213.5.43.4;branch=z9hG4bK4acd.5092baa7.0.
Via: SIP/2.0/UDP
213.5.1.6:5060;rport=57665;x-route-tag="cid:CID-ACN-3@213.5.1.6";branch=
z9hG4bK1594C1AB9.
Remote-Party-ID: "GEOP Papadopoul"
<sip:2116872933@213.5.1.6>;party=calling;screen=no;privacy=off.
From: "GEOP Papadopoul" <sip:2116872933@213.5.1.6>;tag=237E8130-1821.
To: <sip:1012118204501@213.5.43.4>.
Date: Tue, 23 Oct 2007 10:27:46 GMT.
Call-ID: 6E2DF805-808911DC-AA258232-F1669B65(a)213.5.1.6.
Supported: 100rel,timer,resource-priority,replaces.
Min-SE: 1800.
Cisco-Guid: 1848010793-2156466652-3218142496-4228086245.
User-Agent: Cisco-SIPGateway/IOS-12.x.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY, INFO, REGISTER.
CSeq: 101 INVITE.
Max-Forwards: 10.
Timestamp: 1193135266.
Contact: <sip:2116872933@213.5.1.6:57665>.
Call-Info:
<sip:213.5.1.6:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
.
Expires: 180.
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Disposition: session;handling=required.
Content-Length: 290.
P-hint: NATed client request.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 5211 453 IN IP4 213.5.1.6.
s=SIP Call.
c=IN IP4 213.5.1.6.
t=0 0.
m=audio 16454 RTP/AVP 18 0 8 100.
c=IN IP4 213.5.1.6.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=yes.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:100 X-NSE/8000.
a=fmtp:100
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