[OpenSER-Users] INVITE relayed with missing bytes

[Papadopoulos Georgios]

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 at 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 at 213.5.1.6";branch=z9hG4bK1594C
1AB9.
Remote-Party-ID: "GEOP Papadopoul"
<sip:2116872933 at 213.5.1.6>;party=calling;screen=no;privacy=off.
From: "GEOP Papadopoul" <sip:2116872933 at 213.5.1.6>;tag=237E8130-1821.
To: <sip:1012118204501 at 213.5.43.4>.
Date: Tue, 23 Oct 2007 10:27:46 GMT.
Call-ID: 6E2DF805-808911DC-AA258232-F1669B65 at 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 at 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 at 213.5.1.6";branch=z9hG4bK1594C
1AB9;rport=57665.
From: "GEOP Papadopoul" <sip:2116872933 at 213.5.1.6>;tag=237E8130-1821.
To: <sip:1012118204501 at 213.5.43.4>.
Call-ID: 6E2DF805-808911DC-AA258232-F1669B65 at 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 at 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 at 213.5.1.6";branch=
z9hG4bK1594C1AB9.
Remote-Party-ID: "GEOP Papadopoul"
<sip:2116872933 at 213.5.1.6>;party=calling;screen=no;privacy=off.
From: "GEOP Papadopoul" <sip:2116872933 at 213.5.1.6>;tag=237E8130-1821.
To: <sip:1012118204501 at 213.5.43.4>.
Date: Tue, 23 Oct 2007 10:27:46 GMT.
Call-ID: 6E2DF805-808911DC-AA258232-F1669B65 at 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 at 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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