Hi Ovidiu,
So you are saying that despite ngrep does not show it, the whole packet
is correctly sent?
thank you
George
-----Original Message-----
From: Ovidiu Sas [mailto:sip.nslu@gmail.com]
Sent: Tuesday, October 23, 2007 5:23 PM
To: Papadopoulos Georgios
Cc: users(a)openser.org
Subject: Re: [OpenSER-Users] INVITE relayed with missing bytes
Hi Georgios,
It seems that your packet is to big and it was fragmented
(udp fragmentation).
OpenSER is able to deal with fragmented packets (both sending
and receiving).
Regards,
Ovidiu Sas
On 10/23/07, Papadopoulos Georgios <geop(a)altectelecoms.gr> wrote:
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=z9
hG4bK1594C1AB9.
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;Dura
tion=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=z9
hG4bK1594C1AB9;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;Dura
tion=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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