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@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@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@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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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@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=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@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=z9hG4bK1594C1AB9;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@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@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
Disclaimer
The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. _______________________________________________ Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
El Martes, 23 de Octubre de 2007, Ovidiu Sas escribió:
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).
But then he should see the second UDP packet with ngrep, shouldn't he?
-
Iñaki Baz Castillo -
Ovidiu Sas -
Papadopoulos Georgios