[OpenSIPS-Users] help configuring Opensips as proxy
Bogdan-Andrei Iancu
bogdan at opensips.org
Wed Feb 29 19:20:09 CET 2012
Hi there,
My guess is the fault is in the sipp script - the ACK is not properly
generated : instead of using the route set information from the 200 OK
(the contact and RR URIs), it is simply sent with the same RURI as the
INVITE - this is of course bogus.
If you want, I can send you an working SIPP UAC file.
Regards,
Bogdan
On 02/29/2012 08:02 PM, dyatsin wrote:
> Hi,
> I am trying to get familiar with Opensips (1.7.1) so I set up two boxes with
> SIPp, OpenSIPS, and Asterisk:
>
> box 1 (ip: 192.168.1.57) running SIPp and Opensips
> box 2 (ip: 192.168.1.121) running Asterisk
>
> I am trying to get OpenSIPS to run as a proxy between SIPp and Asterisk, but
> this is the problem I have so right now:
>
>
> SIPp OpenSIPS Asterisk
> | | |
> | INVITE | |
> |---------------->| |
> | | INVITE |
> | |---------------->|
> | | |
> | | 100 Trying |
> | |<----------------|
> | 100 Trying | |
> |<----------------| |
> | | |
> | | 200 OK |
> | |<----------------|
> | | |
> | 200 OK | |
> |<----------------| |
> | | |
> | ACK | |
> |---------------->| |
> | | |
> | | |
>
> All the messages look up until SIPp sends an ACK in response to the 200 OK,
> but instead of sending an ACK to Asterisk, Opensips seems to be sending the
> ACK back to itself, and goes into a loop.
>
> These are the logs from tcpdump for the loopback interface on box 1:
>
> ================================================================
> 12:11:46.574820 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP
> (17), length 567)
> 192.168.1.57.sip-tls> 192.168.1.57.sip: SIP, length: 539
> INVITE sip:0119054741990 at 192.168.1.57:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.57:5061;branch=z9hG4bK-18363-1-0
> From: sipp<sip:sipp at 192.168.1.57:5061>;tag=18363SIPpTag001
> To: sut<sip:0119054741990 at 192.168.1.57:5060>
> Call-ID: 1-18363 at 192.168.1.57
> CSeq: 1 INVITE
> Contact: sip:sipp at 192.168.1.57:5061
> Max-Forwards: 70
> Subject: Performance Test
> Content-Type: application/sdp
> Content-Length: 133
>
> v=0
> o=user1 53655765 2353687637 IN IP4 192.168.1.57
> s=-
> c=IN IP4 192.168.1.57
> t=0 0
> m=audio 6000 RTP/AVP 0
> a=rtpmap:0 PCMU/800[|sip]
> 12:11:46.575332 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto UDP
> (17), length 338)
> 192.168.1.57.sip> 192.168.1.57.sip-tls: SIP, length: 310
> SIP/2.0 100 Giving a try
> Via: SIP/2.0/UDP 192.168.1.57:5061;branch=z9hG4bK-18363-1-0
> From: sipp<sip:sipp at 192.168.1.57:5061>;tag=18363SIPpTag001
> To: sut<sip:0119054741990 at 192.168.1.57:5060>
> Call-ID: 1-18363 at 192.168.1.57
> CSeq: 1 INVITE
> Server: OpenSIPS (1.7.1-notls (x86_64/linux))
> Content-Length: 0
>
>
> 12:11:46.577090 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto UDP
> (17), length 765)
> 192.168.1.57.sip> 192.168.1.57.sip-tls: SIP, length: 737
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.1.57:5061;branch=z9hG4bK-18363-1-0
> Record-Route:<sip:192.168.1.57;lr>
> From: sipp<sip:sipp at 192.168.1.57:5061>;tag=18363SIPpTag001
> To: sut<sip:0119054741990 at 192.168.1.57:5060>;tag=as1642d8ff
> Call-ID: 1-18363 at 192.168.1.57
> CSeq: 1 INVITE
> Server: Asterisk PBX 1.8.9.3
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH
> Supported: replaces, timer
> Contact:<sip:0119054741990 at 192.168.1.121:5060>
> Content-Type: application/sdp
> Content-Length: 209
>
> v=0
> o=root 1639240398 1639240398 IN IP4 192.168.1.121
> s=Asterisk PBX 1.8.9.3
> c=IN IP4 192.168.1.121
> t=0 0
> m=audio 10014 RTP/AVP 0
> a=rtpmap:0 PCMU/8000
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> 12:11:46.577303 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP
> (17), length 408)
> 192.168.1.57.sip-tls> 192.168.1.57.sip: SIP, length: 380
> ACK sip:0119054741990 at 192.168.1.57:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.57:5061;branch=z9hG4bK-18363-1-5
> From: sipp<sip:sipp at 192.168.1.57:5061>;tag=18363SIPpTag001
> To: sut<sip:0119054741990 at 192.168.1.57:5060>;tag=as1642d8ff
> Call-ID: 1-18363 at 192.168.1.57
> CSeq: 1 ACK
> Contact: sip:sipp at 192.168.1.57:5061
> Max-Forwards: 70
> Subject: Performance Test
> Content-Length: 0
>
>
> 12:11:46.577613 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto UDP
> (17), length 469)
> 192.168.1.57.sip> 192.168.1.57.sip: SIP, length: 441
> ACK sip:0119054741990 at 192.168.1.57:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.57;branch=z9hG4bKef25.4b9ff153.2
> Via: SIP/2.0/UDP 192.168.1.57:5061;branch=z9hG4bK-18363-1-5
> From: sipp<sip:sipp at 192.168.1.57:5061>;tag=18363SIPpTag001
> To: sut<sip:0119054741990 at 192.168.1.57:5060>;tag=as1642d8ff
> Call-ID: 1-18363 at 192.168.1.57
> CSeq: 1 ACK
> Contact: sip:sipp at 192.168.1.57:5061
> Max-Forwards: 69
> Subject: Performance Test
> Content-Length: 0
>
>
>
> 12:11:46.578151 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto UDP
> (17), length 530)
> 192.168.1.57.sip> 192.168.1.57.sip: SIP, length: 502
> ACK sip:0119054741990 at 192.168.1.57:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.57;branch=z9hG4bKef25.4b9ff153.2
> Via: SIP/2.0/UDP 192.168.1.57;branch=z9hG4bKef25.4b9ff153.2
> Via: SIP/2.0/UDP 192.168.1.57:5061;branch=z9hG4bK-18363-1-5
> From: sipp<sip:sipp at 192.168.1.57:5061>;tag=18363SIPpTag001
> To: sut<sip:0119054741990 at 192.168.1.57:5060>;tag=as1642d8ff
> Call-ID: 1-18363 at 192.168.1.57
> CSeq: 1 ACK
> Contact: sip:sipp at 192.168.1.57:5061
> Max-Forwards: 68
> Subject: Performance Test
> Content-Length: 0
>
>
> 12:11:46.578627 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto UDP
> (17), length 591)
> 192.168.1.57.sip> 192.168.1.57.sip: SIP, length: 563
> ACK sip:0119054741990 at 192.168.1.57:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.57;branch=z9hG4bKef25.4b9ff153.2
> Via: SIP/2.0/UDP 192.168.1.57;branch=z9hG4bKef25.4b9ff153.2
> Via: SIP/2.0/UDP 192.168.1.57;branch=z9hG4bKef25.4b9ff153.2
> Via: SIP/2.0/UDP 192.168.1.57:5061;branch=z9hG4bK-18363-1-5
> From: sipp<sip:sipp at 192.168.1.57:5061>;tag=18363SIPpTag001
> To: sut<sip:0119054741990 at 192.168.1.57:5060>;tag=as1642d8ff
> Call-ID: 1-18363 at 192.168.1.57
> CSeq: 1 ACK
> Contact: sip:sipp at 192.168.1.57:5061
> Max-Forwards: 67
> Subject: Performance Test
> Content-Length: 0
>
> ================================================================
>
>
> This is my opensips.cfg (rest of the config file is the default config)
> ================================================================
> route[1] {
> # for INVITEs enable some additional helper routes
> if (is_method("INVITE")) {
> t_on_branch("2");
> t_on_reply("2");
> t_on_failure("1");
> }
>
> if (is_method("INVITE")) {
> rewritehostport("192.168.1.121:5060");
> if (!t_relay()) {
> xlog("t_relay failed: ret:$retcode\n");
> sl_reply_error();
> } else {
> xlog("t_relay successful\n");
> }
> }
>
> exit;
> }
>
> ================================================================
>
> 1. Is t_relay the right application for me to use?
> 2. Do I need to add a case to handle the ACK's, if yes, how can I do this.
>
> Any help or pointers is appreciated.
>
>
> --
> View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/help-configuring-Opensips-as-proxy-tp7330142p7330142.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
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>
--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
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