[OpenSIPS-Users] help configuring Opensips as proxy
Bogdan-Andrei Iancu
bogdan at opensips.org
Wed Feb 29 19:32:46 CET 2012
Hi David,
Here it is.
Regards,
Bogdan
On 02/29/2012 08:23 PM, David Yat Sin wrote:
> Hi Bogdan,
> Thanks for looking into this. Could you send me your SIPP UAC file.
>
>
>
> Regards,
> David
>
> On 12-02-29 1:20 PM, "Bogdan-Andrei Iancu"<bogdan at opensips.org> wrote:
>
>> 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-Op
>>> ensips-as-proxy-tp7330142p7330142.html
>>> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>> --
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> http://www.opensips-solutions.com
>>
>
--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
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