[OpenSIPS-Users] OpenSIPS Perl module branching and logging

[David Cunningham]

Thanks Bogdan.

Could you give any ideas on this part of my first email?:

Can anyone advise how I can (in the Perl) extract the destination
addresses from $m as strings to verify that all the destinations are
correct after the final $m->append_branch()?


On Thu, May 13, 2010 at 6:09 PM, Bogdan-Andrei Iancu
<bogdan at voice-system.ro> wrote:
> Hi David,
>
> looking at the message, I see that the ACK is wrongly generated. The
> RURI of the ACK must be the contact received in 200 OK (as received by
> the phone). The 200 OK shows ast02, but the ACK goes to ast01 , which is
> bogus.
>
> Also the UAC seams to ignore the RR indication - according to the RR
> received in 200 OK, the UAC should sent the ACK to sip01 and not to
> ast01 .....
>
> Regards,
> Bogdan
>
> David Cunningham wrote:
>> Hi Bogdan,
>>
>> That sounds reasonable, but an ngrep trace on the opensips server is
>> showing the ACK from the phone is being forwarded to the wrong
>> Asterisk, i.e. not the one that sent the 200 OK.
>>
>> As I mentioned, we are passing routing of messages through a Perl
>> module, and this branches messages to all Asterisk servers. I have
>> logged the rewrite_ruri() and append_branch() steps so I know these
>> are happening to all Asterisk servers.
>> I'm therefore not sure how to determine where the fault is, given the
>> discrepancy between the logging in the Perl code, which what we're
>> seeing actually happen on the ngrep trace.
>>
>> Can you give any advice? Much appreciated!
>>
>>
>> I've also copied and pasted in here the 200 OK and the ACK. You can
>> see that ast02 sends the 200 OK to sip01, but then sip01 sends the ACK
>> on to ast01.
>>
>>
>> U 2010/05/11 08:29:01.674692 (ast02):5060 -> (sip01):5060
>> SIP/2.0 200 OK.
>> Via: SIP/2.0/UDP (sip01);branch=z9hG4bKd334.32c9cde4.1;received=(sip01).
>> Via: SIP/2.0/UDP (unknown other
>> IP):5060;rport=5060;received=(phone);branch=z9hG4bKaacc9f8227a7ea553.b59f1ab3b79f7cfcc.
>> Record-Route: <sip:(sip01);lr=on;ftag=b9d0d92b6c>.
>> From: "User" <sip:(calling number)@(sip01):5060>;tag=b9d0d92b6c.
>> To: "**" <sip:**@(sip01):5060>;tag=as1a1f4c27.
>> Call-ID: a7366e4855e295af.
>> CSeq: 18830 INVITE.
>> Server: Enswitch.
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
>> Supported: replaces, timer.
>> Contact: <sip:**@(ast02)>.
>> Content-Type: application/sdp.
>> Content-Length: 314.
>> .
>> v=0.
>> o=root 1399767030 1399767030 IN IP4 (ast02).
>> s=Asterisk PBX 1.6.1.12.
>> c=IN IP4 (ast02).
>> t=0 0.
>> m=audio 12876 RTP/AVP 0 8 18 101.
>> a=rtpmap:0 PCMU/8000.
>> a=rtpmap:8 PCMA/8000.
>> a=rtpmap:18 G729/8000.
>> a=fmtp:18 annexb=no.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=ptime:20.
>> a=sendrecv.
>>
>>
>>
>> U 2010/05/11 08:29:01.984181 (sip01):5060 -> (ast01):5060
>> ACK sip:**@(ast01):5060;transport=udp SIP/2.0.
>> Record-Route: <sip:(sip01);lr=on;ftag=b9d0d92b6c>.
>> Via: SIP/2.0/UDP (sip01);branch=z9hG4bKd334.32c9cde4.3.
>> Via: SIP/2.0/UDP (unknown other
>> IP):5060;rport=5060;received=(phone);branch=z9hG4bKc0d1abc8b8ff65c68.d1538fb17483e517f.
>> Route: <sip:(sip01);lr=on;ftag=b9d0d92b6c>.
>> Proxy-Require: com.nortelnetworks.firewall.
>> Max-Forwards: 69.
>> From: "User" <sip:(calling number)@(sip01):5060>;tag=b9d0d92b6c.
>> To: "**" <sip:**@(sip01):5060>;tag=as1a1f4c27.
>> Call-ID: a7366e4855e295af.
>> CSeq: 18830 ACK.
>> User-Agent: Aastra 57i/2.5.2.30.
>> Content-Length: 0.
>>
>>
>> On Wed, May 12, 2010 at 9:59 AM, Bogdan-Andrei Iancu
>> <bogdan at voice-system.ro> wrote:
>>
>>> In such a case, from SIP point of view, there is a single ACK
>>> corresponding to the winning branch (which sent the 200 OK).
>>>
>>> Regards,
>>> Bogdan
>>>
>>> David Cunningham wrote:
>>>
>>>> Hi Bogdan,
>>>>
>>>> That's correct, the ACK for the 200OK.
>>>>
>>>> On Wed, May 12, 2010 at 8:46 AM, Bogdan-Andrei Iancu
>>>> <bogdan at voice-system.ro> wrote:
>>>>
>>>>
>>>>> Hi David,
>>>>>
>>>>> David Cunningham wrote:
>>>>>
>>>>>
>>>>>> Hello,
>>>>>>
>>>>>> We are using OpenSIPS with a Perl module which needs to branch SIP
>>>>>> messages to multiple Asterisk servers. This is done with a
>>>>>> $m->rewrite_ruri() to the first destination, and an
>>>>>> $m->append_branch() to each following destination.
>>>>>> There's a problem though, that with ACK messages it only seems to
>>>>>> arrive at the first Asterisk server.
>>>>>>
>>>>>>
>>>>>>
>>>>> You mean the ACK for 200OK ?
>>>>>
>>>>> Regards,
>>>>> Bogdan
>>>>>
>>>>>
>>>>>> Does anyone know of a problem using this method to branch ACK messages?
>>>>>>
>>>>>> Can anyone advise how I can (in the Perl) extract the destination
>>>>>> addresses from $m as strings to verify that all the destinations are
>>>>>> correct after the final $m->append_branch()?
>>>>>>
>>>>>> Thanks for your help!
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>> --
>>>>> Bogdan-Andrei Iancu
>>>>> www.voice-system.ro
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>> --
>>> Bogdan-Andrei Iancu
>>> www.voice-system.ro
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>>
>>
>>
>
>
> --
> Bogdan-Andrei Iancu
> www.voice-system.ro
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



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