[OpenSIPS-Users] OpenSIPS Perl module branching and logging

[David Cunningham]

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
>



-- 
David Cunningham, Voisonics
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