[OpenSIPS-Users] ACK timout OpenSIPS 1.5 Still not resolved

[Khan]

Deag Bogdan,

I have force_rport() in the beginning of script as you can see in the
link  http://pastebin.com/mcec311 (highlighted section is where i
added NAT traversal logic)

also the log of failure is at this link <<< call made and ACK timed
out  >>> http://pastebin.com/m1d11246a

I tried to figure out the problem, the highlighted parts might be the
problem area if you could please give a quick look at see where in
configuration script i went wrong?

I know i am asking for too much but please help me, I really
appreciate your help !


Khan

On Fri, Apr 10, 2009 at 7:44 AM, Bogdan-Andrei Iancu
<bogdan at voice-system.ro> wrote:
> Hi Khan,
>
> The 401 is for a REGISTER (look at the Cseq header).
>
> anyhow, the lack of an ACK from the caller means the caller didn't received
> the reply (200 OK). If the caller is behind a nat, be sure you do
> force_rport() in script (at INVITE time) - this will correctly route back
> the replies via the NAT.
>
>
> Regards,
> Bogdan
>
> Khan wrote:
>>
>> Ok,
>>
>> I guess I sort of see the problem but dont know how to fix it... i
>> capture the trafic from the SjPhone UAC which transmit OPTIONS after
>> 200 OK, it seems like its getting 401 from server on authentication,
>> wonder why?
>>
>> Here is a link http://pastebin.com/m298ec8c6
>> please let me know if i am on the right track!!!!
>>
>> Thanks for all your time and efforts...
>>
>> Khan
>>
>> On Thu, Apr 9, 2009 at 11:00 AM, Khan <khansfriend at gmail.com> wrote:
>>
>>>
>>> On Thu, Apr 9, 2009 at 2:13 AM, Uwe Kastens <kiste at kiste.org> wrote:
>>>
>>>>
>>>> Khan,
>>>>
>>>> Would it be possible to add a tcpdump/wirshark on the opensips and on
>>>> the client in the external network? That make it much easier to debug.
>>>>
>>>
>>> I haven't done this before so, let me try to get the tcpdump for you,
>>> I will install wireshark today (like i said im rookie)
>>> I will post the tcpdump today :)
>>>
>>>
>>>>
>>>> One question: If you use xlite internaly, is the call dropped after
>>>> 35secs or not?
>>>>
>>>
>>> No, it only happens outside the network, I believe my NAT traversal
>>> works fine, for some reasons my voice reaches them but theirs is lost
>>> somewhere in clouds :)
>>>
>>>
>>>>
>>>> BR
>>>>
>>>> Uwe
>>>>
>>>> Khan schrieb:
>>>>
>>>>>
>>>>> Uwe,
>>>>>
>>>>> I am using xlite within my network which works fine the problem is
>>>>> outside the network, Xlite sends SUBSCRIBE and SJphone Sends OPTIONS
>>>>> request...
>>>>>
>>>>> An example of debug is as follows,
>>>>>
>>>>>
>>>>> Xlite registered fine the dump during the call process is as follows,
>>>>> the call last for 35 seconds in which other party could hear me but i
>>>>> can see a message on my Sjphone "ACK message awaiting" and then it
>>>>> disconnects with the message "Network failure" please review the
>>>>> following link...
>>>>> http://pastebin.com/dca5bbb0
>>>>>
>>>>> Another example is this SJphone which registers fine but after
>>>>> registration constantly sends the OPTIONS requsts. The link is as
>>>>> follows:
>>>>> http://pastebin.com/d3a4fb379
>>>>>
>>>>> My opensips.cfg is at this link:
>>>>> http://pastebin.com/d6ce3e43d
>>>>>
>>>>> Thanks for all your help ...
>>>>>
>>>>>
>>>>> Khan
>>>>>
>>>>>
>>>>>
>>>>> On Wed, Apr 8, 2009 at 1:46 PM, Uwe Kastens <kiste at kiste.org> wrote:
>>>>>
>>>>>>
>>>>>> Hi Khan,
>>>>>>
>>>>>> A easy way to debug this problem is to use a kind of network sniffer
>>>>>> on
>>>>>> your opensips and directly after your UA. Try to debug this issue with
>>>>>> a
>>>>>> softphone like xlite, so you can start your network dump on the
>>>>>> client.
>>>>>>
>>>>>> BR
>>>>>>
>>>>>> Uwe
>>>>>>
>>>>>> Khan schrieb:
>>>>>>
>>>>>>>
>>>>>>> Hi everyone,
>>>>>>>
>>>>>>> I'm rookie in SIP technology, strugling with several issues. I am
>>>>>>> having problem with UAC's outside network. I have 3 UAC registered
>>>>>>> within the network (SJ Phone, Xlite) they are working fine, I can
>>>>>>> talk
>>>>>>> within the network but the problem arrise when I use the UAC outside
>>>>>>> my network. I am seeing two different things from two different
>>>>>>> UAC's.
>>>>>>>
>>>>>>> 1. Xlite on a network behind NAT try to register, it registers
>>>>>>> successfully after receiving 200 OK it starts senting SUBSCRIBE
>>>>>>> requests, which results in 483 Erro (set up in my config) and when
>>>>>>> call is placed on this it gives ACK time out, person on the other
>>>>>>> side
>>>>>>> can hear me but i cant hear him.
>>>>>>>
>>>>>>> 2. SjPhone is on another network behind NAT, it regiesters fine, and
>>>>>>> after registration it starts sending OPTIONS request, which I have
>>>>>>> configured to respond as 200 OK. It continiously keep sending the
>>>>>>> requst and my config respond to 200 OK.
>>>>>>>
>>>>>>> My question is several parts, what am I doing wrong,
>>>>>>> a) why don't I get ACK after 200 OK,
>>>>>>> b) how do i handle SUBSCRIBE requests
>>>>>>> c) how do i handle OPTIONS request
>>>>>>>
>>>>>>> The sever is simply being used as SIP server for calls, no IM, Video,
>>>>>>> or other applications are implemented yet. There are OpenSIPS, MySQL
>>>>>>> server, and RTPProxy is running on the box.
>>>>>>>
>>>>>>> Please respond to my request considering my skills in the SIP as
>>>>>>> rookie, guide me on how to resolve problem...
>>>>>>>
>>>>>>> Thanks,
>>>>>>>
>>>>>>>
>>>>>>> Khan....
>>>>>>>
>>>>>>> Sorry for such a long email, I am frustrated :(
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Users mailing list
>>>>>>> Users at lists.opensips.org
>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> --
>>>>>>
>>>>>> kiste lat: 54.322684, lon: 10.13586
>>>>>>
>>>>>>
>>>>
>>>> --
>>>>
>>>> kiste lat: 54.322684, lon: 10.13586
>>>>
>>>>
>>
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