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

[Bogdan-Andrei Iancu]

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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