[OpenSIPS-Users] ACK timout OpenSIPS 1.5 Still not resolved
Bogdan-Andrei Iancu
bogdan at voice-system.ro
Wed Apr 22 09:28:31 CEST 2009
Hi Khan,
The logs shows very clear that the 200 OK was received by opensips and
sent out to the caller. Unfortunately there is no ACK coming from
caller. You need to do a SIP capture (IP level) on the caller side to
see if the caller is actually receiving the 200 OK. As said, a missing
ACK from caller probably means it never received the 200 OK.
Regards,
Bogdan
Khan wrote:
> 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
>>>>>
>>>>>
>>>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>
>
>
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