[SR-Users] Call disconnects within 30 seconds

[davy van de moere]

Hey Rahul,

thank you for your pcap file. Your astpp is indeed not succeeding in
getting the OK delivered correctly to your pjsua. Also those 400 Bad
Session Description are no good.

I do see as well fragmented packets passing by, perhaps an easy test would
be to have everything be sent over TCP. Big sip packets tend to suffer over
UDP networks. From your sip packets your setup looks quite standard,
although with a lot of headers (which is fine), so I'ld give it a try with
sending everything over tcp, t_relay_to_tcp should help you out!

Good luck!


--- In reply to ---
I guess the attachment size was greater than 60Kb thereby getting
quarentined.
Re-sending it to you so that this issue could be resolved.


2014-08-12 14:42 GMT+02:00 davy <davy.van.de.moere at gmail.com>:

> If your Kamailio setup is close to vanilla, it should do it by default.
> But Kamailio is a very powerfull tool, it can easily be setup to stop
> passing over ACKs :)
>
> To attempt to answer what explicitly sends along the ACKs, that will most
> likely be the t_relay function, together with the logic which went before
> it…
>
> A good old tcpdump will most likely enlighten us.
>
> Op 12-aug.-2014, om 14:39 heeft Rahul MathuR <rahul.ultimate at gmail.com>
> het volgende geschreven:
>
> Hello Davy,
>
> Thanks for writing back..
>
> Tonight I'll take the tcpdump on Kamailio box and share the file.
>
> Please note that Kamailio and Freeswitch are both on public IP & at
> Freeswitch param, enable_timer=false is set.
>
> Is there any explicit way wherein ACKs can be transmitted to FS ?
>
>
> Thanks in advance !
>
>
> On Tue, Aug 12, 2014 at 1:15 PM, davy van de moere <
> davy.van.de.moere at gmail.com> wrote:
>
>> Are you sure you're getting the ACK correctly to FS?
>>
>> FS typically has this behavior when it did not correctly receive a
>> confirmation of an answer, and after 30 seconds disconnects, as for FS the
>> call has failed.
>>
>> Do you have a trace of the packets?
>>
>> grtz,
>> Davy Van De Moere
>>
>>
>> 2014-08-12 13:37 GMT+02:00 Rahul MathuR <rahul.ultimate at gmail.com>:
>>
>>> Hello,
>>>
>>> I have an iPhone/Android/Windows 8 based UAC, proxy server Kamailio and
>>> Sip server FreeSwitch.
>>> Whenever I call directly from UAC to Sip server, the call gets
>>> established for as long as I want, however when I use the proxy in between,
>>> it gets disconnected within 30 seconds. It seems that FS sends a BYE within
>>> 30 seconds.
>>>
>>> I would really appreciate if anybody please guide me where I am going
>>> wrong in this case ?
>>>
>>> --
>>> Warm Regds.
>>> MathuRahul
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
>
> --
> Warm Regds.
> MathuRahul
>  _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20140813/9780c66b/attachment.html>