[Freeswitch-users] Call dropping after 32 seconds

[Rutu Patel]

Thanks for the reply.

Got your point about NATing issue and no response of 200 OK and as a
resoult ACK Timeout.
So, now to resolve the issue, if you can assist, what could be the possible
fixies?
>From where can i start? where to look?

Thanks.

--
Thanks,
Rutu Patel

<http://www.inextrix.com>

On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi at avimarcus.net> wrote:

> 5 second response: 32 seconds is a timer/[network/NAT] issue.
>
> You have lots of 200s to the user since it's waiting for an ACK and keeps
> retrying, but for whatever network reason (router... sip alg?), it isn't
> getting one, so it triggers a timer to stop the call.
>
>
> -Avi Marcus
> BestFone
>
> On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel at inextrix.com>
> wrote:
>
>> Hello,
>>
>> Having issue of call dropping after 32 seconds, here are the details-
>>
>> x.x.x.174: opensips server
>> x.x.x.166: freeswitch server
>> x.x.x.3:     another opensips server which is registered as gateway on
>> above freeswitch server
>> x.x.x.6: freeswitch server
>> x.x.x.47:  server through which the user is registered
>> I am trying to call from xxxx9 to xxxxxxx29858
>> xxxxxxx00181 is caller-id name and caller-id number
>>
>> Call flow is like this:
>> registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips
>> server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)
>>
>> Outbound-proxy is set to x.x.x.174 in Gateway configuration.
>>
>> 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174
>> 2) Call hit the freeswitch server x.x.x.166
>> 3) After '180 Ringing' and '183 Session Progress' packet
>> sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the
>> gateway x.x.x.3
>> But after 32 seconds call is dropped,
>> Within 32 seconds audio is ok from both end so it should not be the RTP
>> issue.
>> Here I have attached the file with sip logs, you can observer from the
>> file that, there are many '200 OK' from x.x.x.174 to x.x.x.166
>> and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then
>> 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped.
>>
>> What is wrong here? Any help would be appreciated here.
>>
>> Here is the file with sip logs
>> --
>> Thanks,
>> Rutu Patel
>> <http://www.inextrix.com>
>>
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>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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