[Freeswitch-users] Call dropping after 32 seconds

Ahmed Habiba ahabiba at gmail.com
Mon Feb 29 17:13:06 MSK 2016 

Hi,

Where is the firewall, is it between user and freeswitch or freeswitch and opensips

All cases freeswitch should send the firewall IP for the endpoint behind firewall so that the endpoint would be able to communicate back to freeswitch correct external IP.

Thanks,

Ahmed Habiba.


From: Rutu Patel <rutu.patel at inextrix.com <mailto:rutu.patel at inextrix.com>>
Subject: Re: [Freeswitch-users] Call dropping after 32 seconds
Date: February 29, 2016 at 4:15:28 PM GMT+3
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>


Hi Jurijs,

We have to consider the setup with freeswitch and opensips only.
registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) 

And there is a firewall and we have added x.x.x.3 IP there.

Is it possible there is some network related issue Or any sip profile parameter we need to set?

--
Thanks,
Rutu Patel

On Tue, Feb 23, 2016 at 1:07 PM, Jurijs Ivolga <jurij.ivo at gmail.com <mailto:jurij.ivo at gmail.com>> wrote:
Hi,

1) You have very complex set-up and I doubt that you need it.

2) As far as you have user with ip x.x.x.174 and opensips server with same ip x.x.x.174 it very hard to debug. So I propose you to send new log where will be difference between user ip and opensips IP.

3) If you have possibility, try to register directly with a user to x.x.x.3 gateway and check if same issue still exists, if there is no such issue anymore, then thee is definitely issue in your opensips x.x.x.174 and freeswitch x.x.x.166. My point here is that you need to isolate issue and to understand what part of your set-up works as expected and what is faulty.

With kind regards,  

Jurijs

2016-02-23 7:51 GMT+02:00 Rutu Patel <rutu.patel at inextrix.com <mailto:rutu.patel at inextrix.com>>:
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 <mailto: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 <mailto: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/>
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>

Official FreeSWITCH Sites
http://www.freeswitch.org <http://www.freeswitch.org/>
http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
http://www.cluecon.com <http://www.cluecon.com/>

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
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http://www.freeswitch.org <http://www.freeswitch.org/>


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>

Official FreeSWITCH Sites
http://www.freeswitch.org <http://www.freeswitch.org/>
http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
http://www.cluecon.com <http://www.cluecon.com/>

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
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http://www.freeswitch.org <http://www.freeswitch.org/>


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>

Official FreeSWITCH Sites
http://www.freeswitch.org <http://www.freeswitch.org/>
http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
http://www.cluecon.com <http://www.cluecon.com/>

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
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Jurijs

2016-02-23 7:51 GMT+02:00 Rutu Patel <rutu.patel at inextrix.com <mailto:rutu.patel at inextrix.com>>:
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 <mailto: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 <mailto: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/>
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>

Official FreeSWITCH Sites
http://www.freeswitch.org <http://www.freeswitch.org/>
http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
http://www.cluecon.com <http://www.cluecon.com/>

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
http://www.freeswitch.org <http://www.freeswitch.org/>


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>

Official FreeSWITCH Sites
http://www.freeswitch.org <http://www.freeswitch.org/>
http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
http://www.cluecon.com <http://www.cluecon.com/>

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
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http://www.freeswitch.org <http://www.freeswitch.org/>


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>

Official FreeSWITCH Sites
http://www.freeswitch.org <http://www.freeswitch.org/>
http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
http://www.cluecon.com <http://www.cluecon.com/>

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
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http://www.freeswitch.org <http://www.freeswitch.org/>


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>

Official FreeSWITCH Sites
http://www.freeswitch.org <http://www.freeswitch.org/>
http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
http://www.cluecon.com <http://www.cluecon.com/>

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
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