[asterisk-users] Session-timers and TCP

[Aaron Hamstra]

All,

 

We are having issues with one of our customers.  They typically are
using remote sip clients on smart phones.  For the purpose of allowing
the apps to work properly in the background we have to utilize TCP which
works fine.

 

The problem comes up when the softphone loses connectivity for some
reason. The session timers are not ending the call as they do on a UDP
session.  Basically from the sip debug it sends the re-invite for the
session timer according to the sip debug and it appears all is fine
instead of not getting a response back from the client and disconnecting
the call as it does with udp. There is no way it is getting a response
back from the client however as the client has no network connectivity.

 

I have run some tcpdump's on the server and when tracing the call I
actually never see those re-invites going out at all from the server.

 

We are running asterisk 1.8.7.0 currently.

 

I can reproduce the issue at will by establishing a call from a
softphone and then putting it into airplane mode to simulate the
connectivity loss.  

 

Are session-timers expected to work with tcp?  If so can anyone tell me
where to look to see what might be going on?

 

 

Thanks in Advance.

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