On Jul 16, 2014, at 4:05 PM, Andras FOGARASI <fogar...@fogarasi.com> wrote:

> On 7/16/14, 10:00 PM, Frank Carmickle wrote:
>> 
>> On Jul 16, 2014, at 3:54 PM, Daniel-Constantin Mierla <mico...@gmail.com> 
>> wrote:
>> 
>>> Hello,
>>> 
>>> I expect that the signaling is ok at least for call setup.
>>> 
>>> From signling point of view, I can think of following situations:
>>> - endpoints send keep alive packets (or session updates) which are no 
>>> answered. You can add an xlog(...) at the top of request_route{} and 
>>> reply_route{} blocks printing at least the method, call-id, cseq, from and 
>>> to header, plus the response code for reply block. In this case you can see 
>>> if there is some signaling before call is dropped.
>> 
>> Is this happening just on calls between two phones in your domain, or is 
>> there a carrier/federation involved?
>> 
> 
> No other parties are involved, only the two phones involved (and the
> proxy of course).
> 

I would expect that if it was a NAT issue you would see it much sooner than 15 
minutes, 30-60 seconds.  Are session timers being stripped by Kamailio?  You 
say it's a TURN server or is it acting more like a media relay where it is 
signaled into the path?  What TURN server are you using?  How is it configured?


--FC


_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Reply via email to