Benjamin,

On 01/08/2016 03:25 PM, Benjamin Fitzgerald wrote:

1. Sorry to be unclear, the Asterisk channel does not stay up
indefinitely. We do have a max timeout but since a large portion of our
business is based on conference calling, the timeout is rather large. I
will definitely change the RTP timeout as my first attempt.

Yes, but I was referring specifically to the RTP timeout. If the mobile endpoint goes away, it will stop sending RTP. If Asterisk detects that no RTP has been received in x seconds, it should hang up the channel, after prophylactically sending a BYE for the call in the direction of Kamailio/the mobile peer.

I had been under the impression that Asterisk has a fairly conservative default RTP timeout anyway, but it seems I may be mistaken:

https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip.conf.sample#L740

https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L624

(Not sure which SIP channel driver you're using.)

3. I'm not sure this will work in my case because the endpoint is
reachable, but client state is not in sync with the server: i.e.
Kamailio/Asterisk think it's in a call but the endpoint does not. If
sending OPTIONS could tell me if the endpoint thinks it's in a call or
not, then this could potentially work.

Would sending a BYE to both peers not have the effect of synchronising them forcefully to a state of "the call is hung up"?

If you're concerned about sending a BYE to an endpoint that thinks the call is already hung up, don't be. In that case, it'll simply be rejected. You can't negatively affect the state of a dialog that's already dead.

Curious, however: when you say "Kamailio/Asterisk think it's in a call", how does this apply to Kamailio?

Stateful SIP proxies are transaction-stateful, not dialog-stateful.

Thus, by default, Kamailio doesn't know anything about "calls", but only the SIP transactions of which they are made up, and only for so long as those transactions are active. The 'dialog' module allows Kamailio to be call-stateful, at the cost of additional statekeeping complexity, but you should only use this capability if you need it for something (e.g. limiting concurrent calls, keepalive/timeout as described previously, etc.)

On a side note, is there a SIP message that I can send to a client to
have it report its state? (Registered, Auth Failed, In a call, etc.)

There's no standard query mechanism like this that I am aware of; the only way of disseminating such state information with which I'm familiar is presence, which is proactively pushed out by the endpoints and requires server-side support.

4. I do know about SIP Session Timers but chose to not use them during
the initial deployment (because of Asterisk channel timeout which I know
realize is too large). Maybe this will help in conjunction with the
above methods.

SSTs are rather bureaucratic and, in my experience, often incorrectly implemented or unsupported. In the SST conception of things, the roles in keepalive ping-pong are negotiated entirely between the UAs, and it is up to the UAs to maintain those roles. This goes wrong easily enough that server-side solutions such as periodic reinvites and other "pings" (like the Kamailio dialog module's OPTIONS pings) are a rather popular alternative.

Would you mind expanding on endpoint defense? Specifically with mobile
client applications? I agree this would be the ideal solution, I'm just
not sure where to start here.

By "endpoint defence" I simply meant that detecting dead peers should be up to the SIP endpoints (mobile SIP client and Asterisk, by the sound of it) first and foremost, and that any proxy-side measures should be a secondary layer.

-- Alex

--
Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter Center North, Suite 300
Atlanta, GA 30346
United States

Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

_______________________________________________
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