Hi Benjamin, To some extent, this is just a perennial, existential problem of using a proxy, so part of the answer is going to be that you need fundamentally reliable signalling, speaking from the vantage point of something which operates are a signalling relay (i.e. Kamailio). However, I understand that reality does not mirror expectations. As the purveyor of a SIP service delivery platform based entirely on Kamailio, we run into this problem all the time, particularly since our system generates accounting records with billing involvement. There are some well-established and canonical solutions: 1. You make it sound like the Asterisk channel stays up indefinitely in such a situation. Why is that? The normal behaviour is for Asterisk to hang up the call after some number of seconds without incoming RTP. It's likely that tuning the RTP timeout setting to something conservative[1] would solve a lot of your problems off the bat. 2. The Kamailio 'dialog' module can spoof a BYE toward both endpoints based on an absolute dialog timeout (regardless of whether both dialog peers are still actively engaged), which can be set globally or on a per-dialog basis: http://kamailio.org/docs/modules/4.3.x/modules/dialog.html#timeout-avp-id http://kamailio.org/docs/modules/4.3.x/modules/dialog.html#default-timeout-… http://www.kamailio.org/wiki/cookbooks/4.3.x/pseudovariables#dlg_ctx_attr 3. The 'dialog' module also has a dead peer detection / keepalive scheme based on sequential OPTIONS pings: http://kamailio.org/docs/modules/4.3.x/modules/dialog.html#idp1898328 If one or both of the peers don't respond to these, the dialog will be timed out, and if you've set $dlg_ctx(timeout_bye) = 1, this will result in a spoofed BYE toward both peers as well. 4. There are various other signalling-oriented UA-side mechanisms intended to solve this problem as well, such as SIP Session Timers (RFC 4028). ... Of course, all this depends on the maintenance of dialog state in Kamailio, which is an additional complication and a potential wrinkle if that data were to be lost. So, it's a bit hard to say whether Kamailio is the _best_ place to solve this problem. The first line of defence really should be at the endpoint level on both sides of the proxy. Beyond that, Kamailio does offer some pragmatic solutions. -- Alex [1] Notwithstanding RTP interruptions due to VAD, hold, etc. -- 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(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Benjamin, On 01/08/2016 03:25 PM, Benjamin Fitzgerald wrote: 1. Sorry to be unclear, the Asterisk channel does not stay up 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… https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.s… (Not sure which SIP channel driver you're using.) 3. I'm not sure this will work in my case because the endpoint is 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 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 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 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(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users