[SR-Users] Kamailio - Asterisk: Handling loss of SIP BYE and dangling channels

[Benjamin Fitzgerald]

Hi Alex,

Thanks for your quick response.

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.

2. Since Asterisk is also a serving as PSTN gateway, I like this because it
allows me to control calls with SIP endpoints separately. We have no issues
with all PSTN calls and I'd like to keep it that way :)

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. 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.)

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.

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.

Benjamin Fitzgerald
LETS Corporation
(925) 235-1154
ben at letscorp.us




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On Fri, Jan 8, 2016 at 12:08 PM, Alex Balashov <abalashov at evaristesys.com>
wrote:

> 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-id
>
> 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/
>
>
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