[asterisk-dev] Asterisk Segfault After PJSIP Commit 5241
George Joseph
george.joseph at fairview5.com
Wed Mar 2 09:19:01 CST 2016
On Wed, Mar 2, 2016 at 2:56 AM, Ross Beer <ross.beer at outlook.com> wrote:
> Hi George,
>
> I have re-built the 'c1bf014ea08cf66835a6f000e2bd6c7da588da6b' commit
> and PJSIP and Asterisk hasn't crashed after reload. However it did take 25
> mins to load.
>
> As requested I have opened a ticket for the realtime issue:
>
> https://issues.asterisk.org/jira/browse/ASTERISK-25826
>
Got it, thanks.
>
>
> Basically, I think this could be resolved by a configuration option that
> stops sourcery/pjsip loading all peers at start-up as this is not
> needed for the current setup. This has been discussed before on the mailing
> list however it doesn't look like it progresses any further.
>
If you're up for trying something, you can comment out the
qualify_and_schedule_all function
in
line
s
1135
-1147
of res/res_pjsip/pjsip_options.c, then comment out its 2 references on
lines 1245 and 1281. If that drops your startup times, then we know we're
on the right track.
>
> I would like to thank you for all of your help tying to identify the issue
> and hope that we can resolve it soon.
>
No worries!
>
> Kind regards,
>
> Ross
>
> ------------------------------
> From: george.joseph at fairview5.com
> Date: Tue, 1 Mar 2016 16:27:06 -0700
>
> To: asterisk-dev at lists.digium.com
> Subject: Re: [asterisk-dev] Asterisk Segfault After PJSIP Commit 5241
>
>
>
> On Tue, Mar 1, 2016 at 3:07 PM, Ross Beer <ross.beer at outlook.com> wrote:
>
> ok,
>
> That took 15 mins to load and then crashed. This will be due to the
> pjsip_dlg_create_uas_and_inc_lock commit.
>
>
> It should not have crashed. That commit had the fix for it. If it did
> crash with that commit, open a Jira issue and attach a full backtrace.
>
>
>
>
> However 15 mins to start is a long time and would cause issues in a
> production environment.
>
>
> Would you open a Jira issue on the realtime problem (if one isn't already
> open).
> I'm starting to look at alternatives.
>
>
>
>
> Thank you for your help here,
>
> Ross
>
>
> ------------------------------
> From: george.joseph at fairview5.com
> Date: Tue, 1 Mar 2016 14:02:38 -0700
>
> To: asterisk-dev at lists.digium.com
> Subject: Re: [asterisk-dev] Asterisk Segfault After PJSIP Commit 5241
>
>
>
> On Tue, Mar 1, 2016 at 1:04 PM, Ross Beer <ross.beer at outlook.com> wrote:
>
> Hi George,
>
> Using a development test box for testing!!
>
> Asterisk 13.7.2 with no cache takes 4:12 to load, that with PJSIP Commit
> 5240
>
>
> Ok, try this combination...
> "git checkout c1bf014ea08cf66835a6f000e2bd6c7da588da6b"
> pjproject from trunk.
> with caching.
>
> The commit I referenced is the one that handles the
> pjsip_dlg_create_uas_and_inc_lock
>
>
>
>
>
>
>
> Qualify time on the aor is set to zero, I guess a query could be made to
> check for a value greater than zero instead of loading all endpoints.
>
> Ross
>
> ------------------------------
> From: george.joseph at fairview5.com
> Date: Tue, 1 Mar 2016 12:45:28 -0700
>
> To: asterisk-dev at lists.digium.com
> Subject: Re: [asterisk-dev] Asterisk Segfault After PJSIP Commit 5241
>
>
>
> On Tue, Mar 1, 2016 at 12:21 PM, Ross Beer <ross.beer at outlook.com> wrote:
>
> Hi George,
>
> No endpoints are qualified, there are 20,000 endpoints with only 75 static
> contacts defined in the aors. The database is a MySQL cluster.
>
> With the current Asterisk 13 branch with cache disabled and the latest
> PJSIP it takes 5 mins and then before finishing it crashes.
>
> With Asterisk 13.7.2 with cache it takes around 1 1/2 min to load, however
> due to the bug with PJSIP Commit 5241 asterisk crashes when using TLS
> devices.
>
>
> Try 13.7.2 without the cache. I'm trying to understand where the time is
> being spent. I know it will crash because of that bug. You're not doing
> this on a production system are you??
>
>
>
> The main issue here is that the endpoints are loaded as soon as PJSIP
> loads, ideally endpoints would only be loaded once a device registers or
> attempts to make a call. Much in the same way as Asterisk 1.8 chan_sip
> manages realtime.
>
> There is no need to load the endpoints as they are not qualified.
>
>
> How do you know they're not qualified if you don't load them? :)
>
> Time to load up a database with 20,000 endpoints I guess.
>
>
>
> Ross
>
> ------------------------------
> From: george.joseph at fairview5.com
> Date: Tue, 1 Mar 2016 11:58:15 -0700
> To: asterisk-dev at lists.digium.com
> Subject: Re: [asterisk-dev] Asterisk Segfault After PJSIP Commit 5241
>
>
>
> On Tue, Mar 1, 2016 at 11:38 AM, Michael Ulitskiy <mulitskiy at acedsl.com>
> wrote:
>
> Hello,
>
>
>
> Please see this discussion
> http://lists.digium.com/pipermail/asterisk-dev/2015-October/075122.html
>
> I guess you're talking about the same problem.
>
>
> It's possible.
>
>
>
>
>
> Michael
>
>
>
> On Tuesday, March 01, 2016 06:26:27 PM Ross Beer wrote:
>
> > Hi George,
>
> >
>
> > We need to store contacts in realtime for our system. However not all
> endpoints are registered only about 200, yet asterisk loops through every
> endpoint which has been defined. It does this if contacts are in realtime
> or not.
>
> >
>
> > Its almost like pjsip is loading them to check if they need to be
> qualified etc.
>
> >
>
> > Asterisk 1.8 only put things into cache once they were accessed, is this
> an option for sourcery?
>
>
> Well, in order to initiate qualify of contacts, Asterisk does have to
> "access" them all so I'm not quite sure what the problem is.
>
> Can we reset to a known config and see what happens?
>
> pjproject from the published 2.4.5 tarball.
> Asterisk from the published 13.7.2 tarball.
> Disable memory_cache altogether in sorcery.conf.
>
> See what happens.
>
> Give me an estimate of how many endpoints and aors there are in the
> database, how many of those aors have static contacts defined, and what's
> your qualify interval.
>
> An idea of your database setup would help as well. Same server, local,
> remote, etc.
>
> Let's solve 1 problem at a time.
>
>
>
>
>
> -- _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> asterisk-dev mailing list To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
>
> -- _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> asterisk-dev mailing list To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
>
> -- _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> asterisk-dev mailing list To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
>
> -- _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> asterisk-dev mailing list To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20160302/dbd3cea8/attachment-0001.html>
More information about the asterisk-dev
mailing list