_______________________________________________On Tue, Apr 30, 2019 at 3:27 AM Rhys Hanrahan <rhys@nexusone.com.au> wrote:_______________________________________________Hi Everyone,
I would just like to add that I’m also very interested in several of the things Daniel mentioned in this thread. Particularly the RTT/latency information for NAT’d contacts is very useful – so that’s a +1 from me.
As someone who is trying to migrate from using Asterisk as our registrar, to using Kamailio, there’s several things Asterisk does with this info that our support team relies on heavily for day to day operations/support, and I need to find a way to replicate this behaviour.
As an example of how we use this:
- Asterisk will log when SIP endpoints become lagged or unreachable based on the OPTIONS RTT – so you can set a threshold and if a device takes too long to respond, Asterisk will log that it first Lagged, then Unreachable.
- Asterisk will then log when a handset comes back online showing it as “Reachable”.
This is a really handy way of historically trying to diagnose call drop outs or call quality issues, as you can quickly see with a few greps of syslog if all handsets at a particular IP at a particular time are dropping out at the same time, or not. While the goal would be to have proper monitoring of a customer’s internet connection, this can’t always be done. And having access to the latency on these NAT ping packets is extremely helpful in this case. Even with internet monitoring, having stats “per handset” is very useful.
Not everyone would want to spam their logs with this info – but having access to the RTT information so you can decide what to do with it in your config is critical in my opinion.
I was not aware of the UDP limitation of nathelper, but that explains some issues I saw, and that’s going to be an issue for us as well.
So I would be very keen to see the features discussed further. I am trying to learn C at the moment so hopefully I can assist in some way in future as well. :-)
Thanks,
Rhys.
From: sr-users <sr-users-bounces@lists.kamailio.org> on behalf of Nuno Ferreira <nferreira@fuze.com>
Reply-To: "Kamailio (SER) - Users Mailing List" <sr-users@lists.kamailio.org>
Date: Tuesday, 5 February 2019 at 1:37 am
To: "Kamailio (SER) - Users Mailing List" <sr-users@lists.kamailio.org>
Subject: Re: [SR-Users] Kamailio OPTIONS Round-Trip
Hey there,
Just trying to see if there was any conclusion on this topic...
Thanks
On Wed, Jan 16, 2019 at 3:18 PM Sergiu Pojoga <pojogas@gmail.com> wrote:
After re-reading the original question, it appears that it isn't about Asterisk at all, it was a simple reference to it, the actual question being how to get RTT in Kamailios's usloc.
Apologies for the confusion. Let's carry on with the topic, lol.
Cheers,
--Sergiu
On Wed, Jan 16, 2019 at 9:55 AM Sergiu Pojoga <pojogas@gmail.com> wrote:
Correct me if I'm wrong, but wasn't the original poster looking for Asterisk (behind Kamailio) to show the round-trip of its peers based on qualify OPTIONS requests that Asterisk sends out to the peer?
If so, I'm curious what is the impediment not to accept the previously suggested sip PATH approach? Aside from elegance and simplicity to implement, it isn't even subject to the UDP limitation you've brought up.
Not that the topic of usrloc qualify isn't of interest, but it just feels like we are drifting into another topic, although somehow related to the original one.
Best regards,
--Sergiu
On Wed, Jan 16, 2019 at 7:59 AM Nuno Ferreira <nferreira@fuze.com> wrote:
Hello Daniel,
I'm reading this thread with some interests. We were planning to use nat_traversal module to do keepalive, but we came across the UDP only limitation. In our use case, we wanted to offload the registrar from doing keepalive. Of course, that's an option, but it has yet another limitation when having active/active registrar servers using dmq_usrloc. If one of the registrars goes down which server will be in charge of doing keepalive for the contacts previously registered on the faulty registrar? That was one of the reasons for us to seek doing keeplive on the edge with nat_traversal, but again it's only valid for UDP.
If usrloc/dmq_usrloc provides some automatic election mechanism to keepalive orphan AORs, I would prefer going with it for the task. Another benefit like I read from your words is that we would automatically have available latency/rtt attached to each contact and that is a big plus.
Regards,
Nuno
On Wed, Jan 16, 2019 at 12:26 PM Daniel-Constantin Mierla <miconda@gmail.com> wrote:
Hello,
maybe we can just add this feature to the usrloc module -- right now the nat keepalive is done from nathelper module, which queries usrloc module to retrieve the list of the contacts to send OPTIONS to. Of course, the nathelper has the other variant witj 4-bytes pings, but I expect not many are using it these days.
Furthermore, because the nathelper has some options to forge the source ip address as well as willing to be lightweight, it sends the packets directly, no relying on tm module.
However, it seems that it is an increase interest in having more feedback based on these keepalives. Including the ability to do mirroring for sipcapture (a feature request being open in the tracker). Other request in the past was to send OPTIONS also for non-UDP contacts, nathelper does it only for UDP.
So we can consider adding a transaction based keepalive layer, which of course might take a bit more resources that current nathelper implementation, but can bring extra benefits. I think we can leave nathelper as it is and add this feature directly in the usrloc module, avoiding to pass data between modules, but also because we have to set/updates some fields in the contract structure (like this round trip time).
There are other modules that do keepalive, some mentioned dispatcher, there is a dedicated one named keepalive, and, afaik, also nat_traversal can do it. I am listing them so others can assert where it would be better to add the new feature -- as said, I would do it in usrloc, but I am open for other suggestions as well (eventually accompanied with a pull request).
Cheers,
DanielOn 15.01.19 21:12, Julien Chavanton wrote:
Depending on the use case, you could use the dispatcher module latency stats.
Regards
On Tue, Jan 15, 2019 at 2:29 AM Daniel Tryba <d.tryba@pocos.nl> wrote:
On Sun, Jan 13, 2019 at 10:08:31PM +0300, Soltanici Ilie wrote:
> With Asterisk, we are able to get some peer round-trip connection statistic by setting qualify=yes for the specified peer.
> It sends periodic OPTIONS to the peer and calculates the time round trip time.
> It's something like - "Status: OK (30 ms)".
> Is there any way to achieve this in Kamailio by using nathelper??module, or any other?
I think the only way to do this is to make this yourself (tm). In your
favorite scripting language, query the locations and fire OPTIONS and
measure the time for a response (if any) on basis of the "random" callid
you create. If you route these requests through kamailio you will
prevent any NAT problems or connection with TCP endpoints.
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_______________________________________________Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users--Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/micondaKamailio World Conference - May 6-8, 2019 -- www.kamailioworld.comKamailio Advanced Training - Mar 4-6, 2019 in Berlin; Mar 25-27, 2019, in Washington, DC, USA -- www.asipto.com_______________________________________________
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