[SR-Users] Kamailio OPTIONS Round-Trip
Daniel-Constantin Mierla
miconda at gmail.com
Tue Feb 25 13:42:32 CET 2020
On 19.02.20 21:55, David Villasmil wrote:
> +1 here. This feature would be a BIG plus. I’m also interested in what
> Nuno pointed out, how is it decided which registrar will send the
> OPTIONS to the UAC if we have multiple registrars sharing contacts via
> DMQ?
What scenarios/network topologies are considered here? The registrar
servers being behind an edge proxy (sbc) or also anycast? With no edge
proxy, the registrar server that received the register request has to do
the keepalive.
Currently, iirc, the server_id can be used to group the contacts and do
keepalives from a specific server -- as done now by nathelper in
stateless mode.
The feature is on my to-do list for quite some time, just didn't get a
long enough spare frame to start doing it. Hope to get to it soon, if
nobody else jumps in before.
Cheers,
Daniel
>
> On Wed, 19 Feb 2020 at 20:18, Joel Serrano <joel at textplus.com
> <mailto:joel at textplus.com>> wrote:
>
> Just for future reference:
>
> https://github.com/kamailio/kamailio/issues/2223
>
> On Tue, Apr 30, 2019 at 3:27 AM Rhys Hanrahan
> <rhys at nexusone.com.au <mailto:rhys at 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 at lists.kamailio.org
> <mailto:sr-users-bounces at lists.kamailio.org>> on behalf of
> Nuno Ferreira <nferreira at fuze.com <mailto:nferreira at fuze.com>>
> *Reply-To: *"Kamailio (SER) - Users Mailing List"
> <sr-users at lists.kamailio.org <mailto:sr-users at lists.kamailio.org>>
> *Date: *Tuesday, 5 February 2019 at 1:37 am
> *To: *"Kamailio (SER) - Users Mailing List"
> <sr-users at lists.kamailio.org <mailto:sr-users at 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 at gmail.com <mailto:pojogas at 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 at gmail.com <mailto:pojogas at 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 at fuze.com <mailto:nferreira at 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 at gmail.com
> <mailto:miconda at 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,
> Daniel
>
> On 15.01.19 21:12, Julien Chavanton wrote:
>
> Depending on the use case, you could use
> the dispatcher module latency stats.
>
>
>
> https://kamailio.org/docs/modules/devel/modules/dispatcher.html#dispatcher.p.ds_ping_latency_stats
>
>
>
> Regards
>
>
>
> On Tue, Jan 15, 2019 at 2:29 AM Daniel
> Tryba <d.tryba at pocos.nl
> <mailto:d.tryba at 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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> --
>
> Daniel-Constantin Mierla -- www.asipto.com <http://www.asipto.com>
>
> www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda>
>
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>
> David Villasmil
> email: david.villasmil.work at gmail.com
> <mailto:david.villasmil.work at gmail.com>
> phone: +34669448337
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