On Tue, Apr 30, 2019 at 3:27 AM Rhys Hanrahan <rhys(a)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(a)lists.kamailio.org> on behalf of Nuno
Ferreira <nferreira(a)fuze.com>
*Reply-To: *"Kamailio (SER) - Users Mailing List" <
sr-users(a)lists.kamailio.org>
*Date: *Tuesday, 5 February 2019 at 1:37 am
*To: *"Kamailio (SER) - Users Mailing List"
<sr-users(a)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(a)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(a)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(a)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(a)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.…
Regards
On Tue, Jan 15, 2019 at 2:29 AM Daniel Tryba <d.tryba(a)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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