+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(a)textplus.com
<mailto:joel@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(a)nexusone.com.au <mailto: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(a)lists.kamailio.org
<mailto:sr-users-bounces@lists.kamailio.org>> on behalf of
Nuno Ferreira <nferreira(a)fuze.com <mailto:nferreira@fuze.com>>
*Reply-To: *"Kamailio (SER) - Users Mailing List"
<sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org>>
*Date: *Tuesday, 5 February 2019 at 1:37 am
*To: *"Kamailio (SER) - Users Mailing List"
<sr-users(a)lists.kamailio.org <mailto: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(a)gmail.com <mailto: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(a)gmail.com <mailto: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(a)fuze.com <mailto: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(a)gmail.com
<mailto: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,
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
<mailto: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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