asterisk users - Jan 2015 - PJ SIP realtime with Kamailio / opensips

Hi all,

I saw Matt Jordan's recent Kamailio world talk and was interested in the
idea he proposed of stripping out authentication and registration from
asterisk and letting Kamailio handle it.

All of the tutorials I've seen (e.g. on asipto) show Kamailio forwarding
registrations to asterisk.

In order to do what Matt suggested would I be correct in assuming I would
have to use the asterisk database rather than the Kamailio database? I've
compared the two and the table structures are different.

If I use the asterisk database I guess asterisk still needs to be
responsible for handling authentication, registration and writing the
contacts to the database. If I use the Kamailio database how would I dial a
local extension from asterisk if I'm using multiple domains?

For example 100 at domainA.com -> 200 at domainA.com

Or even

100 at domainA.com -> 300 at domainB.com

How would pjsip find the contact to dial?

As far as I can tell asterisk will have no idea who is registered or how to
find them (contact details). Maybe I'm over thinking something simple, or
maybe I'm not. Either way I would love your thoughts on how this could be
done.

My Kamailio is public facing, but talks to asterisk over an internal
network. Asterisk can face the internet but I'd rather not.

Thanks in advance for your help,

C
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On Wed, Jan 21, 2015 at 9:55 AM, Chirag Desai <djchillerz at gmail.com>
wrote:
> Hi all,
>
> I saw Matt Jordan's recent Kamailio world talk and was interested in
the
> idea he proposed of stripping out authentication and registration from
> asterisk and letting Kamailio handle it.
>
> All of the tutorials I've seen (e.g. on asipto) show Kamailio
forwarding
> registrations to asterisk.
>
> In order to do what Matt suggested would I be correct in assuming I would
> have to use the asterisk database rather than the Kamailio database?
I've
> compared the two and the table structures are different.
>
> If I use the asterisk database I guess asterisk still needs to be
> responsible for handling authentication, registration and writing the
> contacts to the database. If I use the Kamailio database how would I dial a
> local extension from asterisk if I'm using multiple domains?
>
A lot of this depends on what you're trying to accomplish.

If what you want to do is keep Asterisk as a 'media application server'
-
and leave as much of the SIP behaviour in Kamailio as possible - than you
can certainly have Kamailio act as the registrar. In that case, Kamailio
really should be responsible for all of the SIP fiddly bits of making sure
everything is okay before passing the requests over to Asterisk. That
includes:
* Authentication
* Endpoint identification
* Routing of requests from different groupings of 'users' (identified by
domain, or whatever you'd like) to specific Asterisk servers, etc.

In that scenario, Asterisk becomes a very simple media handling box. Need
VoiceMail? Great. Send an INVITE request, pull some specific info out of
some SIP headers that identify 'who it is', and let Asterisk do its
VoiceMail thing. Need Conferencing? Send an INVITE request to the
Conferencing box and do the same thing. Even better, you can go really
generic and use ARI to write and control the media applications that are
executing on each server. (Bonus points: use OpenStack and spin up Asterisk
instances on the fly.)

In the PJSIP stack today, that means you only use an anonymous endpoint
identifier. There's a draw back to this approach: all devices have to look
exactly the same, have the same sets of codecs, etc. Of course, as Daniel
and I were discussing on the -dev list earlier last week [1], you could
have a custom endpoint identifier that looks at SIP headers to map inbound
requests to PJSIP endpoints. This would make things a bit more flexible
than what is currently possible today.

[1] http://lists.digium.com/pipermail/asterisk-dev/2015-January/072309.html

> For example 100 at domainA.com -> 200 at domainA.com
>
> Or even
>
> 100 at domainA.com -> 300 at domainB.com
>
> How would pjsip find the contact to dial?
>
If you use an anonymous endpoint identifier, than everything that is
received inbound maps to a single endpoint. Outbound, that doesn't have to
be the case - but since Kamailio 'knows' where/who everyone is, you
should
let Kamailio deal with the routing. In that case, you would have an
endpoint(s) representing your Kamailio servers, and you would send the
requests to them.

Hence, your dialing would look something like:

Dial(PJSIP/kamailio_server/sip:300 at domainB.com)

If you want more logic on your Asterisk servers, I would use func_odbc to
extract who to dial from the Kamailio database tables, and use the result
of the ODBC query as the dial string in the Dial application.

> As far as I can tell asterisk will have no idea who is registered or how
> to find them (contact details). Maybe I'm over thinking something
simple,
> or maybe I'm not. Either way I would love your thoughts on how this
could
> be done.
>
> My Kamailio is public facing, but talks to asterisk over an internal
> network. Asterisk can face the internet but I'd rather not.
>
> Thanks in advance for your help,
>
>
This is one of those deployment considerations where I don't believe there
is a single answer. The question is: who takes on what role? Asterisk's
PJSIP stack is very flexible, and can be made as smart or as simple as
you'd like it to be. If you want Asterisk to act as the registrar, than it
can do that and expose that information. It can also leave that job to
Kamailio - although when you do that, you do lose all of the location based
information. But that can be a good thing, if you tailor your applications
to expect that.

Matt

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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