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62mkv at mail.ru
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 Posted: Thu Jan 29, 2015 3:43 am Post subject: [asterisk-users] any valid up-to-date info about Kamailio-As |
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Hi all
Have recently watched Matt Jordan's session on Kamailio World 2014
On slides 26-29 of his presentation
(http://www.kamailio.org/events/2014-KamailioWorld/day1/09-Matt.Jordan-Asterisk12-And-PJSIP.pdf)
he speaks about a (completely new, for me at least) approach to build
scalable telephony systems, using N instances of Kamailio and N
instances of Asterisk
Are there any whitepapers, howtos, "implementation experience reports",
whatever, available, that would describe such an approach in details and
help some not-so-advanced admins to at least understand "if is it what
they need, or not exactly, or not at all" ?
We are planning to look closer at Kamailio (or any other proxy, like
OpenSip) as a way to do both load-balancing and failover solutions, so
that refusal of any Asterisk instance should have minimal possible
effect on the overall system availability.
A lot of questions howevere arise, like: what if one SIP user got
REGISTERed at Server 1, and the other on Server 3, so how can they call
one another ?
Also, outbound registrations can be done from one instance at a time,
say it's done from Server1 for Trunk1, so how can users, that got
authenticated at Server2, call thru that registration (Trunk1) ?
Also, Kamailio itself has to be protected from failing, and probably
even from overload...
Would be great to read something in-depth about that
Thanks!!
Kirill Marchuk
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mjordan at digium.com
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| Posted: Thu Jan 29, 2015 10:52 am Post subject: [asterisk-users] any valid up-to-date info about Kamailio-As |
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On Thu, Jan 29, 2015 at 2:43 AM, Kirill Marchuk <62mkv@mail.ru> wrote:
| Quote: | Hi all
Have recently watched Matt Jordan's session on Kamailio World 2014
On slides 26-29 of his presentation
(http://www.kamailio.org/events/2014-KamailioWorld/day1/09-Matt.Jordan-Asterisk12-And-PJSIP.pdf)
he speaks about a (completely new, for me at least) approach to build
scalable telephony systems, using N instances of Kamailio and N instances of
Asterisk
Are there any whitepapers, howtos, "implementation experience reports",
whatever, available, that would describe such an approach in details and
help some not-so-advanced admins to at least understand "if is it what they
need, or not exactly, or not at all" ?
We are planning to look closer at Kamailio (or any other proxy, like
OpenSip) as a way to do both load-balancing and failover solutions, so that
refusal of any Asterisk instance should have minimal possible effect on the
overall system availability.
|
The best documentation out there - that I'm personally aware of - is
Daniel's guide on integrating Kamailio and Asterisk:
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
While there have been quite a few improvements made in Asterisk (and I
imagine, Kamailio as well) since that was written, that guide would be
a good starting point, regardless of the versions involved.
| Quote: | A lot of questions howevere arise, like: what if one SIP user got REGISTERed
at Server 1, and the other on Server 3, so how can they call one another ?
|
There are many different ways of handling this.
First, you have to ask yourself what you want Asterisk and Kamailio to
do in your set up. Some sample questions:
* Who acts as the registrar?
* Who manages subscriptions?
* Should each Asterisk server have a special purpose, or should they
be treated as a generic pool of media servers?
* Should Asterisk be involved in 'normal' calls (two-party, no media
manipulation), or should it only be used when special services are
needed?
Your goal, in any scenario, should be to keep the Asterisk dialplan as
simple as possible. That typically means not placing customer specific
logic in the dialplan, but instead relying on func_odbc to pull
customer specific information from a database. In later versions (such
as Asterisk 13), you can remove much of the logic from the dialplan
and use ARI to build custom media applications.
But no, not a lot of this is written down yet.
| Quote: | Also, outbound registrations can be done from one instance at a time, say
it's done from Server1 for Trunk1, so how can users, that got authenticated
at Server2, call thru that registration (Trunk1) ?
|
If your Asterisk servers are sitting behind Kamailio, they should
probably just be registering to their Kamailio instances. Again, if
Kamailio is handling the registration, identification, and
authentication, then you probably don't want Asterisk doing any of
that. You would instead just have Asterisk "trust" that Kamailio is
sending it the right calls, and have it handle them accordingly.
| Quote: | Also, Kamailio itself has to be protected from failing, and probably even
from overload...
|
That's pretty standard stuff for Kamailio.
| Quote: | Would be great to read something in-depth about that
|
--
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
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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miconda at gmail.com
Guest
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| Posted: Tue Feb 03, 2015 7:24 am Post subject: [asterisk-users] any valid up-to-date info about Kamailio-As |
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|
From Kamailio point of view, the tutorial referred here
(http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb)
should be quite actual. As Matt said, we do have new features with more
recent releases 4.1.x and 4.2.x but the relevant parts in the relation
with Asterisk (authentication, registration, etc.) are more or less the
same.
If Asterisk preserved pretty much its old realtime mechanism and
database structure, then should be straightforward to adjust in case of
small changes.
I hope to get a new tutorial that uses latest Kamailio and Asterisk 13
in the near future, targeting to use ARI instead of database for making
the integration of the two applications.
Cheers,
Daniel
On 29/01/15 16:52, Matthew Jordan wrote:
| Quote: | On Thu, Jan 29, 2015 at 2:43 AM, Kirill Marchuk <62mkv@mail.ru> wrote:
| Quote: | Hi all
Have recently watched Matt Jordan's session on Kamailio World 2014
On slides 26-29 of his presentation
(http://www.kamailio.org/events/2014-KamailioWorld/day1/09-Matt.Jordan-Asterisk12-And-PJSIP.pdf)
he speaks about a (completely new, for me at least) approach to build
scalable telephony systems, using N instances of Kamailio and N instances of
Asterisk
Are there any whitepapers, howtos, "implementation experience reports",
whatever, available, that would describe such an approach in details and
help some not-so-advanced admins to at least understand "if is it what they
need, or not exactly, or not at all" ?
We are planning to look closer at Kamailio (or any other proxy, like
OpenSip) as a way to do both load-balancing and failover solutions, so that
refusal of any Asterisk instance should have minimal possible effect on the
overall system availability.
|
The best documentation out there - that I'm personally aware of - is
Daniel's guide on integrating Kamailio and Asterisk:
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
While there have been quite a few improvements made in Asterisk (and I
imagine, Kamailio as well) since that was written, that guide would be
a good starting point, regardless of the versions involved.
| Quote: | A lot of questions howevere arise, like: what if one SIP user got REGISTERed
at Server 1, and the other on Server 3, so how can they call one another ?
|
There are many different ways of handling this.
First, you have to ask yourself what you want Asterisk and Kamailio to
do in your set up. Some sample questions:
* Who acts as the registrar?
* Who manages subscriptions?
* Should each Asterisk server have a special purpose, or should they
be treated as a generic pool of media servers?
* Should Asterisk be involved in 'normal' calls (two-party, no media
manipulation), or should it only be used when special services are
needed?
Your goal, in any scenario, should be to keep the Asterisk dialplan as
simple as possible. That typically means not placing customer specific
logic in the dialplan, but instead relying on func_odbc to pull
customer specific information from a database.
|
| Quote: | In later versions (such
as Asterisk 13), you can remove much of the logic from the dialplan
and use ARI to build custom media applications.
But no, not a lot of this is written down yet.
| Quote: | Also, outbound registrations can be done from one instance at a time, say
it's done from Server1 for Trunk1, so how can users, that got authenticated
at Server2, call thru that registration (Trunk1) ?
|
If your Asterisk servers are sitting behind Kamailio, they should
probably just be registering to their Kamailio instances. Again, if
Kamailio is handling the registration, identification, and
authentication, then you probably don't want Asterisk doing any of
that. You would instead just have Asterisk "trust" that Kamailio is
sending it the right calls, and have it handle them accordingly.
| Quote: | Also, Kamailio itself has to be protected from failing, and probably even
from overload...
|
That's pretty standard stuff for Kamailio.
| Quote: | Would be great to read something in-depth about that
|
|
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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