Greger V. Teigre wrote:
Paul, I fully support the approach: Make one
reference design with a
complete ser.cfg. This will give us a Getting Started. We can
later
add sections on the more advanced stuff like redundancy, radius,
etc.
Thanks for your review of the components in such a reference design
(I'll relate to those further below).
I believe there are two hurdles to get on top of ser: Get a first
working config up and running and then understanding the concepts
good
enough to start tweaking. Many will not have all the components of
the full reference system you describe, Paul, so a starting point
with
a minimum system is probably needed. I.e. Get a UA registered
without
auth, etc (I see some questions on this too)
I'd like to add a third hurdle, keeping this or any documentation
up-to-date. One of the biggest issues
I've faced is keeping a working, production supporting, configuration
"correct" across release changes.
The situation doesn't get better if there is alot of out dated
documentation.
In addition to a few core examples I'd suggest a clearly worded
changelog. The changelog needs to
be clearly show what has changed and what is impacted by the change
on a
release by release basis.
$0.02
I thus see the following things that must be
addressed:
- How to read the basic ser.cfg
- The basic ser.cfg, what does it do, what is the reference design
(is
the ser.cfg in cvs appropriate?)
- A description of the reference design with a "component list"
- The complete ser.cfg
- Conceptual explanations of each logical part of the ser.cfg
- External systems (Asterisk, mediaproxy/nathelper), configs, etc
See my inline comments with regards to a reference design.
> My setup uses SER v0.9 and Asterisk-1.0.2. The Asterisk server is
> used
> __ONLY__ for voicemail because - well lets face it, Asterisk sucks
> as
> a SIP router because it just isn't designed to be one.
>
> So all users are managed by SER and Asterisk only comes into play
> for
> voicemail and for playing recordings such as "the party you are
> calling has blocked your call" when a call block is enabled.
We also use 0.9, but does not yet support voicemail. I think we
should concentrate on 0.9 capabilities and forget about 0.8.14.
Most
people starting up now will probably use 0.9, at least shortly when
it
is released as stable.
Voicemail adds a layer of complexity in terms of scalability and
redundancy. IMHO we should leave out voicemail from the reference
design, not because it is something most people would not want, but
because it introduces an external component and complexity that is
better added later in the document (like redundancy). That being
said,
I think we should include voicemail and voiceprompts as part of the
initial work on this document, just not leave it as the main
reference
design.
Sems is a bit less complex than Asterisk and uses the same style
config, could it be an alternativ in the reference design?
> We should leave redundancy out of the picture for now because fault
> tolerant SER is still something many users don't use and it's
> something that is still maturing in SER. Besides, my opinion on
> this
> matter is that a "ser clustering" should be a product of the Linux
> HA
> technologies (expect for registration functionality).
Yes, I agree that we should leave redundancy out. Using Linux HA
does
not necessary make it simpler... Also, in order to get network
redundancy when you have distributed users, you need geographic
distribution of ser servers. But, again, the complex stuff should be
left until later.
> The ser.cfg we present should also show how to use MySQL for
> accounting, usrloc, etc.
Agree. We use RADIUS-based authentication and authorization with
distributed RADIUS servers. Only usrloc is stored in mysql (we use
avp_radius_load), but we do accounting to mysql. I can maybe
volunteer to do a RADIUS-section later, covering auth, uri, avps
etc,
but we should concentre on the basics first.
> serweb should be avoided altogether because this is nothing more
> than
> a reference implementation that I believe not a primetime offering,
> again, just my humble opinion.
Agree. But, maybe somebody will volunteer to add an add-on section
on
serweb?
> Failover PSTN gateways must be covered as well as NAT traversal.
> The
> NAT traversal I use is mediaproxy because it seems to just work
> better, especially in distributed deployments.
NAT Traversal, I agree. Failover PSTN GW is a more advanced option.
Especially if that means introducing the new lcr module from cvs
head.
:-)
> On this NAT note, my ser.cfg only proxies RTP streams when one or
> more
> SIP clients is behind a NAT firewall. The exception to this is
> when a
> SIP client needs to hit the Asterisk server. The reason for this is
> that the Asterisk server is physically a differenet machine that
> does
> not have direct access to the internet. Instead I use the SER
> server
> with two (2) ethernet interfaces, whereby eth0 is the public
> interface
> and eth1 is a 10.0.0.0 private network and I use a crossover cable
> to
> the Asterisk server, which has only one private 10.0.0.0 interface.
We use rtpproxy where ser is located on one server and the rtpproxy
on
another. They communicate across udp (inside an ipsec tunnel). I.e.
they are geographically distributed to keep the rtpproxy server as
close as possible to the subscribers.
Our UAs are configured with STUN and the RTP streams are only run
through our proxy server if an UA is behind a symmetric NAT and gets
an incoming conversation (or both are behind symmetric NAT).
Calls where both UAs are behind the same NAT will always be forced
through the rtpproxy (to avoid hairpin problem).
> Since almost all serusers seem to be interested in voicemail I'd
> suggest detail instructions on the Asterisk integration. I use the
> ast_data patch, which is kindly provided by bwsys because this
> makes
> managing Asterisk mailboxes a function of the MySQL database. And
> the
> only other real hard part to Asterisk integration is the Message
> Waiting Indicator, which I have modifed the app_voicemail.c file in
> Asterisk to handing SUBSCRIBE messages a bit differently and I use
> sipsak to send NOTIFY messages back to SER, which then proxies the
> NOTIFY message to registered SIP clients to turn their MWI on or
> off.
IMHO, this is not a basic reference design, but rather advanced...
;-) Of course, there are many people who would love to see this
design described.
> Call features should also be covered in the ser.cfg. Things like
> call
> blocking, speed dialing, click2dial, etc. Things like 3-way
> calling,
> call waiting, etc should not be covered because they are functions
> usually implemented as IAD features.
Agree.
> Our company has a full tcp/ip networking patch that we plan to
> release
> to the ser project. This tcp/ip patch gives us full FIFO
> functionality
> as a TCP socket, and this is something we hope would be commited to
> CVS and maintained in the core. As far as we can tell the
> networking
> patch is stable, but we need to prove this further.
Good news! You have probably seen Andreas' effort in this same
direction using XMLRPC. I guess you have patched the core like Juha
suggested in the XMLRPC dialogue? This is an area where a lot of
parallel work can be avoided.
> So in closing, if anyone things we're better off coming up with a
> ser.cfg in private, then let me know. If everyone things that the
> serusers list is the place to do this then lets start for the
> benefit
> of everyone!
If you start out by making an initial draft by dumping in you config
and making some headers, you can send it to us for adding content.
If
you submit it on the list with a call to submit suggestions and
wishes, we can either rotate the document edit privilege or work on
different parts of it?!
Best regards,
g-) aka
Greger V. Teigre
AxxessAnywhere, Oslo, Norway
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