Perhaps, but I think we really need to focus on ser-0.9 rather than the unstable cvs code. The exception might be the LCR module because it is an extremely important piece of functionality which can in many cases make a CLEC / VoIP player profitable or not.
Anyhow, to remain focused I think we should focus on ser-0.9 only.
P
On Mon, 21 Feb 2005 22:29:14 -0000, Simon Miles simon@systemsrm.co.uk wrote:
What I have struggled with recently is the introduction of new modules into the code - such as avp. Even now I'm not sure if I should be using it and for what reason.
So a description of the modules as we have them today would be useful.
Can we assume that what is in the dev tree at the moment is what will be released soon? I did see some emails about this topic.
Simon
-----Original Message----- From: Java Rockx [mailto:javarockx@gmail.com] Sent: 21 February 2005 19:59 To: Simon Miles Cc: Greger V. Teigre Subject: Re: [Serusers] "Best practice" document,was Warning: sl_send_reply: I won't send a reply for ACK!!
Outstanding.
If you guys give me a few days I'll put together a rough document of my ser.cfg, Asterisk setup, patches, etc, etc and then we can start stripping it apart to generate something useful.
P.S. - have you guys tried gmail? It absoutely kicks ass when dealing with long threaded conversations such as these. Anyone wants a gmail invitation just let me know. I have about 200 I can give out.
On Mon, 21 Feb 2005 19:28:07 -0000, Simon Miles simon@systemsrm.co.uk wrote:
Great - I'll try and knock something together tonight ( in word ! ! )
Simon
-----Original Message----- From: Java Rockx [mailto:javarockx@gmail.com] Sent: 21 February 2005 18:48 To: Simon Miles Cc: Greger V. Teigre Subject: Re: [Serusers] "Best practice" document,was Warning: sl_send_reply: I won't send a reply for ACK!!
I can agree to this. We can tweak our direction as things progress.
As for MS Word --- there's no Windoz here so Open Office will have to be good enough. :-)
On Mon, 21 Feb 2005 18:24:17 -0000, Simon Miles simon@systemsrm.co.uk wrote:
Paul, Greger,
I think we are all about on the same page. I do prefer the approach for a 'getting started' document to start simple and lead to a more complex environment.
So based upon Greger's ToC, can I suggest the following for the sections :-
Getting Started - what is ser and how does it workReference Design I would like to see here a picture of the complex designthat we will build to, but then have a picture that shows the simple design
of a UA <> NAT <> ser / mediaproxy
Ser.cfg A config file with line numbers that describe each sectionand
why we have it and what it does.
Ser in Action * Based upon the above ser.cfg, describe how ithandles
registration / INVITEs * Handling of NAT Describe a brief summary of the options and why we use
mediaproxy * Basic Call Features * On Failure Handling * Billing / Accounting
Adding Multiple Domains Describe how to configure multiple SIP serversAdding Asterisk Support for voice messagesAdding RADIUS SupportControlling ser with XML / Web Services / serwebAppendix A The complete ser.cfg
Appendix ? Other php etc support files
Appendix ? How to download the latest version of ser ( from the
dev
tree ?? )
How does this look.
I agree about using Microsoft word format ( I use Office as I haven't bitten the bullet about OpenOffice yet ! )
If we agree I'll start a word document off.
Simon -----Original Message----- From: Java Rockx [mailto:javarockx@gmail.com] Sent: 21 February 2005 14:49 To: Greger V. Teigre Cc: Simon Miles Subject: Re: [Serusers] "Best practice" document,was Warning: sl_send_reply: I won't send a reply for ACK!!
Hi All.
Let me toss out this idea as a way to keep us from loosing site of the
ultimate outcome - which I believe is this;
A novice/newbie must be able to read our documentation and get a complete (and complex) SIP proxy with voicemail and NAT traversal. Not
to mention call features, etc.
So my idea is for us to begin with a complete ser.cfg/Asterisk/mediaproxy SIP proxy and remove parts to get to the
"Getting Started" ser.cfg which is super simple. Then do a step by step guide to reconstruct the full SIP proxy.
This way novices/newbies can follow along and we can document features
as we use additional bits of functionality.
We can also postpone the more advanced aspects until later in the documentation.
Thoughts??
Paul
On Mon, 21 Feb 2005 15:34:32 +0100, Greger V. Teigre greger@teigre.com wrote:
Simon, I agree. I suggest that we use Paul's previous email with components as a starting point. We can also save some time by starting out with Paul's config and strip it down to where we want
it. I vote for making
the reference design as simple as possible in the beginning and then
rathe elements one-by-one.
Diagram suggestion:
UA | NAT | ser --- mediaproxy | (asterisk)
I suggest adding asterisk as an addon and not as part of the initial
reference ser.cfg. Getting Paul's asterisk config to work is far too
big a hurdle.
Rough suggested table of contents:
- Getting started, how to read ser.cfg, ser architecture (very
short)
- Reference design, functionality overview 3. ser.cfg reference
config file indexed with line numbers and sections 4. Basic message routing and structure of ser.cfg (incl. record routing and loose_route) 5. Registration and INVITE authentication (incl. spoof control, external INVITEs) 6. Handling of NAT 7. Basic call features (call fwd, etc) 8. On failure handling 9. Handling local vs. PSTN calls (forwarding) Appendices
- Voicemail
- RADIUS
- serweb
Do we work in word? g-)
Simon Miles wrote:
Can I suggest that the three of us start this off and produce this
Getting Started guide.
I don't think that designing this committee of the masses will be a benefit.
So we first need to agree the reference design and then construct a ser.cfg to support it.
If this is correct then maybe we should just list the basic concepts
of the design and then draw a diagram that we agree on.
Simon
-----Original Message----- From: Greger V. Teigre [mailto:greger@teigre.com] Sent: 21 February 2005 08:43 To: Java Rockx; Simon Miles Cc: serusers@lists.iptel.org Subject: Re: [Serusers] "Best practice" document,was Warning: sl_send_reply: I won't send a reply for ACK!!
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 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