[Serusers] "Best practice" document, Business system open source

[Charles Wang]

lakmal, Paul and ALL:

I heard your discussion about best practice of ser sip (maybe 0.9.0 or
above) for a long time. And you also said the documents would be
posted for everybody to enjoy it. Could you please tell me where to
find the documents?

Best Regards
Charles


On Wed, 9 Mar 2005 05:45:11 +0100 (CET), lakmal silva
<ruwan_lakmal at yahoo.fr> wrote:
> Hi Hans,
> 
> This is a great thing. Can't we place this BSS code at
> berlios? And then  may be put a link from iptel.org
> site under 3rd party software. As Alexey willing to
> set up a SER community site may be you can place the
> docs there as well.
> 
> By the way are you involved in Yxa and the minisip
> developments at KTH?
> 
> Regards,
> 
> Lakmal
> 
> 
> --- Hans Eriksson <hans at hecc.se> wrote:
> 
> > Guys,
> >
> > this discussion faded away. Is it still hot but
> > carried out somewhere
> > else?
> >
> > Anyway, I am committed to put my business support
> > system in the open
> > source. We use it both in projects at Royal Institue
> > of Technology
> > (KTH), Stockholm, as well as commercially
> > (www.xtrafone.com). The BSS
> > handles order, customers, accounts, rating, billing
> > (pre-paid),
> > customer My Pages, etc. The lot!
> >
> > KTH use SER as its proxies wheras we use the
> > Asterisk in
> > www.xtrafone.com. As long as there are CDRs placed
> > in a MySql tables,
> > the BSS can pick that up and rate, charge.
> >
> > I have also written a couple of How-To (install ser,
> > mysql etc).
> >
> > All this I'd like to add to the pot. I am all for to
> > create a complete
> > package for a commercial or non-commercial VoIP
> > operator that includes
> > a proxy, gateway and BSS. And also the docs
> > describing best practices
> > (ser.cfg, logging, NAT traversal, etc). Just add
> > marketing and
> > customers and you roll.
> >
> > For the BSS I don't really know where to place the
> > code. Beside the
> > ser? sourceforge? It is written in /bin/sh, awk, SQL
> > and php (no,
> > nothing to compile!). Runs on any Linux, MacOSX and
> > FreeBSD system.
> >
> > Can we get this discussion thread going again and
> > get started putting
> > our stuff into a shared pool where we can get going
> > to change the world
> > (I just could not hold back :-).
> >
> > /hans
> >
> >
> >
> >
> > 2005-02-21 kl. 14.28 skrev Iqbal Gandham:
> >
> > > Great idea, can I also suggest, and I can help out
> > if needs (simply
> > > because I am one of them struggling users :-)) is
> > a debug guide, all
> > > devices seem to have a few quirks to the setup,
> > and they all seem to
> > > have different setting, eg some support only STUN,
> > some use the
> > > proxying, others you can dela with at the server
> > end etc etc, what I
> > > think owuld be useful is a guide to what its
> > supposed to look like,
> > > i.e the debug log.
> > >
> > > I have been through the entire sip syntax , to
> > figure out where the
> > > messages go/come from, but with the contact
> > headers, From, and c= I
> > > can see how it can get a little confusing
> > >
> > > Iqbal
> > >
> > > PS 0.10 works quite nicely
> > >
> > > Java Rockx wrote:
> > >> Steve,
> > >> I fully agree - and this is the exact reason that
> > this cannot be a
> > >> single person endeavor.
> > >> Regards,
> > >> Paul
> > >> On Mon, 21 Feb 2005 07:28:02 -0500, Steve Blair
> > >> <blairs at isc.upenn.edu> wrote:
> > >>> 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
> >
> === message truncated ===
> 
> 
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