Re: [GnomeMeeting-list] Open discussion about ILS/SIP/FWD

[daniel huhardeaux <devel tootai net>]

Craig Southeren a écrit :

> On Mon, 06 Sep 2004 22:13:18 -0700
> Florin Andrei <florin andrei myip org> wrote:
>
>  
>
> > On Fri, 2004-08-27 at 06:25, Damien Sandras wrote:
> >
> >    
> >
> > > However, I planned to deprecate that system by GnomeMeeting 1.2 and to
> > > replace it by a gatekeeper/LDAP system where users would sign up for an
> > > account, register to the gatekeeper with their account info and be
> > > published on ldap.seconix.com.
> > >      
> > I betcha quite a few people are waiting for SIP support in GnomeMeeting
> > to use it with Asterisk:
> >
> > http://www.asterisk.org/
> >
> > In addition to being a SIP proxy, it is also a full-blown PBX that can
> > interface VoIP with PSTN.
> >    
>
> While Asterisk is nice for certain applications, it is far from being
> a complete solution in all situations. I think that most potential users
> of GM with SIP are *not* potential Asterisk users.
>  
They are but without knowing it. Today, when installing OpenGK or 
Asterisk, you always have to decide which soft client you will use. Once 
GM will be compatible with both, you will have a global solution. And 
more, when GM will run under windows. So I decide for my customers which 
software they will use.

> > There are already quite a few SIP soft- and hard-clients (either
> > audio-only or audio-video) that work very well with Asterisk.
> > There is some H.323 support in Asterisk, but it's rather sketchy. No one
> > seems genuinely interested in H.323 anymore these days.
> >    
>
> This is simply not correct as a general statement. 
>  
I agree

> It is true that SIP is being pushed very hard by equipment vendors (as
> it represents an ideal opportunity to make new sales) and SIP also has
> very real advantages when used in an IP-only environment. But H.323 is
> still by far the most commonly deployed protocol for trunking and PSTN
> connectivity, despite what the very vocal SIP advocates would have you
> believe.
>  
IMHO also better quality for H323.

>  
>
> > But to get back to the issue you mentioned:
> > I have a feeling that many people will simply be interested in a
> > full-featured SIP client, rather than in a SIP software that comes with
> > some kind of LDAP registry. A solid SIP foundation should be the main
> > focus, the rest is secondary.
> >    
>
> Again this is not true as a general statement.
>
> There are many applications that require a dynamic directory of
> registered endpoints, perhaps linked into existing authentication
> methods, or federated together to create a distributed network. Both SIP
> and H.323 are designed to do this - Asterisk is not.
>  
To Get LDAP info in asterisk: http://www.mezzo.net/asterisk/

>  
>
> > > Notice they also provide an outbound proxy for audio-only allowing
> > > people with symmetric NAT (linux) to transparently go through their
> > > firewalls.
> > >      
> > While there are NAT workarounds for H.323 and SIP, the NAT issue is
> > solved in an elegant fashion by IAX, the protocol developed with
> > Asterisk:
> >    
>
> The very feature that makes Asterisk very nice for PBX-style
> applications (combining signalling and media into a single data stream)
> makes it very unsuitable for highly distributed configurations. Every
> entity that handles IAX has to be both a signalling and media gateway
> which introduces both latency and complexity.
>  
Don't forget that if asterisk introduce IAX which is *really* superbe to 
avoid NAT problems, he _never_ put aside SIP, which is very well 
supported. So, for me, asterisk=IAX *AND*  SIP. And with the feature of 
being an outbound proxy, it confirm my opinion. On the other hand, H323 
is also supported (2 channels) but I would say less integrated in 
asterisk. But working well (the native one I use for sure).

My 2 cts

[...]
--
daniel