[Asterisk-Users] Re: asterisk + OH323 + NAT + gnomemeeting

[Jesse Guardiani]

On Sun, 17 Apr 2005 21:24:30 +0200, Bruno Hertz wrote:

> Jesse Guardiani <jesse at wingnet.net> writes:
> 
>>> Wait a sec...  COME TO THINK OF IT!
>>> Why not run asterisk on your linux box that you are running GnomeMeeting 
>>> on, and use it to convert between H.323 and IAX and SIP???
>>> 
>>> After all, it is a penguin...
>>
>> That's certainly a good alternative. I'm currently in the process of
>> hacking up the latest linphone (1.0.1) to fix a few personal
>> show-stoppers. If I can get it to the point that I like it, then I'll
>> probably just go with linphone. But you're right. If it's took much work,
>> then I'll probably just start running asterisk on my laptop to do H.323 to
>> SIP conversions. Thanks for the suggestion! I hadn't thought of that yet.
>> I'd been looking at things like the commercial sip323 program, but I
>> hadn't thought of doing it with a local copy of asterisk.
> 
> If your only reason to stick to H323 is Gnomemeeting you could try
> other softphones as well. Especially, the XLite beta for Linux looks
> promising, and some people like SJphone for Linux.

I don't know about X-Lite, but sjphone seems only to support OSS. One
of my requirements is ALSA support. Thus linphone and gnomemeeting.

But, interestingly, gnomemeeting seems to be the only client capable
of full duplex audio using ALSA+DMIX+DSNOOP+ASYM.


> Also, SIP support for Gnomemeeting is underway, but development is
> slow. I'm constantly pointing out to them how much interest there is,
> but things still seem to take their time ...
> 
> Finally, on a recent discussion about the future design of GM on their
> list, I was surprised to learn that quite a few people really use it
> for direct PC to PC video calls over the internet. So somehow, after
> extensive NAT and router fiddling I guess, direct calls apparently
> work even with H323 (there is already support built into GM for
> external IP address discovery, as you know, so those remarks about
> transmission of bogus IP addresses on H323 level probably don't really
> apply in this case).

Yeah. It supports STUN too, which seems to be the silver bullet for
SIP. So I'm thinking the problem is more asterisk related. That's why
I asked about gnugk. It seems to have more NAT translation support
than asterisk, but my attempts at a working config haven't worked so
far.

-- 
Jesse Guardiani, Systems Administrator
WingNET Internet Services,
P.O. Box 2605 // Cleveland, TN 37320-2605
423-559-LINK (v)  423-559-5145 (f)
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