Jesse Guardiani wrote:
It should be possible with h323, if you have control over the NAT points, with linux iptables/netfilter and the h323 NAT and conntrack helper modules. (netfilter.org patch-o-matic) And you stated you don't have control over one of the NATs, so that'd be out anyway. (better to go a different direction anyway, IMHO) And yes, SIP has the same problem, although many clients (hard and softphones) and * can usually compensate for this.On Sun, 17 Apr 2005 01:39:09 -0400, Karl J. Vesterling wrote:
H.323 will not traverse NAT.
Sorry... I know, I was a big proponent of it when H.323 was the only "standard" VoIP protocol out there. Probably because when it came out NAT wasn't even thought of.
The problem is that the control channel in H.323 discloses the internal IP address, and the various connections attempt to connect to each other. So you wind up with problems like audio only in one direction, etc...
I thought SIP had the same problem though. Can't this be solved with
address translation inside asterisk? You know, like the externip,
localnet, and nat=yes options in sip.conf?
Wouldn't it be simpler (and less resources) to set up an openvpn tunnel between the client and the * box? (since you're talking about softphones - for hardphones obviously you'd need to tunnel from another box then NAT or bridge) With openvpn or another vpn/tunnel solution, you can either bridge the client and asterisk LANs, or just create 1-1 tunnels from the client machine (if it's a softphone) to the * box. Either way you don't need to worry about NATs. (I'm doing this now for one of our hardphones,with an openVPN tunnel between linux gateway routers at each end.)Or is it simply impossible due to limitations within the H.323 spec? It's difficult to find information about this sort of thing on the internet. H.323 is such a broad spec...
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.
j
disclaimer - I know linux routing and firewalling, but only have a few months exposure to VOIP...
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