[opensuse] Re: VOIP solution?

[Eberhard Roloff]

Sunny wrote:
 On 3/22/07, Zhang Weiwu
 [EMAIL PROTECTED] wrote:
 On Fri, 2007-03-23 at 10:22 +0800, Zhang Weiwu wrote:

  I am thinking perhaps it's not difficult to set up some software on the
  server that do the routing, e.g. it serve as a call center that both
  office login to a VOIP software and it connects to the server, the
  server talk to both sides. This is the fastest solution and it should
  work. That's only my imagination, I am still searching for such
  software.

 Certainly THIS would work: set up VPN on the server and both office dial
 into the VPN before they start to use some SIP software. This can solve
 the problem, but I think it's over complicated.

 Besides, I never tried VPN on Linux, only did it on Windows: on windows
 the downside is once a host has dialed up VPN, local network connection
 is hidden for it, that I can no longer access the hosts in the same
 office that has not yet dialed in the same VPN. This is not acceptable
 for us.

 
 If you go with VPN solution you can use openvpn - that way you can
 setup exactly which IPs you want to route trough the vpn, and then any
 p2p VOIP software should work. Plenty of them for linux (including
 GAIM). You can setup Jabber server and use it. servers:
 http://www.jabber.org/software/servers.shtml, clients:
 http://www.jabber.org/software/clients.shtml
 
 Or you can take a look at http://www.asterisk.org/
 
For me it seems that the bandwith shortage is the first problem to solve
or cope with.

If Skype does not work solely because of bad connections to the Skype
Network, this is ok and you might try with a peer to peer client for
testing, if direct connectivity works better.
A quick google showed IHU (I hear you).  There is also an rpm for 10.2
on packman. The description says, that it works without using any
session protocol.

Otherwise, imho, if Skype does not work ok, hardly anything else will
do, especially not a SIP connection, not to mention multiple concurrent
SIP connections at the same time.

With Asterisk you get a superior PBX Solution that handles anything you
can imagine. So you might benefit from Asterix in your internal networks
at the respective locations, but, sadly, the IAX (Inter Asterix
Protocol), from what I saw, does not offer you better bandwith usage
than SIP, so Asterix might not be the obvious solution to handly your
bandwith issues.
kind regards
Eberhard

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Re: [opensuse] Re: VOIP solution?

[Dave Howorth]

Eberhard Roloff wrote:
 Or you can take a look at http://www.asterisk.org/

 For me it seems that the bandwith shortage is the first problem to solve
 or cope with.
...
 With Asterisk you get a superior PBX Solution that handles anything you
 can imagine. So you might benefit from Asterix in your internal networks
 at the respective locations, but, sadly, the IAX (Inter Asterix
 Protocol), from what I saw, does not offer you better bandwith usage
 than SIP, so Asterix might not be the obvious solution to handly your
 bandwith issues.

I agree that it looks like a good idea to put an Asterisk server in each
office. They will handle local calls within each office so you don't use
any external bandwidth. It can also connect your phone lines.

Then you need to connect the two Asterisk servers via your hosted
server. I guess you could put another copy of Asterisk on that server,
or just setup the routing tables to ensure that conections from one to
the other go via the hosted server. Details depend on exactly how that
very special technology is set up.

If you use a VPN to communicate with your hosted server then I assume
that Asterisk traffic will also use it. But if you don't already need to
use a VPN, I'm not clear of the benefit of setting one up.

Cheers, Dave
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