Steve Totaro wrote: > On Thu, Nov 20, 2008 at 1:13 PM, c james <[EMAIL PROTECTED]> wrote: >> Steve Totaro wrote: >>> Just for sh1t$ and giggles, try sip to sip and drop the IAX piece. >>> IAX2 is not all it is cracked up to be. >>> >>> Also, do a ping to see latency, 200ms is pretty much my standard. >>> >> Coming from outside the network, setting up for a couple rounds of >> NATting isn't going to work well. They are not seeing it between >> phones. Others, using the polycom phones have reported echo between two >> SIP on a 4ms ping trip. >> > > NAT is manageable with OpenVPN and very easy. You just need a box on > both sides. > > Also, a more difficult setup will allow SIP to work through NAT if > both sides are behind a NAT. I just prefer OpenVPN because it is set > it and forget it. > > Anyways, it is quite simple to switch to SIP to test. IAX2 has made > me quite a bit of money because of it's "issues", where SIP "Just > Works" >
I'll get the network guards involved and see. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
