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" -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) _______________________________________________ -- 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
