[asterisk-users] SIP to IAX2 with delayed echo
stotaro at totarotechnologies.com
Thu Nov 20 12:24:21 CST 2008
On Thu, Nov 20, 2008 at 1:13 PM, c james <cjames at callone.net> 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
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
+18887771888 (Toll Free)
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