[asterisk-users] SIP to IAX2 with delayed echo

[Steve Totaro]

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
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)