[asterisk-users] SIP to IAX2 with delayed echo

[c james]

Steve Totaro wrote:
> 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"
> 


I'll get the network guards involved and see.