[Sarp-users] Re: SARP Testing

[Andrew R. <an...@ra...>]

Hi,

I won't get the chance to answer more for a day or so, so I'll give some 
quick answers now and more detailed answers later if needed.

Adrien Pestel wrote:

> Hi,
> 
>  
> 
> I would like to test your solution to make VoIP calls behind NAT.

Go for v0.2.1. I've started working on some cool new features in CVS for 
0.3 and I really don't know how well it will tie together yet. If you 
wanna play with keepalive's for instance, you can go with CVS. I usually 
try to avoid uploading anything that doesn't work :-).

Just be aware that it is still very much development. For instance all 
status messages are sent to the console until I get a good 
cross-platform loggin system in place. That will hopefully be in for 0.3

> I’ve got few questions.
> 
>  
> 
> 1) What UA are compatible with your solution (Linux & Windows if possible)

I'm aiming for 100% compatibility (if that's even possible :-( ). So far 
tested UAs are X-Lite and SJPhone. I don't have any hardware devices for 
the time being. Personally I prefer X-Lite especially since it uses the 
Speex codec which is _very_ impressive! Both of these are Windows 
programs since I haven't been real happy with any of the Linux clients.

> 2) Is it compatible with a SIP Porxy and/or SIP registrar

Dunno. Keep meaning to look into this. I really can't see why not. It 
does have a registrar built in and for what I want works perfectly. 
You'd be talking about FWD or iptel?

> 3) Imagine Endpoint1 is behind a NAT and Endpoint2 is behind another 
> NAT. Is it possible to make a Call like Endpoint1 call Endpoint2 ?

Certainly! In fact we use it that way for communication during 
development. Almost everyone I know is behind a NAT of some kind and it 
seems none of can afford all the Cisco gear a lot of SIP developers 
assume you should have :-).

As to whether it will work if you have the following:

UA1 <-> SaRP <-> NAT <-> NAT <-> UA2

That will depend a little on your NAT and a little on UA2. For instance 
there is no reason it won't work if UA2 is SJPhone and calls originate 
from it since it sends RTP data on the same port as it receives and 
therefore most NAT devices will pass the audio through in both 
directions (SaRP does the same thing). X-Lite sends RTP from a seperate 
socket and therefore different port to what it receives and therefore 
doesn't play nicely with any NAT. This is infact something I keep 
meaning to raise with them.

> Thank you for your answers.

No problem. I'd be very happy to hear your comments and experiences. If 
you have any problems let me know.

> Cheers,
> --
> 
> Adrien Pestel (EPITA/3IE)

Regards,

Andrew Radke