[asterisk-users] Skype For Asterisk (SFA)

[Kevin P. Fleming]

On 11/16/2011 10:44 AM, Gordon Henderson wrote:

> The other thing - LAN to LAN calls STAY ON THE LAN! So I can "Skype" my
> wife next door and it doesn't use up any of my own broadband bandwidth
> wheras if I use a hosted SIP service, calls go out & come back in again.
> Skype also seems to be able to run the lines at max. rate too - some
> sort of adaptive bandwidth - we get large and high resolution video
> calls from one end of the country to the other with the output bandwidth
> running at near max (800Kb sec in our case)

As SIP endpoints (servers, phones, etc.) get upgraded to support the ICE 
NAT traversal mechanism, this will start happening for regular SIP calls 
as well. This *should* already happen with the Blink softphone, for 
example, since it fully supports ICE.

Also note that you are using the term 'calls' when you really mean 
'media streams'; in all of the cases you outlined, the 'call' signaling 
still follows the same path it did originally, but the media stream path 
can be shortened if the two endpoints are able to exchange media directly.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org