[asterisk-users] Skype + Asterisk

[Michael Graves]

On Fri, 14 Sep 2007 18:53:53 +0000, John Meksavan wrote:

>Alejandro,
>
>  Thanks for replying.  I did come by this website before.  I was just 
>wandering, if anybody actually tried Skype with Asterisk.  My 
>experimentation with the Sip Protocol and Asterisk is at end because I  
>could never get QOS with any sip provider, ie Broadvoice, Vitelity, and 
>Teliax, when connecting directly to the "General Internet".
>
>  In my past experience, Skype has been the only VOIP that works very well.  
>If I could just integrate this with my Asterisk at work, it would really 
>make my boss happy.

Huh? In my experience QOS is that it's enirely something I have to deal
with....not the upstream providers. It matters most with respect to how
I manage outbound bandwidth. Other traffic across my router can cause
trouble for the voip service if I don't ensure adequate bandwidth. 

You simply can't get assured QOS over the internet. But the problem
usually isn't the internet, it's the edge. For me QOS tagging has been
less usefull overall than "traffic shaping"...which is an edge process
for managing bandwidth at the user-end router. 

Skype is seriously problematic since it will use various ports in an
unpredicatable fashion. That makes it difficult to manage its traffic
since you don't know where its acting.

I've tried two software programs intended to interface Skype to SIP
(PSGW 3.0 & ?? ) and they both had problems. Most noticably high
latency since they used the Skype API to take the call back to baseband
audio then re-encoded it into a SIP call.

I've found that using traffic shaping and G.792 codecs makes SIP very
reliable for a small office on a decent DSL connection.

Michael


--
Michael Graves                           mgraves at pixelpower.com
Sr. Product Specialist                          www.pixelpower.com
Pixel Power Inc.                                 mgraves at mstvp.com

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