[VOIPSEC] Really RTP is end-to-end?

[Nicolas BELAN]

Hi,

This depends on asterisk and x-lite versions.

If your softphone (or even your hard phone) is able to handle re-INVITEs, 
you can save your asterisk bwidht.

You may take a look at 
http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite

Asterisk will send reinvite packets - but it do not make any assumptions 
about softphone capabilities when sending reinvte ..

Just try and listen !

Nicolas. 

Voipsec-bounces at voipsa.org a écrit sur 17/07/2006 10:26:19 :

> Hello,
> 
> The problem is that usually asterisk dosen't do that because of the 
> internat design. Asterisk usually gets audio and video agregated with 
> the signaling path. However in the case of SIP it can send an reINVITE 
> to x-lite to tell him to send the RTP to the other x-lite. But i have no 

> idea how to use asterisk to do that since i've quited asterisk a long 
> time ago in favor of Yate, which does that by default by using rtp 
> forward function.
> 
> Diana
> 
> rgigli at libero.it wrote:
> 
> >Hi,
> >I have two x-lite softphone, A and B, and an asterisk server, C, 
> when A call B initiates a SIP session whit C. My problem is in RTP 
> session because A and B don't communicate in end-to-end manner but 
> through C. I think there is a problem in my x-lite configuration, isit 
right?
> >
> >
> 
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