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

[Diana Cionoiu]

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, is it right?
>
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