I’m trying to separate the audio from video in asterisk, a device that gets the audio and video on another device. Can anyone help me where I can change in Asterisk?, Or is it better to create a new application, AGI, AMI, SIPSERVLET …?
This will require major surgery to the source code. You are probably going to have to a lot of rewriting of the channel technology driver and the Dial application.
My feeling is that, as you needed to ask, you don’t have the level of knowledge of Asterisk internal needed to take on that job. Also the fact that you asked as support question on a discussion forum is not a good start.
If you really want to do this, you need to use a developer support resource, not an end user one.
Note that SIP can negotiate end points with different addresses from the signalling end point, and from each other. I think the latest bridging code can support this (note that the peer dictates the split, not Asterisk). However, I think you want to set up a call with different signalling end points, and that is what requires deep changes.