how asterisk handles conference calls? I want to get an understanding on sip level. does it use centralized, full mesh, end system mixing or uni-cast receive or multicast send media distribution model?
My second question is regarding a sip and rtp confusion. I have studied that after the call is established between two sip endpoints, the media flow between them using rtp and they don’t use the resources of the asterisk server. What if the two endpoints don’t support common set of codecs i.e. isn’t there a way that asterisk server helps a sip endpoint in converting a codec to assist the endpoint in the call? how does ChanSpy() application works if the data is passing between end points?