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?
Centralised (although it will also work with phones that do end node conferencing, but will treat the calls as normal calls).
Direct media is only used if:
direct media is enabled for both parties
there is a compatible sub-set of codecs
no services requiring inspection or monitoring of the RTP stream are in use
both end points actually honour a re-invite into a direct media configuration
Note that Asterisk is not a SIP proxy, it is a back to back user agent.
I’m not sure that Asterisk will back out direct media from the other side if one side rejects the re-invite. If the rejecting side cannot cope with incoming direct media, you will need to disable it in the configuration, for that side.