short question : does Asterisk reserve RTP ports for every IP-phone that is being called ?
If for instance an incoming call makes 10 IP-phones ring, does this mean that Asterisk preserves 10 x 2 RTP ports for audio ?
I guess Asterisk sends in the SIP INVITE an SDP body with an RTP port number for audio ? If this is the case for the 10 IP-phones to which an INVITE is send to, this means at least 10 RTP ports are reserved for incoming audio, correct ???