Does RTP Always Go Through Asterisk?
Understanding SIP and RTP in Asterisk In a standard VoIP setup, SIP is used for signaling—establishing, managing, and terminating the call. Once the call is established, RTP (Real-Time Transport Protocol) carries the actual voice media. By default, Asterisk can either relay the media (RTP flows through Asterisk) or allow direct media, where RTP flows peer-to-peer … Read more