IP telephony is the concept of converting analog voice calls into IP packets and steering them over the IP network. Call signaling protocols such as SIP will be used to establish the call session between the endpoints and will then leverage Real-Time Transport Protocol (RTP) as the application layer protocol to transfer the media streams over the IP network. The audio (or video) packets will be encapsulated with RTP header and typically run over UDP.

In this recipe, we will see the normal operation of IP telephony and how RTP and RTCP are used for an end-to-end audio stream using Wireshark-based capture analysis.