Analyzing SIP connectivity

As we learned in the previous recipe, SIP (RFC 3261 and various extensions) is a signaling protocol that is used for creating, modifying, and terminating user sessions between one or more participants. While sending SIP requests, the session parameters are sent via SDP (SDP, RFC 4566) which enable users to agree on a set of compatible media types between them. When sessions are created, the voice or video is carried by RTP and optionally controlled by RTCP (RTCP is optional, and can be used by multimedia applications, but it is not a mandatory protocol).

SIP defines endpoints as User Agents (UAs), and the process of creating a session involves UA negotiation in order to agree on a characterization of a session that they would like to create. For additional services such as locating session participants, registration, call forwarding, and others, SIP defines network hosts called servers to which UAs can send registrations, invitations to sessions, and other requests...