Session Initiation Protocol
SIP is the PSTN of the internet. Every VoIP platform (Twilio, Vonage, AWS Chime, Asterisk, FreeSWITCH) speaks SIP. Building any communications product requires understanding INVITE flows, registration, and codec negotiation via SDP.
What It Defines
SIP is the signaling protocol for initiating, modifying, and terminating multimedia sessions (voice, video, messaging) over IP. Defines INVITE/ACK/BYE/REGISTER/OPTIONS/CANCEL methods, the SIP URI scheme (sip:, sips:), dialog management, Via/Contact/To/From/CSeq header semantics, and proxy/registrar/redirect server roles.
Canonical (Normative)
Convenient (Practical)
Related References
The canonical publication point for finalized RFCs. If a protocol is standardized as an RFC, the RFC Editor text is the normative final reference. Published by the IETF, IRTF, IAB, and independent stream.
Related Specs
SDP is the handshake language between any two endpoints setting up a call. WebRTC, SIP, and RTSP all use it. You can't debug codec mismatches, ICE failures, or video call setup without reading SDP.
RTP carries all real-time media: VoIP, WebRTC, live video, and video conferencing. Understanding SSRC demultiplexing, jitter buffers, and RTCP feedback loops is essential for any media application.
Any media you send over the internet should be encrypted. SRTP is the protocol. In WebRTC it's automatic via DTLS-SRTP; in SIP deployments you must configure it explicitly. Know it when auditing VoIP security.
SIP over WebSocket is how browser-based softphones and WebRTC-SIP gateways work. If you're building a web-based communications app that bridges to PSTN or SIP trunks, this is the transport spec.
Every phone number your app touches — Twilio, AWS SNS, WhatsApp, SIP — uses E.164. Getting number formatting and parsing right (libphonenumber patterns) requires understanding CC/NDC/SN structure.