SIP (Session Initiation Protocol) is a communication protocol for establishing, modifying, and terminating voice and video call sessions over IP networks. Standardized in IETF RFC 3261, it is the de facto standard for VoIP and underpins modern telephone infrastructure.
SIP is a text-based protocol similar to HTTP, exchanging human-readable messages. Key methods include INVITE (call request), 100 Trying (processing), 180 Ringing (ringing), 200 OK (answer), ACK (answer confirmation), and BYE (disconnect). Importantly, SIP handles only call "signaling" - actual voice data is transmitted separately via RTP (Real-time Transport Protocol). This separation of signaling and media is a defining design characteristic of SIP.
SIP trunking is a service that connects enterprise phone systems to carriers over IP lines. Traditionally, PBX systems were connected to carrier exchanges via physical trunk lines (analog or ISDN lines), but replacing these with SIP trunks enables cost reduction, flexible scaling of line capacity, and consolidation of numbers across multiple locations. Most cloud PBX services are built on SIP.
A security concern with SIP is that unencrypted SIP communications are vulnerable to eavesdropping and tampering. The recommended configuration encrypts signaling with SIP over TLS (SIPS) and voice data with SRTP. Additionally, SIP brute-force attacks against SIP servers have been reported, where corporate phone lines are hijacked to make mass international calls. As the foundational technology for IP phones and VoLTE, understanding SIP is essential for anyone working with telephone infrastructure.