Session Initiation Protocol (SIP) is the open signaling standard that sets up, manages, and tears down real-time sessions—voice, video, and messaging—over IP networks. If you’re asking what is Session Initiation Protocol, it’s the way devices find each other, agree on capabilities, and keep calls alive.
How it works: user agents (phones, softphones, SBCs) talk to proxy/registrar servers using methods like INVITE, REGISTER, BYE, and re-INVITE. SIP exchanges session details with SDP, then hands media to RTP for the actual audio/video. DNS and ENUM help route calls; policies handle features like forking and transfer.
Where it shows up: enterprise VoIP, contact centers, conferencing, and cloud UC. Teams adopt SIP because it’s vendor-neutral and easy to scale globally.
- Interoperability: Open standards with broad device support.
- Routing control: Number management across sites and carriers.
- Resilience: Multiple trunks, failover, and geo redundancy.
- Security: TLS for signaling and SRTP for media (when configured).
Our take? SIP turns telephony into software—programmable, portable, and ready for cloud.
Planning carrier connectivity or a PBX-to-cloud move? Start with our SIP trunking Guide for SBC, routing, and E911 patterns, then read Planning For a Successful SIP Trunking Implementation to turn best practices into a phased, low-risk rollout.
