Glossary
SIP (Session Initiation Protocol)
SIP, the Session Initiation Protocol, is a text-based signalling protocol defined by IETF RFC 3261. It establishes, manages, and terminates real-time communication sessions such as voice calls, video calls, and instant messaging over IP networks.
General definition
SIP handles the call setup phase of real-time communication. When you dial a VoIP number, SIP negotiates who is calling, what media codecs to use, and where to send the audio or video stream. The media itself travels over RTP (Real-time Transport Protocol) or SRTP, not over SIP itself.
- Session initiation, modification (hold, transfer, codec renegotiation), and termination
- Authentication and presence via SIP REGISTER and NOTIFY methods
- Interoperability with public telephone networks through SIP trunking
- Support for one-to-one and conference sessions
SIP has been widely deployed in enterprise PBX systems and carrier VoIP for over two decades. In browser-based applications, WebRTC has largely replaced SIP for peer-to-peer calling because WebRTC works natively without plugins. SIP gateways still bridge WebRTC endpoints to the PSTN and legacy telephony infrastructure.
In the Ethora ecosystem
Ethora’s voice and video calling is built on WebRTC rather than SIP, which means calls run natively in browsers and mobile apps without additional plugins or PBX infrastructure. SIP interoperability can be layered on via a gateway when a deployment needs to bridge to a carrier network or a legacy enterprise phone system.
For organisations in healthcare or finance that operate existing SIP infrastructure, Ethora’s self-hosted deployment model gives teams the flexibility to integrate at the signalling layer through standard gateway configurations rather than replacing incumbent telephony investments.
