SIP stands for Session Initiation Protocol. It sets down the rules for enable devices and clients to communicate by transferring data between each other.
It’s similar to HTTP, which is the protocol used by the world wide web. The difference is, SIP governs how data packets for communications – for example, Voice over IP (VoIP) telephone calls – are handled over the internet.
SIP initiates a session over the internet, enables software to manipulate it the session and closes it when it’s no longer needed.
SIP is used for a wide range of modern communications, such as VoIP, multi-media conference calls, web chat sessions, instant messages, click-to-dial on web pages and more.
What is SIP Trunking?
SIP Trunking is the service offered by carriers that routes and transfers voice and data from the PSTN (Public Switched Telephone Network) to a PBX. SIP Trunking uses the SIP protocol to control calls as they travel between the PBX and Public Switched Telephone Network.
Benefits of SIP Trunking
- Reduce internal call costs – ethernet SIP lines cost much less than ISDN lines and offer more scalable bandwidth. And you can route internal calls between sites over your WAN
- Lower call rates – typically, SIP trunking call rates are lower than ISDN services
- Reduce costs of CPE equipment
- Greater flexibility with Network convergence – Convergence of voice, data and video over a common IP network delivers economies of scale and reduces network complexity
- Unified Communications can be more efficiently delivered over SIP.