SIP Trunks
Future-proof your communications with SIP Trunks. By connecting your on-premise PBX to the public telephone network via the internet, you can enjoy a scalable, cost-effective, and flexible telephony solution. SIP Trunks also enable unified communications, allowing you to use your system for voice, video, and messaging, perfect for multi-site and remote working.
A SIP Trunk is a virtual communications link that connects your organisation’s on-premise IP-enabled Private Branch Exchange (IP PBX) to an Internet Telephony Service Provider (ITSP) over a data network. It is a critical component for businesses migrating to Voice over Internet Protocol (VoIP) and serves as the modern, digital replacement for traditional ISDN lines.
Technical Operation
A SIP trunk functions through a series of steps to establish, manage, and terminate communication sessions:
The process relies on the Session Initiation Protocol (SIP), a robust signaling protocol defined in RFC 3261. SIP is used to “set up” a call, defining the call parameters, codec information, and the endpoints involved.
Unlike the circuit-switched method of ISDN, SIP trunks use a packet-switched network (i.e., the internet). Voice, video and other multimedia communications are broken down into digital data packets and sent over a data connection.
While SIP initiates and manages the call, the actual voice and video data packets are carried over the network using the Real-time Transport Protocol (RTP). For secure sessions, the Secure Real-time Transport Protocol (SRTP) provides encryption.
A SIP trunk is comprised of multiple virtual “channels” that determine the maximum number of concurrent inbound and outbound calls your system can handle. This is distinct from ISDN, where capacity was fixed to physical lines