SIP trunking is a VoIP technology that connects a business telephone system (PBX) to the public switched telephone network (PSTN) via the internet using the Session Initiation Protocol (SIP). Rather than using physical copper phone lines, SIP trunks carry voice traffic over IP infrastructure. A single SIP trunk can support multiple concurrent calls, replacing the need for traditional ISDN lines. SIP trunking enables businesses to scale call capacity on-demand, reduce per-minute costs, and consolidate voice infrastructure.
SIP trunking connects your on-premises or cloud PBX to the PSTN over the internet using the Session Initiation Protocol. Your PBX sends SIP signalling messages (INVITE, ACK, BYE) to the SIP trunk providerโs server, which routes the call to the destination number via carrier infrastructure. Voice is carried as RTP (Real-time Transport Protocol) packets over your IP connection. You configure your PBX with the providerโs SIP credentials (server address, username, password) and define inbound/outbound routing rules.
VoIP is the broad technology of transmitting voice over IP networks. SIP trunking is a specific implementation of VoIP that connects a business PBX to the telephone network. Think of VoIP as the category and SIP trunking as one product within it. Other VoIP implementations include hosted PBX (where the phone system is in the cloud), softphones (software-based desk phones), and WebRTC (browser-based calling). SIP trunking specifically describes the carrier link between your telephony infrastructure and the PSTN.
A SIP trunkโs call capacity is defined by its concurrent channel count, not a fixed physical limit. You purchase a set number of simultaneous call channels โ for example, 10 channels means 10 calls can be active at the same time. Most SIP trunk providers offer elastic capacity that can scale up or down on demand. For sizing, a rough guide is one channel per 6โ8 employees for typical business call patterns, adjusted up for contact centres or high call volumes.
To use SIP trunking you need: a SIP-compatible PBX (hardware-based such as Cisco or Avaya, or software-based such as Asterisk or FreePBX), a stable broadband internet connection with sufficient bandwidth (approximately 100 kbps per concurrent call for standard G.711 codec), and optionally a Session Border Controller (SBC) for enterprise-grade security and interoperability. Many modern IP-PBX systems are SIP-ready out of the box and require only provider credentials to configure.
