Educational Resource
Free guides explaining the protocols, standards, and architecture behind business phone systems.
Written for IT admins, business buyers, and anyone setting up a modern phone system. Read in order or jump to the topic you need.
VoIP (Voice over Internet Protocol) converts voice into digital packets and transmits them over IP networks. Unlike traditional PSTN lines, VoIP calling uses your existing internet connection to make and receive calls with higher flexibility, lower cost, and no dedicated phone line hardware required.
Read guideSIP (Session Initiation Protocol) is the signalling protocol that sets up, manages, and tears down real-time communication sessions — voice calls, video calls, and messaging — over IP networks. SIP handles the negotiation between endpoints so media can flow after the session is established.
Read guideA PBX (Private Branch Exchange) is a private telephone network that manages internal call routing between users within an organisation and connects external calls through trunk lines to the PSTN. Modern PBX systems can be on-premise hardware or cloud-hosted software.
Read guideSIP trunking replaces traditional analogue phone lines or PRI circuits with virtual connections over the internet. A SIP trunk carries multiple concurrent voice channels through a single IP connection to your PBX or phone system, eliminating per-line hardware costs.
Read guideRTP (Real-time Transport Protocol) carries the actual voice audio between VoIP endpoints after SIP establishes the session. RTP packets contain digitised voice samples encoded with a codec and include sequence numbers and timestamps that allow the receiver to reconstruct audio in the correct order.
Read guideSTIR/SHAKEN is a framework of protocols that authenticate caller ID information for calls carried over IP networks. STIR (Secure Telephone Identity Revisited) and SHAKEN (Signature-based Handling of Asserted Information Using toKENs) work together to verify that the caller ID displayed on inbound calls has not been spoofed.
Read guideE.164 is the international numbering standard maintained by the ITU that defines the format and structure of telephone numbers worldwide. Every public telephone number in the world follows E.164, and VoIP systems require numbers in this format for proper routing between carriers.
Read guideA VoIP codec (coder-decoder) converts analogue voice into a digital bitstream for transmission over IP networks and decodes incoming streams back into audio. The codec choice directly affects call quality, bandwidth usage, and CPU load on endpoints and PBX servers.
Read guideA DID (Direct Inward Dialling) number is a virtual phone number that routes directly to a specific extension, user, or IVR destination within a PBX or VoIP system without requiring a live operator to transfer the call. DIDs decouple the phone number from the physical phone line.
Read guideHosted PBX and cloud PBX are often used interchangeably, but they describe different deployment architectures. Hosted PBX means the provider manages the PBX infrastructure in their data centre and you connect via SIP. Cloud PBX is a specific type of hosted PBX built on multi-tenant cloud infrastructure with shared resources.
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