SIP trunking is the main vector of today’s fixed telephony.
But how does it work? Is this relevant to you? What are the costs and benefits?
In order to explain what a SIP Trunk is, it is first necessary to introduce the concepts of SIP and VoIP.
VoIP: Voice over Internet Protocol designates telephonic communications carried out over an internet connection, in contrast with communications established using traditional telephone lines (called analog telephony).
SIP: the Session Initiation Protocol is a protocol used to initiate, maintain and complete telephone calls on VoIP. In a way, SIP enables a VoIP call to happen.
SIP Trunk: a SIP Trunk is the concept of collecting the calls of a set of phones on a single infrastructure (the one of your telephony provider). The calls will then be carried to their destination. The term “trunk” is a metaphor illustrating the fact that all the telephone lines join up at a single point (the provider’s infrastructure) and follow a long channel, hence the trunk shape.
IPBX: Most companies now have what we call an IPBX. The Internet Private Branch Exchange links the employees' phones between themselves. Telephony-wide, it is the entry point to the company. It helps the IT team to manage their phone fleet and their users scenarios (like forwarding a call, editing a voicemail, changing a number). As it is at the center of the company's network, SIP Trunk providers will generally connect to their clients at the IPBX level. In other words, the IPBX is the base of the Trunk.
Other telephony protocols
Note that there are other fixed telephony formats. PSTN, for example, is the name given to copper-borne, analog telephony (stands for Public Switched Telephone Network). For a long time, this was the dominant telephone network. However, all around the world, its disappearance beckons. In the UK, BT has recently decided to end it (with an horizon of 2025) in favor of VoIP, which is more suited to modern usage. We talk about this at greater length in the white paper “The end of PSTN".
Web RTC (RTC stands for "Real-Time Communication") is another way of carrying voice over the internet, using your web browser. Skype or Zoom heavily use this protocol.
Despite all this, among all the options available today, SIP Trunking is the dominant service for fixed telephony.
Finally, to complete the jigsaw: you need numbers. These numbers will then be linked to your PBX. Via this PBX, you can assign a number to any phone or employee.
More precisely, how could we break down a Trunk service ?
If you are using a provider, the latter will connect your “voice gateway” (i.e. a router) or your PBX to its infrastructure (its core network in telecom terms). You can then benefit from its call rates in your country and internationally.
If one of your users makes a call, the SIP session (session initiation) will pass through your gateway (or Centrex), will then be transmitted to the provider’s SBC (Session Border Controller) which will serve as a filter, for a matter of security. The signal then arrives at the Proxy, which compares all the potential routes and chooses the most adequate one. Indeed, one single call could take an infinity of different “routes” (through one operator or another) to reach its destination. The session initiation will then be redirected to a second SBC, dedicated to the provider which holds the chosen route. This third-party provider will then complete the call to the recipient. If this recipient is available, it will send an acceptance in the opposite direction up to your user’s telephone. The call can start!
The functioning of a telephone call is quite similar to the mail system! The SIP would be a letter sent by person A to a mailbox, your gateway. It arrives at a post office, the SBC, will filter out defective parcels. The letter is then directed to the sorting office, the Proxy, which chooses the destination post office (the second SBC). A postman delivers the letter to its destination, which is our third-party provider. He will then deliver it to person B. B sends an answer using the same path.