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?
What is an SIP trunk?
In order to explain what SIP is, it is first necessary to present to you its vector: 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).
As regards SIP (Session Initiation Protocol), this is a protocol used to initiate, maintain and complete telephone calls on VoIP. In various ways, SIP allows communication on VoIP. One of their combined benefits is to allow multi-media exchanges including voice, video and messages.
By using SIP, a telecom provider can connect your business to the telephone network. They will also have to provide you with a PBX (the Private Branch Exchange is a tool managing your call flows).
This grouping is what one may call a SIP trunking service. “Trunking” derives from the term “trunk”, which is a metaphor illustrating the fact that all the telephone lines join up at a single point, the PBX, and on a long channel (hence the trunk shape): the provider’s infrastructure.
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 is another type of telephony using your web browser (RTC stands for "Real-Time Communication"). It is mainly by using this tool that solutions such as Skype have proliferated.
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 Centrex. Via this Centrex, you can assign a number to any phone. Thanks to this, your employees will be able, for example, to change their telephone according to their movements, while still retaining the same number.
How does a Trunk SIP work?
More precisely, what does a Trunk service break down into?
If you are using a provider, which is the most likely case, 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 reasons of security. The signal then arrives at the Proxy, which compares all the potential routes and chooses the most beneficial. 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 now start!
The functioning of a telephone call is in the end 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, which filters 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.