SIP as a halfway house between traditional and fully hosted systems

by | Jan 24, 2020 | Hosting, Telephone System, VoIP (Voice over Internet Protocol)

Transferring systems from a traditional PBX across to a hosted solution may seem like the obvious choice. However, for many organisations this simply isn’t practical. This is where SIP comes in.


It provides a half-way house.

Where there is existing infrastructure in place i.e. a physical PBX telephone system, moving to a hosted system doesn’t always fit with the existing business model.  In this case, SIP is a good option.

It can fit seamlessly into the existing call-handling software and PBX systems.

For instance, it gives users the option of cost-savings on calls, as well as the security benefits that are not necessarily possible with a hosted system.

Using SIP trunking also negates the need to replace existing infrastructure. In addition, it allows for multiple solutions to work together. In essence, SIP can offer the same functionality as hosted, but without the need to change what you already have.

What is SIP?

It stands for session initiation protocol. This is an IP telephony signalling protocol for VoIP (voice over internet protocol) calls and is the most widely used protocol in IP telephony.  We have a free video guide SIP trunks explained here.

Voice calls involve two stages, namely a call setup stage and a data transfer stage. As such, protocols allow a phone to connect to another phone or device during the call setup stage. To transfer voice packages between these devices requires further protocols (stage two).

SIP is a call setup protocol. It facilitates connection, monitoring and disconnection of VoIP sessions. In doing so, this allows internet service providers to integrate basic IP phone capabilities with web, email and on-line chat. It means that users can make telephone calls, video and audio calls as well as operate instant messaging between devices over the internet and for free.

For voice calls, it utilises the VoIP network to set up the call. It then uses another protocol – the real time transport protocol (RTP) to send the voice data between calls. Put simply, it is a text-based protocol similar to HTTP. A SIP end station has a URI rather than a traditional style telephone number. This is similar to a URL and may look something like this:


However, for ease of use, most of these addresses are numbers translated into a SIP address by the device when used.

It supports all the features of traditional telephony such as call transfer, on hold and call forwarding. This means calls between one or more users at the same time are possible. You can disconnect a user at any time during the call. In addition, users can join the call during a session and operators can enable or disable video connection at any time.

What do you need?

1.) A provider – an Internet Service Provider will provide SIP in the form of a program installed on a computer or mobile device. Unlike traditional telephony services requiring physical wires, you only need an internet connection with sufficient bandwidth to support voice calls and video communication.
2.) A SIP address – most likely to appear numeric for ease of use, this will translate to a SIP address when used by your device.
3.) SIP phones – similar to traditional phones in appearance, SIP phones have in-built hardware to enable them to use the internet to make and receive calls. 
4.) Software – to enable a device such as a computer, tablet or mobile to operate as a SIP phone requires equipment such as headsets, microphones, sound cards, earpieces and web cams.
5.) SIP trunking – used as an addition or alternative to existing ISDN BRI or PRI lines, SIP trunking allows users to make full use of an IP phone system.

SIP trunk providers work in a similar way to traditional telephony providers. They lease users a phone number and lines. In other words, there are no physical lines to install and maintain. Set up and call costs tend to be lower.

Connection is via a line used only for SIP trunking, over a dedicated line that carries SIP trunking with other IP traffic, or over the Internet on a virtual private network (VPN). This type of trunking allows scalability to cater for increased call volumes. It means you only pay for the lines you need, and it is possible to share the capacity across multiple sites.

This type of trunking also enables extension of voice over IP (VoIP) telephony beyond an organisation’s firewall without the need for an IP-PSTN gateway. It is becoming particularly popular amongst businesses adopting flexible working options.

Among large-scale enterprises and SMB’s, the worldwide SIP trunk market is strong. However, in the UK, this is not the case.

Speculation is that the PSTN and ISDN phase-out is too far in advance to have an impact on the current market. In Europe, some countries have already made the switch and the move towards this has been much more rapid. Therefore, this has had a greater impact on the uptake of SIP trunking.

However, in the UK, the many providers focus on VoIP. For example, there are over 90 hosted telephony providers, as opposed to only a handful of SIP ones.

Does SIP have a future?
With BT phasing out the UK Public Switched Telephone Network (PSTN) and Integrated Services Digital Network (ISDN) by 2025, SIP trunking will replace traditional phone lines running from a phone system to the national, international and mobile telephone networks.  Here is a free video guide about the ISDN Phase Out.

If you would like to learn more about what MFTS can offer your business as an alternative to ISDN, call us on 01892 577 577. Alternatively, you can send us an email.

MF Telecom Services is a leading UK business telecoms solutions provider specialising in Telephone Systems, Telephone System Maintenance, Voice Communications, Business Mobiles, Business Broadband and IT Support.