Understanding the difference between ISDN and SIP is an important part of your potential migration. ISDN circuits are physically installed into your premises and connected to your PBX. With SIP, the requirement for physical infrastructure is removed and replaced with a virtual version in the form of trunks transmitting data.
Session Initiation Protocol (SIP) is the protocol of VoIP communication that allows users to make and receive calls over the internet. SIP works by sending messages from one SIP address to another. These messages are typically voice calls. However, SIP also powers messages in the form of video calling and instant messaging.
SIP is an over-the-internet exchange of information. A session is setup to initiate a transmission of these voice, video or instant messages. Once a SIP session is established, commonly referred to as a handshake, the data is sent, managed and ended by SIP. This sequence of events happens over milliseconds. This is how SIP calls can provide call quality better than that of a traditional phone line.
Hosting the call
The parameters of SIP only include the call itself. Equipment or software (endpoints) are required at either side of the call (SIP session). These could be standard telephone to telephone interactions, a blend of desk phone to softphone or from dedicated SIP devices like conference phones out to the external PSTN network where non-SIP users are able to receive calls as standard.
If you are familiar with the PSTN or ISDN networks, SIP works in a similar way to connect a PBX with the outside world. Instead of routing calls over your copper lines, SIP routes calls over a data network. This could be your internal LAN, an MPLS network that combines your sites together or even the public internet.
Using servers that manage signalling and voice access to a central resource, usually a PBX, SIP is the middle man between your phone system and any external networks like the internet.
The term SIP trunk derives from lines that conduct electrical currents to switch data signals to one another. SIP does this in a virtualised environment, with no need for the physical connection. SIP acts as a virtual connection between a business and a SIP platform by linking SIP trunks to other IP traffic, or via the internet and out to the PSTN network.
You need to connect a SIP trunk to a SIP compatible device. With no endpoints, there is no data to transmit. The most common use case is a phone system. As SIP trunks only carry the call data, this data needs to be delivered from one endpoint to another. Examples are vast but include:
- Cloud phone system to cloud phone system
- Cloud phone system to non-cloud phone system
- IP PBX to PSTN network (outside world)
- IP PBX to soft phone user
As businesses demand more and more mobility, enterprises are using a mix of desk phones, soft phones and mobile endpoints. These all come with a variety of configurations to enable your business phone number to appear on any such device – removing the traditional need to be fixed to a desk.
Just because SIP is a virtualised technology, it doesn’t mean it is incompatible with other telephone services. A common misconception is that SIP is restricted to SIP to SIP calling. SIP simply replaces the vehicle used to carry call traffic. Your calls can still be delivered as before, often in better quality.
Once SIP is connected to an IP PBX, it is capable of routing calls over data networks, but also differs from traditional phone systems in other ways. An IP PBX should support the extended range of communication services offered by VoIP providers and configuration is simple to deploy.
This is where we often see businesses base their business continuity and communications strategies from the data and processes they control with their SIP trunks. Adoption of SIP opens the door to many business enhancements other than cheap rental and calling,
For SIP calls to be transferred over the internet, it must be encoded. This means speech must be transmitted as data. The audio signals found in speech are translated into codecs and passed from a SIP endpoint to its desired destination. For audio calls, SIP most frequently uses G.711 and G729 codecs. Other codecs are available for lower quality traffic and specialist high quality transmissions.
G7.11 is the standard codec used in SIP. For better quality of speech, this codec transmits the audio signals without compressing the voice data. Quality is not lost when using G7.11 as it utilises the bandwidth available to transfer speech as if in the same room as the caller.
G.729 is a reduced quality codec, where the voice data is compressed. G7.29 is typically seen in scenarios where bandwidth is restricted or unavailable for dedicated voice use. This is often the case in temporary deployments when clients have ADSL and FTTC connections rather than installing a dedicated internet connection.
Getting started with SIP trunking
Part of being an IT Manager when introducing SIP trunking is transforming your business. As businesses change, in line with digital and modern technology, people are going to ask questions as to why you are changing and how it will impact them. As the IT Manager, you need to be empowered so you can best respond to these questions and ensure you drive the business in the right direction.
With SIP trunking, there is a lot to learn and many different options available. The best place to start with SIP, like most things, is at the beginning.
- Have you captured your requirements, involving representation from all stakeholders in your business?
- Do you know the defined budget you are working to, or what savings your business is aiming to achieve?
- Have you conducted an audit of everything that is currently connected to your PBX, PSTN and ISDN sockets?
You’ll naturally want to learn the installation requirements for SIP trunking and know how to sift through the wealth of SIP providers out there. Don’t panic. There’s plenty of time before you even start to plan your SIP migration. We’ll be continuing this blog series and answering these questions for you, and you can download the complete guide below.