A Brief History
Voice over IP is a technology that came to the fore in the nineties due to the growth of the usage of IP in local area networks and the Internet. Prior to the explosive growth of the internet there was not a real need for most people PC’s in business to run a network protocol as practically all their communications was done via the local area network. The Internet changed all that and IP, which was a longstanding internet protocol found rapid adoption.
However, it soon became clear that IP could transport a lot more than just data, you could also packetize voice and then sent it encapsulated within the IP packets. This was the humble origins of VoIP. The simplest systems used analogue to digital convertors to convert the analogue voice signal captured from a microphone and convert it into a digital representation of binary bits, basically a sequence of 1 and 0’s that represented the analogue signal. This digital representation could then be transported across a network and at the other end be reconstructed via a digital to analogue converted to retrieve the original analogue voice signal. On LANs, with at the time 10 Mbps bandwidth, this worked pretty much as well as a plain old telephone set.
VoIP Hits a Snag
The problem was though when you tried to communicate the VoIP over the WAN or worse the Internet, because then you had bandwidth constraints, network latency, delay, packet loss and jitter to content with because typical internet links were still at 56Kbps and voice would be contenting with data for bandwidth. The initial solution to this problem was two-fold; firstly use codec algorithms to condense the voice packets from being 64Kbps, which is their natural packet size on the real telephone network to a more bandwidth efficient size without deteriorating the sound quality too much. Secondly, try to boost the voice packets priority by labeling their packets as priority.
The reason that VoIP packets need priority handling is that voice packets are severely affected by latency and packet loss, even delays of over 120ms in one direction can cause a breakdown in voice quality. The standard telephone network, does not suffer from this because the PSTN telephone networks transport voice packets within a synchronous circuit called a T1/E1 trunk, which makes them extremely predictable in their timing. On the internet, however there is no such thing as predictable delivery as you don’t know the path each of the packets will take from the source to the destination. Indeed packets might take different routes to the same destination, some may get lost, others delayed, and this affects overall voice quality.
Broadband to The Rescue
Setting codec and assigning priority goes a long way to alleviating some of the problems with bandwidth. However, it was the advent of broadband internet and the greatly increased capacity that it brought that enabled VoIP over the internet to become a feasible technology. Soon technologies emerged that took advantage of these high bandwidth low latency links and applications such as Skype became very popular with consumer customers as it provided a method for providing free long distance phone calls.
The Advent of Consumer VoIP Services
Skype started as a PC-to-PC technology using a peer-to-peer network to find paths from sources to recipients using IP. However soon other protocols came about such as SIP which could connect to any SIP enabled device such as an IP telephone and service providers soon arrived selling connection services from IP phones and PC’s to real telephones and mobiles. Companies such as Vantage started selling global call rates to mobiles and telephones and VoIP really took off.
However not everyone was happy many government owned telecom companies tried to block SIP at their firewalls to shutout what they saw was a free service, which could destroy their profitable long distance trunk calls.
However, VoIP became so ubiquitous that even the major telecoms had to relent as they saw there was no practical way to shut down VoIP except on a very limited regional basis. Now it is common to find PSTN/VoIP gateways, which will enable bridging of the two networks and this is a common feature in the IP PBX’s found in most companies.
Business Takes Note of VoIP
VoIP was not just a residential phenomenon as small, medium and even enterprise companies soon adopted it as it represented tremendous potential saving over their old PSTN PBX systems. However, one new VoIP came into being with the rise of VoIP and the advent of cloud computing and that was the virtual telephone system.
A virtual telephone system is a cloud-hosted telephone PBX that can host multi-tenants and provide each customer a service over the internet. This software as a service approach to VoIP became very successful as the service provider for a monthly subscription took on the capital and operational expense of running a complex telephone system for the tenants.
In the next generation of VoIP, far more complex systems virtual telephone systems were developed, which acted as a virtual PBX. These virtual systems had all the complex features and functions of a physical PBX but without the huge price tag and onerous maintenance costs. Cloud hosted IP/PBX systems became very popular with small, medium and enterprise companies as business came to trust the cloud.
Nowadays it is commonplace to see even small businesses supporting call centers, with HD video conferencing, remote presentation and desktop sharing over VoIP. All functions that only five years ago were out with the financial means of all but the largest of companies. VoIP has changed that landscape and now everyone can afford video conferencing and PBX functions and features by using virtual telephone systems that are built in the cloud spreading the cost of ownership across many thousands of customers.