If you have used the internet, in your office or at home, then you are familiar with the concept of data being sent and received by your computer. VoIP, or IP telephony, is yet another way of using the internet. VoIP calls can be made using your existing business computer network. Calls to a multi-site business can also be made through the appropriate existing internet connections.
Typically, to make a VoIP call you do need an IP phone. You can make calls from it to any type of phone anywhere in the world: to other IP phones, and via the use of an ITSP (Internet Telephony Service Provider) to traditional phones and mobiles, locally, nationally or internationally. IP handsets work just like ordinary handsets so making a VoIP call the same as making a traditional call – except that the way the voice data is carried from caller’s handset to the receiver’s handset is different.
So how is a VoIP call different?
The caller’s voice is ‘digitized’ and compressed at the handset. It is then chopped into small manageable chunks, or ‘voice data packets’. These ‘packets’ are transmitted, or travel, from the phone through network devices ‘switches’ to the edge of the company (a router and a VoIP gateway) the outside the company (using traditional lines or via internet) and on to the recipients handset, to be ‘re-packaged’.
Vint Cerf, known as the father of the internet, has a good analogy explaining ‘packets’ on a network: he describes them as postcards being sent through the postal system. A message is written over several postcards. On arrival, they have to be sequenced so that the message makes sense. For voice, ‘packets’ to arrive in a timely manner and so must be given priority on the data network to ensure an acceptable call quality.
You can click below for more in depth detailed and technical information. We have also listed definitions and terminology in our Glossary for your reference.
+Typically VoIP calls can be made through IP phones but can also be made computer-to-computer (or ‘IP-to-IP’) with the user of ‘soft phones’ and using traditional phones using an adaptor (ATA).
The voice is 'digitized' and compressed using speech CODEC’s. This compression capability reduces network bandwidth. There are several standard and widely adopted VoIP CODEC’s in the industry, the most common being G.711 and G.729. The CODEC is also used at the receiver end (or at ‘breakout’ at the ITSP) and decompresses the signal for the listener. The higher the compression the reduced requirement for bandwidth – but this is to be contrasted with a lower voice quality call. Conversely, the lower compression CODEC’s require a greater bandwidth, which leads to higher voice quality. G.711 has minimal compression (64 Kbps) so will give the best voice quality at the expense of greater bandwidth use. The G.729 offers a voice bit rate of 8 Kbps. This voice compression of an 8 to 1 ratio, only equates to bandwidth reduction of about 3 or 4 to 1. This is because the packet overhead negates some of the voice compression bandwidth savings. Generally, as a ‘rule of thumb’ it is suggested that 16 Kbps is added to the compressed bandwidth, to calculate the total bandwidth required. In addition some contingency should be allowed also and in short, the reality of the bandwidth taken is detailed in the table below.
CODEC Bandwidth IP Bandwidth G.711 64 Kbps 90 Kbps G.729 8 Kbps 40 Kbps In order for VoIP to function correctly, voice data packets must be given the utmost priority such that their journey is not affected or impeded. The issue is how to guarantee that the packet traffic is not delayed or dropped due to interference from other lower priority traffic on the network. Factors that can affect the quality of the speech include latency, jitter and packet loss. In our view, priority of voice data packets to ensure voice quality must be the minimal level of service offered by any IP telephony provider.
The signalling protocol used for VoIP is SIP, Session Initiation Protocol. It can be likened to HTTP, the web protocol, in that messages contain headers and a message body. Since SIP is a flexible protocol, it is possible to add more features and keep downward interoperability. SIP is the industry de-facto open standard, though some systems do use proprietary protocols.
Only a handful of business-focused IP telephony providers have the appropriate knowledge and quality of channel partners to affect such a solution. Here at 500 we pride ourselves on our commitment to providing our clients with end-to-end quality of service (QoS).
