HomeTechnologyNetwork pre-requisites for VoIP deployment

Network pre-requisites for VoIP deployment

IN our article last week, we learned about the key equipment required for VoIP (Voice over Internet Protocol) deployment which are an IP-PBX , IP Phones, Analogue Terminal Adapters/VoIP Gateway, GSM Gateway, PSTN Gateway, soft phones and, of course, a network which can be Ethernet or fibre optics or wireless. In today’s article, we continue demystifying VoIP, focussing on the network itself.

Deployment pre-requisites

VoIP deployment is heavily dependant on the quality of the internet connection one has. Conduct a bandwidth test on your network. This is a litmus test to establish if you have adequate room to carry your voice traffic. Your bandwidth requirements come in two parts. The first part is the connectivity between your branches. You want to first link all these branches via a private network.

This is basically done via VPNs (virtual private networks) and Zimbabwean companies have mastered this for years. Calls made between branches should be of high quality since you have direct control of the connection within your private network as you are not getting outside your private network for intra-office and inter-office calls.

Interconnections become necessary when you want to send calls outside your private network, say to a landline or a cellular phone user, through licensed providers by The Postal and Telecommunications Regulatory Authority (Potraz).

The second part of the bandwidth equation relates to connecting to the internet so that you can make calls beyond our borders. This “boundarylessness” of the internet makes the making of international calls cheaper.

Vendor selection

Before you make a decision to switch over to VoIP, you will need to conduct some research and make consultations with your VPN network provider. The VPN network provider will come on site to understand your present situation. Things like number of branch offices, employees, internet users, your internet connectivity, your current bandwidth, power supplies, etc, come into play.

Quality of service (QOS)

VoIP deployment challenges include latency, packet loss and jitter.

Latency: As mentioned earlier, your bandwidth and the type of equipment you will use will have a huge bearing on the quality of VoIP network you are setting up.

Remember, you are trying to use a data network to carry voice traffic. And voice is very delay sensitive. The amount of time it takes for your traffic to travel from, say, one branch office in Harare to another in Bulawayo has a huge bearing on the quality of the call.

Technically, this is known as latency. Ideally, you want a latency of around 150ms to have toll-grade voice quality, among other factors.

So besides good connections, you need a network that is QOS intelligent. This basically means your equipment must have the ability to prioritise process voice packets over data since you might be using the same physical infrastructure for your Internet and data applications which are bandwidth intensive.

Packet loss: Your overall network must not lose more than 2% of voice packets sent from point A to point B. Serious packet loss will result in incoherent conversations. I am sure you have heard people talking on the cellphone and saying “you are breaking”. This can be caused by loss of packets on the network.

You want to design your network in such a way that you have a minimal number of nodes and interconnections. Too many nodes and interconnections add to your latency and contribute to packet loss.

On poor or overloaded IP connections, the amount of data traffic — i.e. the number of packets — may exceed the capacity of the connection. When this occurs, packets are discarded (lost) by the router or host computer at the point of congestion.

Jitter: This is the measure of the variation from packet to packet in round-trip time. This measure is calculated as the standard deviation of the individual packet’s round-trip time. If the variation is constant, then it is fairly easy to have a decent conversation; if jitter is very high, the receiving end phone might not be patient enough to wait for a delayed voice packet and technically that packet is lost and the speech quality is poor.

Jitter can be dealt with by use of buffers (used to compensate for varying delay), but do not forget that the buffers themselves will add to the end-to-end delay. Jitter must therefore be minimised


According to research, for an organisation that has more than five branches, 50% of the telephone bill is made up of calls made within an organisation, thus branches calling each other. Take advantage of VoIP and make free telephone calls within the branch network and save big.

This article is published by Standard Global Communications (SGC). The company has been deploying telecoms and electronic security solutions in Zimbabwe for over 20 years. SGC offers free consultation, designs, deploys and provides after-sales support. Our flexible and scalable solutions help reduce telephone costs, while increasing operational efficiency at an affordable cost. For inquiries or more information, questions and comments, please call: +263 242 790 791-5, mobile: +263 772 875 577, e-mail info@standardglobal.co.zw or follow us on Facebook, LinkedIn, Twitter or visit www.standardglobal.co.zw for more or if you wish to download this article.

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