How do you connect VoIP?

To be able to place or receive calls using VoIP, you need a hardware setup that will allow you to speak and listen. You might simply need a headset with your PC or a complete set of network equipment including routers and phone adapters. Here is a list of the equipment that is normally required for VoIP.

PS. Don’t be discouraged with all the technicalities, because you won’t be needing them all. What you need depends on what you use and how you use it.

VoIP Gateways / Routers

VoIP Gateways are the most common as they enable traditional/legacy PBX’s to become VoIP enabled devices. A VoIP Gateway is connected to a PBX (either via Analog, ISDN BRI, PRI ports) which takes the call, and converts the voice signals into data packets and sends it to the VoIP provider via your Internet Connection.

The existing PBX needs to have available ports (or licenses) to connect and should be programmed for Least Cost Routing (LCR) forcing all calls to first be sent to the VoIP Gateway, thus ensuring Cost Savings against the other local connectivity.

Lastly, you will need to have a Router with available ports to connect the VoIP Gateway. If you have an existing Broadband Service then you already have a Broadband router that the VoIP Provider may be able to connect to.

ATA (Analog Telephone Adapters)

An ATA is commonly called a phone adapter. It is an important device used to act as an interface between an analogue telephone and an Internet VoIP line. You don’t need an ATA if you are using PC-to-PC VoIP, but you use it use an existing Analogue telephone which you want to use in conjunction with your VoIP service.

Telephone Sets

The VoIP or Internet Protocol phone is another VoIP device that can be used. It looks like your standard telephone but is equipped with the Internet or Network wiring to make it VoIP capable. The IP phone simply connects to your computer’s router, and you are ready to make and receive calls. Several types of phones can be used with VoIP, depending on the circumstances, your needs and your choice.

You can check out a few of the top VoIP phones by clicking here.

PC Handsets or Headsets

Handsets resemble telephones but they connect to your computer through USB or a sound card. If your computer has an Internet connection, PC speakers, a sound card, and a microphone, or USB port you can download available software and make phone calls from your computer (like using Skype). This typically isn’t used for Business VoIP.

How much bandwidth will I need to run VoIP?

For each line, you may need 30kbps upload and download bandwidth dependent on the Compression Codec used by the VoIP Provider (we have used the most common Codec in our example – G.729).

Be aware that in South Africa, some Broadband may be Asymmetric; meaning that the upload and download speed isn’t the same. A good example would be a Telkom 4Mbps ADSL where the download speed is 4Mbps, however, the upload speed remains capped at 512kbps.

If you already had one VoIP line and wanted to add another two, you should accommodate for an extra 60kbps of your upload and download speed, so a total of 90kbps of upload and download capacity is required.

There are Bandwidth Optimisation methods such as IAX2 and ViBE, just enquire with your provider as to which they support.

Will my existing Router suffice for Voice connectivity?

Whenever Voice is going to be shared with PC’s or any other device on the network that will be using Internet Bandwidth, it is always important to set Quality of Service (QoS) which will ensure that Voice is given Priority over Data. You will also need an available Ethernet RJ-45 port to connect the VoIP device.

Is my Internet Service Provider adequate for Voice connectivity?

With VoIP, you are creating a connection highway to a Service Provider via the Internet, so how that communication channel is opened (and the highway it takes) contributes towards optimal call quality.

It is always suggested to go with a Tier 1 Internet Service Provider and to ensure that Voice is given priority over other applications such as Internet and Email.

What impact will VoIP have on my Internet Cap or usage?

Bandwidth costs need to be taken into account when implementing VoIP services as VoIP uses data bandwidth just as surfing the Internet or receiving emails.

Most VoIP providers work with G.729 Codec which consumes 30 kilobytes per second of upload and download data. At 30kbps per call, data usage per 60 minutes of voice conversation amounts to 13.5Mb. One month usage at 60 minutes per day will work out to 297Mb per month, or 1Gb of data will give you 74 hours of Talktime. Make sure that you accommodate for this in your Internet plan.

What about my Data or DSL Line?

It is always recommended to have a line dedicated for Voice Connectivity unless the Provider can guarantee optimal voice quality on a shared line.

If a new line needs to be ordered, and it happens to be ADSL, usually this process is expedited by converting an existing Fax Line to ADSL.

Note there are other great alternatives for connectivity such as Wireless – check out our Wireless Provider section by clicking here.

What about my current Telephone Lines?

With Number Portability, you have the option to port your existing Telkom numbers to a VoIP provider – meaning that you keep your Telkom number but replace their service for VoIP.

Will my current phone system integrate with VoIP?

Absolutely Yes. As long as your PABX has incoming line trunk ports, or supports SIP Trunking (is VoIP enabled).

It doesn’t even matter what ports these areas there are so many VoIP Gateways to ‘VoIP enable’ Legacy PABX’s.

Integration with your existing services

We discuss the Pitfalls of VoIP in the About VoIP section, but be aware that your VoIP service may not be able to offer Faxing or connectivity of Point of Sale, Merchant Station, or Alarm systems.

What tools can I use to test my connection?

As no VoIP testing tool will allow us to test each provider from your premises we can recommend carrying out the following tests from your PC, using the Internet connectivity that you intend on adding VoIP onto.


The traceroute utility checks how many “hops” (transfers through other computers on a network) it takes for your computer to contact another computer. You can use traceroute if you know the Providers IP address. The ideal result should be less than 5 hops.

The Ping Test

Ping is a facility that sends a series of packets over a network or the Internet to a specific server/computer to generate a response from that server/computer. The other server/computer responds with an acknowledgement that it received the packets. Ping was created to verify whether a specific computer on a network or the Internet exists and is connected.

The result will show the IP address of the computer you’re pinging, the round-trip time in milliseconds for each packet, the number of packets sent and received, and the number and percentage of how many packets got lost.

To access either of these facilities, open the command prompt:

Windows: From the Start menu, in the search field, type cmd, and then press Enter.

At the command prompt, enter ping or tracert [example]. Replace [example] with the IP address of the server you are trying to access.

Near Toll Quality

For near toll-quality service, we recommend the following performance characteristics:

Network Delay (Latency)
One-way delay (UDP traffic) is 80 ms or less (assumes all jitter buffers are set to accommodate the 40 ms maximum jitter level specified below).

Network Jitter
Average jitter is 20 ms or less, with a maximum of up to 40 ms.

Network Packet Loss
Uniformly distributed packet loss is 1% or less.

Business Communications Quality

For business communications quality service, we recommend the following performance characteristics:

Network Delay (Latency)
One-way delay (UDP traffic) is 120 ms or less (assumes all jitter buffers are set to accommodate the 80 ms maximum jitter level specified below).

Network Jitter
Average jitter is 40 ms or less, with a maximum of up to 80 ms.

Network Packet Loss
Uniformly distributed packet loss is 3% or less.

What does it mean to have Quality of Service?

Quality of Service (QoS) in the VoIP environment refers to the priority of Voice over an Internet or Data line.

If a subscriber is using an Internet line with no priority for Voice, factors such as browsing the internet, or receiving an email will affect the throughput, and therefore Voice Quality. Remember, Voice is a real-time application, and therefore needs to be given priority over all other traffic.

Quality of Service is not only affected on the Subscribers side by having a Router that supports SIP Quality of Service (QoS), but this QoS needs to be recognised, managed and controlled by the Internet Service Provider.

There is a multitude of last-mile connections on the market and the ideal is to either go with a direct 1:1 connection such a link provided by your Internet Service Provider that is Managed, Controlled and Optimised for VoIP connectivity (i.e. Leased Line, Microwave, etc) or with a High-Speed Broadband service that stays off? the Public Internet, otherwise this would be subjected to a best-effort Quality of Service policy.

How to Assess Voice Quality on a VOIP Network

When using a landline telephone, most people don’t think about quality unless they are on their cellphone and notice a drop-in service.  Most people assume that using Voice Over IP (VOIP) services will have a fairly high level of quality, especially if they’re connecting from a hardwired internet connection that is fairly fast.  However, that’s not always true.  There is a way of measuring the quality of your VOIP network, and there are a few things you can do to improve the overall quality of your calls.

Measuring your VOIP Quality

Mean Opinion Score (MOS)

VOIP quality is measured by its MOS. To arrive at a networks MOS, users rate their call experiences from 1 (unacceptable quality) to 5 (excellent quality).  In most cases, a VOIP call falls in between 3.5 and 4.2.  Most are good enough that there’s no issue with communication.  Everyone on the call can be heard clearly, and there are no misunderstandings.  Generally, anything above a 4 is considered good, while anything below a 3.6 is considered poor.

MOS can be calculated by using the International Telecommunications Union (ITU) standards P.861 and P.862 or using ITU Standard G.107.  These standards outline how to make a test call and then evaluate the quality of that call.  P.563 can also be used to get a network’s estimated MOS, but it’s a passive way of determining the score rather than someone actively listening and scoring the system.  Some newer systems have a tool built-in for measuring call quality.

The R-Factor

Another way of determining the quality of your calls on a VOIP enabled business phone is to look at its R-Factor.  This score is an alternative to the MOS.  It runs on a scale that ranges from 0 to 120.  Because it has a larger scale, it can be more precise than MOS. It uses a combination of objective factors and user input to determine a networks final score.  When looking at a network?s R-Factor, you will want to be sure the score is at least an 80.  Anything below that indicates that at least some of the users were dissatisfied with the overall quality of the calls they were making or receiving.  At a score of less than 60, almost all users of the network report dissatisfaction.  A score of 93 or better is considered excellent on the R-Factor scale.

Factors Included in Measuring Quality

Several different factors can be studied to determine the quality of your VOIP system.  Note that some VoIP quality tests weigh these factors differently or use different factors to arrive at a systems MOS.

This is how long it takes for a person’s voice to be heard on the other end of the call. It’s also called latency, a term many computer users are familiar with. The delay may occur during the time the message is being sent across the network or during the time it takes for the message to leave or enter the routers, modems, and other equipment at either end.

Jitter is another word for static or garbled communication. When a VoIP network has a high amount of jitter, it’s possible that communication will randomly cut out, leading to missed words.  Static or other odd distortions may also work their way into the call, making it impossible to understand what’s being said.

Lost Data
This occurs when the network loses data packets.  The result may be a moment of lost audio, resulting in missing a syllable or an entire word from the conversation. If a call has severe packet loss, it may be impossible to understand one another.

A Quality Network

Using either the MOS scale or the R-Factor, a business can determine if its VoIP network is truly meeting its needs or not.  If it’s not, several things can be done. The internet connection may need to be upgraded.  It’s also possible that the equipment the company has been using to connect to the internet is obsolete or simply not suited for VoIP. There may also be an issue affecting the quality that is outside of your control, such as a natural disaster or storm that has damaged your internet connection.

However, note that before making any changes, it’s important to evaluate the network using multiple calls to multiple locations. Otherwise, it’s possible the quality issues are actually on the other end of the call.  Make sure to fully test your system before doing anything because it’s possible the problem is not yours.

Source: ittoolbox