Frequently Asked Questions

VOIP Frequently Asked Questions

What is VOIP?

What Internet connections work with VOIP?

But I already have Skype / MSN / Yahoo Chat...

Can I use my existing phone?

Can I use my cellphone?

Can I use my computer?

Where's my free phone?

Sjoe, these IP phones are expensive...

What's an ATA?

What's FXS and FXO?

What's PoE?

What's an Asterisk Server?

My phone's not logging in.

Call quality is bad.

There's an echo on the line when I make or receive a call.

My question still hasn't been answered

Q: What is VOIP?

A: Voice-over-Internet protocol (VOIP) is a protocol optimized for the transmission of voice through the Internet or other packet-switched networks. VOIP is often used abstractly to refer to the actual transmission of voice (rather than the protocol implementing it). This latter concept is also referred to as IP telephony, Internet telephony, voice over broadband, broadband telephony, and broadband phone.

Cost savings are due to using a single network to carry voice and data, especially where users have underused network capacity that can carry VOIP at no additional cost. VOIP-to-VOIP phone calls are sometimes free, while VOIP calls connecting to public switched telephone networks (VOIP-to-PSTN) may have a cost that is borne by the VOIP user.

Voice-over-IP systems carry telephony signals as digital audio, typically reduced in data rate using speech data compression techniques, encapsulated in a data-packet stream over IP.

VOIP can facilitate tasks and provide services that may be more difficult to implement or more expensive using the PSTN. Examples include:

  • The ability to transmit more than one telephone call over the same broadband connection. This can make VOIP a simple way to add an extra telephone line to a home or office.

  • Conference calling, call forwarding, automatic redial, and caller ID; zero- or near-zero-cost features that traditional telecommunication companies (telcos) normally charge extra for.

  • Secure calls using standardized protocols (such as Secure Real-time Transport Protocol.) Most of the difficulties of creating a secure phone connection over traditional phone lines, like digitizing and digital transmission, are already in place with VOIP. It is only necessary to encrypt and authenticate the existing data stream.

  • Location independence. Only an Internet connection is needed to get a connection to a VOIP provider. For instance, call center agents using VOIP phones can work from anywhere with a sufficiently fast and stable Internet connection.

  • Integration with other services available over the Internet, including video conversation, message or data file exchange in parallel with the conversation, audio conferencing, managing address books, and passing information about whether others (e.g. friends or colleagues) are available to interested parties.

  • Advanced Telephony features such as call routing, screen pops, and IVR implementations are easier and cheaper to implement and integrate. The fact that the phone call is on the same data network as a user's PC opens a new door to possibilities.

The nature of IP makes it difficult to locate network users geographically. Emergency calls, therefore, cannot easily be routed to a nearby call center. We strongly recommend that you have some form of backup telephone link (such as a cellular phone) for making emergency calls.

Q: What Internet connections work with VOIP?

A: Most broadband connections (ADSL, 3G, HSDPA, cable or WiFi) should work.

The important thing is that your bandwidth provider / ISP should prioritise VOIP traffic to ensure good quality connections between your handset(s) and our VOIP servers.

Unfortunately, many ISPs cheat by using transparent web proxy servers to increase download speed and simply don't have the capacity to provide decent quality VOIP connections.

Q: But I already have Skype / MSN / Yahoo Chat...

A: SIP is an "open" standard. With our service, your phone, PABX or Asterisk server logs into a SIP server (or several SIP servers) and you can make and receive calls.

The Skype Privacy Policy clearly states that your computer may be used by Skype:

disk space, bandwidth and processing power may be utilized to provide the Skype Services

It further describes how your computer may act as a hub (essentially a server) for use by others:

From time-to-time your computer may become a Supernode… This may include the ability for your computer to help anonymously and securely facilitate communications between other users of the Skype Software…

And Skype of course reassures us that:

The system has been designed so that being a Supernode will not interfere with the normal operations of your computer.

Whew. That’s a relief.

Oddly, the Skype End User License Agreement (EULA) is far less clear on the point:

4.1 Permission to utilize Your computer. In order to receive the benefits provided by the Skype Software, you hereby grant permission for the Skype Software to utilize the processor and bandwidth of Your computer for the limited purpose of facilitating the communication between You and other Skype Software users.

Here they tell us we are granting them permission to use our resources just for the purposes of facilitating the communication between me and other users. Now wait a minute. What’s up with this discrepency? Letting Skype use my computer to facilitatie my own communications is one thing. But it is an entirely different matter to grant permission for Skype to use my private property to facilitatie the communications of strangers, communications to which I am not a party.

Forgetting this discrepency (which itself seems somewhat dubious), the fact is with Skype as it is today, your computer can become a hub (Supernode) and carry the conversations of others, without your explicit knowledge or active consent. The parallels to Palladium are many. In both cases, a big brother in the sky tells us whose computer we can trust, as well as when and how we should trust it. And all the protocols and algorithms are secret, not exposed to peer review or the kind of extensive public scrutiny required to affirm the security of the design.

I quote an associate:

There is a “social cost” to using Skype. You willingly help bad guys get their work done, in addition to all the good that gets done over Skype. Good defined as things you personally consider good or benign — all from your personal perspective.

If you are willing to pay this price of letting bad guys use your machine, then that’s up to you. No one can stop you. But users should be aware of these costs and offer or withhold their consent accordingly.

This is a personal decision. Contrast this with paying taxes that build roads that the bad guys use to flee their bankrobbery. I don’t have a personal choice in paying taxes, nor whether they are spent on building roads that lead to banks.

But I do have a choice with how my computer is used. It’s my personal, private property.

I wonder how many Skype users even know their computers and internet bandwidth can be used to carry traffiic for others. This includes SkypeOut calls that Skype is making money on.

Again, SIP is an "open" standard. Your phone, PABX or Asterisk server logs into a SIP server (or several SIP servers) and you can make and receive calls. The system does not use your bandwidth for any other purpose than your phone calls.

Because Skype uses your bandwidth at will, you may find that your bandwidth is eaten up without you using it, especially if you are using prepaid or limited bandwidth options. Thus you are paying for other people to use your bandwidth.

Q: Can I use my existing phone?

A: Yes. All you need is an ATA to connect your POTS phone to our servers.

Q: Can I use my cellphone?

A: We've achieved success connecting smartphones to our SIP servers using the settings built into certain phones.

We also have a Windows Mobile softphone that is available for download.

Caveat Emptor Getting SIP to run on some Symbian phones is a bit of a hit and miss affair. Please contact us before spending a large sum of money on a phone that may or may not connect to our SIP servers.

Q: Can I use my computer?

A: You sure can. You'd need to download and install a Softphone though.

Q: Where's my free phone?

A: We don't tie our clients down through contracts and we don't offer 'free' phones. A 'free' phone simply isn't free! You are paying much more for it in the long run.

You may also find that the companies doling out free phones and free PABX's only offer non geographical / local access only "087" numbers. Have you ever tried calling an "087" number from outside of South Africa? Call us old fashioned if you want, but we think that our clients should be contactable from anywhere in the world, which is why we offer "011","021" and "031" numbers in addition to the traditional "087" numbers.

Q: Sjoe, these IP phones are expensive...

A: We sell phones that vary in price from a couple of hundred Rand through to several thousand Rand.

The price of our LAN Cordless DUALphone is on par with DECT phones from hypermarkets and superstores.

The price of our entry level AT510P is on par with similar offerings from Telkom shops.

An Asterisk server is a fraction of the price of a comparable dedicated PABX.

When you consider the fact that in many cases you are buying an entry level computer that can also browse the web and send and receive emails, then top of the range IP phones are not nearly as expensive as you may have thought them to be.

Q: What's an ATA?

A: An analog telephony adapter, or analog telephone adapter, (ATA) is a device used to connect one or more standard analog telephones to a digital and/or non-standard telephone system such as a Voice over IP based network.

An ATA usually takes the form of a small box with a power adapter, one Ethernet port, one or more FXS telephone ports and may also have a FXO link. Users can plug one or more standard analog telephone devices into the ATA and the analog device(s) will operate, usually transparently, on a VoIP network.

ATAs are used to replace a user's connection to a traditional telephone company.

The most common ATA is a box with at least one Foreign eXchange Station (which includes a telephone jack), used to connect a conventional telephone, and an Ethernet jack used to connect the adapter to a LAN. Using such an ATA, it is possible to connect a conventional telephone to a remote VoIP server. The ATA communicates with the server using a protocol such as H.323, SIP, MGCP, SCCP or IAX, and encodes and decodes the voice signal using a voice codec such as G.711, G.729, GSM, iLBC or others. Since the ATA communicates directly with the VoIP server, it does not require any software, such as a softphone, to be run on a personal computer.

Q: What's FXS and FXO?

A: Foreign eXchange Subscriber (FXS) and Foreign eXchange Office (FXO) are the names of the two most common interfaces (ports or plugs) found in analog telephony environments

FXS - Plug your normal phone into the FXS device, program the device and your normal phone becomes an IP phone.

FXO - plug your Telkom phone line onto the device, program the device and your Telkom number will ring on your IP PBX or Asterisk server.

Q: What's PoE?

A: Power over Ethernet or PoE technology describes a system to transfer electrical power, along with data, to remote devices over standard twisted-pair cable in an Ethernet network. This technology is useful for powering IP telephones, wireless LAN access points, network cameras, remote network switches, embedded computers, and other appliances where it would be inconvenient, expensive (mains wiring must often be done by qualified and/or licensed electricians for legal or insurance reasons) or infeasible to supply power separately. The technology is somewhat comparable to POTS telephones, which also receive power and data (although analog) through the same cable. It doesn't require modification of existing Ethernet cabling infrastructure.

There are several general terms used to describe this feature. The terms Power over Ethernet (PoE), Power over LAN (PoL), and Inline Power are synonymous terms used to describe the powering of attached devices via Ethernet ports.

Q: What's an Asterisk Server?

A: Asterisk is an open source/free software implementation of a telephone private branch exchange (PBX) originally created in 1999 by Mark Spencer of Digium. Like any PBX, it allows a number of attached telephones to make calls to one another, and to connect to other telephone services including the public switched telephone network (PSTN). Its name comes from the asterisk symbol, *, which in Unix (and Unix-like operating systems such as Linux) and DOS environments is a wildcard character, matching any sequence of characters in a filename.

Q: My phone's not logging in.

A: Is it powered on?

Is it plugged in to your network switch?

Is your network router connected to the Internet?

If all else fails, contact us.

Q: Call quality is bad.

A: This is usually a sympton of congestion. Either between you and the VOIP Interconnect, or between the VOIP Interconnect and the internet.

Q: There's an echo on the line when I make or receive a call.

A: Your equipment needs to be fine tuned. Please contact us.

Q: My question still hasn't been answered.

A: Please contact us by clicking on this link