VoIP FAQs and Glossary

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Internet VoIP

Traditionally businesses have used a land-line telephone service to handle their telecommunications. Internet VoIP service is a superior technology that involves the sending of packet switched protocols over computer networks. When communicating on Internet VoIP, the voice signal is converted to a digital signal and sent over a network.


VoIP technology is universal; however adoption of VoIP technology by both consumers and service providers has grown substantially in the past years. According to an SBA, Small Business Research Summary, “Approximately 15 percent of the small business respondents use voice-over-Internet-protocol (VoIP), compared with 3.3 percent in the 2003 study.”

Before switching to a Voice over Internet Protocol service, it is a good idea to make certain that your current Internet bandwidth will support the technology. Each VoIP call requires dedicated bandwidth. As a reference, for perfect voice quality, it will likely require 75kbps, where “tollbooth” quality will require around 8kbps-dedicated bandwidth. As a general rule of thumb, it is recommended that your ISP is offering around 5Mbps total bandwidth for a 10-line phone system.

Tying Everything Together

One of the biggest advantages to being connected online, is the option to have all of your businesses communication channels tied in to one system. Unified communications offer the ability to tie email, fax, voicemail, online chats, and phone systems in to one interface. Unified communications enable a sales force to pull up real time customer data and for management to reduce response time. Managers are also able to track outgoing communication traffic and tie ROI to all of its outbound communication efforts. Many managed VoIP systems offer a unified communications solution for businesses.

What is VoIP?

Voice over IP (VoIP) is a type of phone system that uses an internet connection, rather than a traditional phone line, to transmit voice traffic. One of the most popular reasons for choosing VoIP is cost savings: VoIP service plans are typically less expensive than traditional phone system costs, systems require minimal maintenance, and costs are assessed as a monthly rate only- no long distance charges are applied when you use a VoIP system.

VoIP service provides standard features like voicemail, caller ID, and call waiting, as well as advanced features like video conferencing, call tracking, and online system management.

To use a VoIP system, you’ll need three things: A high-speed internet connection, a VoIP service plan, and phone equipment. Most residences and businesses already have a broadband internet connection fast enough to accommodate a VoIP system- typically a bandwidth of 90 kbps is sufficient. Finding a VoIP service plan is easy if you know what to look for- VoIP vendors design service plans to suit small companies, large corporations with multiple offices, households, and every type of user in between. Most companies will require that you use VoIP-compatible phone equipment- you can purchase VoIP phones or VoIP adaptors, or use the microphone and speakers connected to your computer to make calls. Many providers also allow you to access your VoIP service account at any location on certain mobile devices and cell phones.

Benefits of Using VoIP

Cost savings is the number-one reason most people switch from a traditional phone service to VoIP. The most basic VoIP service plans can be as little as $20/month for a standard phone plan that includes call waiting, unlimited minutes, and one phone line with several extensions. Larger companies or businesses can save big, too-
using a VoIP service, you can virtually eliminate the equipment and maintenance charges that go along with owning a traditional on-site phone system.

VoIP service provides significant cost savings because calls are routed using an internet connection, not traditional phone lines. Since outgoing calls cannot be geographically located, there is no way to assess long distance charges.
A call placed using computer/telephony integration will cost the same whether the destination is across the street or across the country – most users can save thousands or more in long distance charges alone.

Cost savings aren’t the only reason to make the switch. VoIP provides unique features, allows you to access calls from anywhere an internet connection is available, and provides valuable call tracking and system management features that can be easily accessed from a computer. Read MORE about the benefits of VoIP

Switching to VoIP

How do you know if VoIP service is right for your business? What types of companies benefit the most from using internet-based phone technology? Whether you’re looking for preliminary information or actively seeking to make the switch to a VoIP service plan for your home or business, here are a few signs that VoIP technology can help you save big.

Frequently Asked Questions

Do you need to keep a landline to use VoIP? Can you send faxes over a VoIP line? Before you switch, make sure you get all your questions answered – here are the answers to the most frequently asked questions about VoIP service plans and VoIP phone systems.


What is VoIP?

VoIP, or Voice over Internet Protocol, is a type of phone system that uses computer/telephony integration technology to make calls. VoIP routes calls using an internet connection instead of the Public Switched Telephone Network (PTSN). VoIP is often referred to as an internet-based phone system.

Is VoIP less expensive than using a “landline” phone?

VoIP phone bills, or monthly service costs, are typically less expensive than traditional phone system expenses for a few reasons. Since VoIP calls are made using an internet connection, there is no geographic origin for the calls you make – there is no way to assess long distance charges. Most VoIP providers charge a flat rate for service that is considerably lower than traditional phone system costs. VoIP also requires minimal equipment, and can be installed easily without the help of a technician in most cases.

Can I keep my number?

Most VoIP companies allow you to “port” your number over from a landline provider. You can also ask for a number in a different area code, or for a 1-800 number.

Do I need any special equipment?

Most VoIP systems do not require any special equipment – you can even use your current phone with the system. If you’re using an enterprise-grade or business phone system, you may need to purchase VoIP phones or adaptors, which convert a traditional phone signal into a VoIP signal. You can also use a headset and microphone connected to your computer to make calls.

How do I set up a VoIP system?

The way your system is set up will depend on the provider you choose. Most VoIP companies allow you to download system software directly from their website – a process that takes less than an hour- so that you
can begin using the system right away. If your system has multiple lines, extensions, or designated bandwidth, you may need assistance in configuring it.

What does it cost?

VoIP service can cost as little as $20 a month or less and as much as several hundred dollars monthly for business or enterprise grade systems with loads of features. Keep in mind that the monthly cost is just for the phone service – you’ll still need to pay for your own connection costs, usually through an internet service provider. Some providers also charge fees for setup or initial system configuration.

How does a VoIP calling plan work?

VoIP calling plans work similarly to cell phone plans – you can choose an “unlimited” option or purchase a plan with monthly minute limitations. Plans are typically billed as a monthly service charge.

Do I still need a landline?

Some VoIp users like to keep a landline in case of emergencies. Since VoIP uses an internet connection to make calls, service can be interrupted in the event of power outages or surges. Many VoIP providers offer “backup” solutions to ensure you’ll always have phone service when you need it.

If you connect to VoIP using DSL, you will be required to maintain a landline in order to keep your connection active. You can, however, change your phone service plan to a less expensive option, since the phone itself will only be used to connect to the internet.

Can I make calls to non-VoIP phones?

Absolutely. VoIP service allows you to make and receive calls exactly in the same way you would using a landline.

Can I make international calls?

Yes – a VoIP phone can place a call to any phone with a dial tone, no matter where it is located. VoIP calls are also less expensive than international calls using traditional phones.

Can I send faxes using the VoIP connection?

Some VoIP service providers, such as Vonage, ViaTalk, Packet8 and Lingo offer faxing capabilities. Many VoIP providers offer a dedicated fax line as part of a business service package. Taking advantage of such a plan is a good idea- you’ll be able to send and receive faxes without adversely impacting the traffic that travels over your phone connection.

Can I use a wireless internet connection with a VoIP system?

The connection method isn’t as important as the speed of the connection. Typically, a standard wired broadband connection works more effectively than a wireless connection, but if your wireless access is reliable and fast, you can use it to connect your VoIP system anywhere.

Is VoIP call quality as good as that of a regular phone?

Yes. When VoIP was first becoming popular, many users were concerned about “packet loss,” which usually occurs when calls are routed over the “open” internet – other traffic can interfere with voice data. Using a dedicated connection or designating bandwidtch for voice data alleviates these concerns. As internet connections have become faster, voice data is transmitted more effectively – call quality is similar to that of a traditional or “landline” phone.

I don’t understand the terminology. Can you help?

Yes. We’ve created a VoIP service buyer guide just to walk you through the process of switching. In this guide, we have a list of key terms that are frequently used in the industry.

Why should I use VoIP?

Users often cite cost savings as one of the main benefits to using VoIP, but there are also many others. VoIP systems are flexible – you can answer calls from the same number just about anywhere. They’re also highly customizable, easy to manage, and great for small businesses that are hoping to expand.



Average Call Duration.


An application programming interface (API) is a source code interface that a computer system or program library provides to support requests for services to be made of it by a computer program. We currently offer an XML API to query our current rates.


Answer Seizure ratio (ASR) is the number of successfully answered calls divided by the total number of calls attempted (seizures). Since busy signals, calls not answered and other rejections by the called number count as call failures, the calculated ASR value can vary depending on user behavior.


An open source communications platform, Asterisk® is a complete IP PBX in software. It runs on a wide variety of operating systems including Linux, Mac OS X, OpenBSD, FreeBSD and Sun Solaris and provides all of the features you would expect from a PBX.


Formerly Asterisk@Home), the asterisk-based solution, trixbox, enables the home or small business user to quickly set up a VOIP Asterisk-based PBX. A web GUI makes configuration and operation easy. http://www.trixbox.com


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, and one or more FXS telephone ports. 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.


Caller ID (caller identification or CID, and more properly calling number identification – CNID) is a telephony service that transmits the caller’s telephone number to the called party’s telephone equipment during the ringing signal or when the call is being set up but before the call is answered. Where available, Caller ID can also provide a name associated with the calling telephone number.

The information made available to the called party is visible on a small liquid crystal display imbedded on the telephone, or on a separate unit which is connected to the telephone.

Call origination

Call Origination, also known as voice origination, refers to the collecting of the calls initiated by a calling party on a telephone exchange of PSTN, and handing off the calls to a VoIP endpoint or to another exchange or telephone company for completion to a called party.


Telephone exchanges generate so called Call Detail Records (CDRs) which contain detailed information about calls originating from, terminating at or passing through the exchange. Not surprisingly CDRs are used for billing.


Cisco Systems, Inc. (NASDAQ: CSCO, SEHK: 4333) is a global company headquartered in San Jose, California, USA, that designs and sells networking and communications technology and services under four brands: Cisco, Linksys, WebEx and Scientific Atlanta. Initially, Cisco manufactured only enterprise multi-protocol routers, but today Cisco’s products can be found everywhere from the living room to the enterprise to service provider networks. Cisco’s vision is “Changing the Way We Live, Work, Play and Learn.” Cisco’s current tagline is “Welcome to the human network.”[1].


A CNAM database contains calling party names to be used when identifying a calling party. We offer optional CNAM service on our US and Canadian DIDs as well as on our toll free dids.


Voice transmission is analogical, whereas the data network is digital. The process to sample analogical waves into digital information is made by an encoder-decoder (CODEC). There are many standards to sample an analogical voice signal into a digital one. The process is often quite complex. Most of the conversions use pulse code modulation (PCM) or variations.

We support the following codecs:

Bit Rate
Nominal Ethernet Bandwidth (Kilobits)
64 Kbps
87.2 Kbps
8 Kbps
31.2 Kbps
13 kbps
29.2 kb/s


The comma-separated values (or CSV; also known as a comma-separated list or comma-separated variables) file format is a file type that stores tabular data. The format dates back to the early days of business computing. For this reason, CSV files are common on all computer platforms.

CSV is one implementation of a delimited text file, which uses a comma to separate values. However CSV differs from other delimiter separated file formats in using a ” (double quote) character around fields that contain reserved characters (such as commas or newlines). Most other delimiter formats either use an escape character such as a backslash, or have no support for reserved characters.


Direct Inward Dialing. “DID” numbers have particular relevance for VoIP communications. In order for people connected to the traditional PSTN network to call people connected to VoIP networks, DID numbers from the PSTN network are obtained by the administrators of the VoIP network, and assigned to a gateway in the VoIP network. The gateway will then route calls incoming from the PSTN across the IP network to the appropriate VoIP user. Similarly, calls originating in the VoIP network will appear to users on the PSTN as originating from one of the assigned DID numbers, if the user setup his callerid accordingly.


E.164 is an ITU-T recommendation which defines the international public telecommunication numbering plan used in the PSTN and some other data networks. It also defines the format of telephone numbers. E.164 numbers can have a maximum of 15 digits and are usually written with a + prefix. To actually dial such numbers from a normal fixed line phone the appropriate international call prefix must be used.


FreePBX is the most powerful GUI (Web Based) configuration tool for Asterisk. It provides everything that a standard legacy phone system can, plus a huge amount of new features. All documentation and information is avalable from http://www.freepbx.org. FreePBX is included in the trixbox distribution.


G.711 is an ITU-T standard for audio companding. It is primarily used in telephony. The standard was released for usage in 1972.

G.711 represents logarithmic pulse-code modulation (PCM) samples for signals of voice frequencies, sampled at the rate of 8000 samples/second.

There are two main algorithms defined in the standard, the µ-law algorithm (used in North America & Japan) and A-law algorithm (used in Europe and the rest of the world). Both are logarithmic, but A-law was specifically designed to be simpler for a computer to process. The standard also defines a sequence of repeating code values which defines the power level of 0 dB.

The µ-law and A-law algorithms encode 14-bit and 13-bit signed linear PCM samples (respectively) to logarithmic 8-bit samples. Thus, the G.711 encoder will create a 64 kbit/s bitstream for a signal sampled at 8 kHz.


G.723 is a ITU-T standard wideband speech codec. This is an extension of Recommendation G.721 adaptive differential pulse code modulation to 24 and 40 kbit/s for digital circuit multiplication equipment application.

Superseded by G.726, this standard is obsolete.

Note that this is a completely different codec from G.723.1.


G.723.1 is an audio codec for voice that compresses voice audio in 30 ms frames. An algorithmic look-ahead of 7.5 ms duration means that total algorithmic delay is 37.5 ms.

Note that this is a completely different codec from G.723.

There are two bit rates at which G.723.1 can operate:

* 6.3 kbit/s (using 24 byte frames) using a MPC-MLQ algorithm (MOS 3.9)

* 5.3 kbit/s (using 20 byte frames) using an ACELP algorithm (MOS 3.62)

G.723.1 is mostly used in Voice over IP (VoIP) applications due to its low bandwidth requirement. Music or tones such as DTMF or fax tones cannot be transported reliably with this codec, and thus some other method such as G.711 or out-of-band methods should be used to transport these signals. The complexity of the algorithm is below 16 MIPS. 2.2 kilobytes of RAM is needed for codebooks.


G.726 is an ITU-T ADPCM speech codec standard covering the transmission of voice at rates of 16, 24, 32, and 40 kbit/s. It was introduced to supersede both G.721, which covered ADPCM at 32 kbit/s, and G.723, which described ADPCM for 24 and 40 kbit/s. G.726 also introduced a new 16 kbit/s rate.

The four bit rates associated with G.726 are often referred to by the bit size of a sample, which are 2-bits, 3-bits, 4-bits, and 5-bits respectively.

The most commonly used mode is 32 kbit/s, since this is half the rate of G.711, thus increasing the usable network capacity by 100%. It is primarily used on international trunks in the phone network. It also is the standard codec used in DECT wireless phone systems and is used on some Canon cameras.


G.729 is an audio data compression algorithm for voice that compresses voice audio in chunks of 10 milliseconds. Music or tones such as DTMF or fax tones cannot be transported reliably with this codec, and thus use G.711 or out-of-band methods to transport these signals.

G.729 is mostly used in Voice over IP (VoIP) applications for its low bandwidth requirement. Standard G.729 operates at 8 kbit/s, but there are extensions, which provide also 6.4 kbit/s and 11.8 kbit/s rates for marginally worse and better speech quality respectively. Also very common is G.729a which is compatible with G.729, but requires less computation. This lower complexity is not free since speech quality is marginally worsened. G.729 is patented by Sipro in a number of countries. The use of G.729 may require a license fee and/or royalty fee.


H.323 is an umbrella recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. It is currently implemented by various Internet real-time applications such as NetMeeting and Ekiga (the latter using the Open H323 implementation). It is a part of the H.32x series of protocols which also address communications over Integrated Services Digital Network (ISDN), Public switched telephone network (PSTN) or Signaling System 7 (SS7). H.323 is commonly used in Voice over IP (VoIP, Internet Telephony, or IP Telephony) and Internet Protocol (IP)-based videoconferencing. Its purpose is thus similar to that of the Session Initiation Protocol (SIP).


We refer to inbound as traffic we receive and that is directed to you, such as calls to your DID numbers.

Billing Increments

It’s the way we calculate our rates in order to bill your calls. For example, if you call USA for 10 seconds, you will be charged for 12 seconds (2 x 6 seconds since this is a 6 seconds increment call) of a minute, not the whole minute.


Jitter is a typical problem of the connectionless networks or packet switched networks. Due to the information is divided into packets each packet can travel by a different path from the emitter to the receiver.

Jitter is technically the measure of the variability over time of the latency across a network. Real time communications (for example VoIP) usually have quality problems due to this effect. In general, it is a problem in slow-speed links or with congestion. It is hoped that the increase of QoS (quality of the service) mechanisms like priority buffers, bandwidth reservation or high-speed connections (100Mb Ethernet, E3/T3, SDH) can reduce jitter problem in the future although it will keep on being a problem for a long time.


Latency has the reputation of being the enemy of VoIP. It is also called lag. Latency is the time between the moment a voice packet is transmitted and the moment it reaches its destination. It of course leads to delay and finally to echo. It is caused by slow network links. This is what leads to echo.

There are two ways latency is measured: one direction and round trip. One direction latency is the time taken for the packet to travel one way from the source to the destination. Round-trip latency is the time taken for the packet to travel to and from the destination, back to the source. In fact, it is not the same packet that travels back, but an acknowledgement.

Latency is measured in milliseconds (ms) – thousandths of seconds.


IAX is the Inter-Asterisk eXchange protocol used by Asterisk, a dual licensed open source and commercial PBX server from Digium and other soft switches and PBXs. It is used to enable VoIP connections between servers, and between servers and clients that also use the IAX protocol.

IAX now most commonly refers to IAX2, the second version of the IAX protocol. The original IAX protocol has been deprecated almost universally in favor of IAX2.

NAT Transversal

NAT (Network Address Translation) is a technology most commonly used by firewalls and routers to allow multiple devices on a LAN with ‘private’ IP addresses to share a single public IP address. A private IP address is an address, which can only be addressed from within the LAN, but not from the Internet outside the LAN. In order to let a device with a private IP address communicate with other devices on the Internet, there needs to be a translation between private and public IP addresses at the point where the LAN connects to the Internet, that is within the firewall/router connecting the LAN to the Internet. Such a translation is commonly referred to as NAT (for Network Address Translation) and a router doing such translation is often called a NAT router or NAT firewall/router. Sometimes NAT is also called IP Masquerading. The passing of traffic through NAT is called NAT Traversal.

Packet Loss

VOIP is not tolerant of packet loss. Even 1% packet loss can “significantly degrade” a VOIP call using a G.711 codec and other more compressing codecs can tolerate even less packet loss. (Intel whitepaper)

Cisco says: The default G.729 codec requires packet loss far less than 1 percent to avoid audible errors. Ideally, there should be no packet loss for VoIP


An IP PBX is a private branch exchange (telephone switching system within an enterprise) that switches calls between VoIP (voice over Internet Protocol or IP) users on local lines while allowing all users to share a certain number of external phone lines. The typical IP PBX can also switch calls between a VoIP user and a traditional telephone user, or between two traditional telephone users in the same way that a conventional PBX does. The abbreviation may appear in various texts as IP-PBX, IP/PBX, or IPPBX.

Predictive Dialer

A predictive dialer is a computerized system that automatically dials batches of telephone numbers for connection to agents assigned to sales or other campaigns. Predictive dialers are widely used in call centers.

Prepaid Model

Prepaid refers to services paid for in advance. Examples include tolls, pay as you go cell phones, and stored-value cards such as gift cards and preloaded credit cards. Prepaid options can have substantial cost reductions over postpaid counterparts because they allow customers to monitor and budget usage in advance.


The public switched telephone network (PSTN) is the network of the world’s public circuit-switched telephone networks, in much the same way that the Internet is the network of the world’s public IP-based packet-switched networks. Originally a network of fixed-line analog telephone systems, the PSTN is now almost entirely digital, and now includes mobile as well as fixed telephones. It is sometimes referred to as the Plain Old Telephone Service (POTS).


SIP Express Router (SER) is a high-performance, configurable, free software SIP (cit. RFC 3261 ) server . It can act as SIP registrar, proxy or redirect server. SER features presence support, RADIUS/syslog accounting and authorization, XML-RPC-based remote control, etc. Web-based user provisioning, serweb, is available. SER’s performance allows it to deal with operational burdens, such as broken network components, attacks, power-up reboots and a rapidly growing user population. SER can be configured for many scenarios including small-office use, enterprise PBX replacements and carrier services.

SER is publicly available under the terms of the GNU General Public License.


“The Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences.” (cit. RFC 3261). It was originally designed by Henning Schulzrinne (Columbia University) and Mark Handley (UCL) starting in 1996.

The latest version of the specification is RFC 3261 from the IETF SIP Working Group. In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IMS architecture. It is widely used as a signaling protocol for Voice over IP, along with H.323 and others.

Soft Phone

A softphone is a software for making telephone calls over the Internet using a general purpose computer, rather than using dedicated hardware. Often a softphone is designed to behave like a traditional telephone, sometimes appearing as an image of a phone, with a display panel and buttons with which the user can interact. A softphone is usually used with a headset connected to the sound card of the PC, or with a USB phone.

Soft Switch

A softswitch is a central device in a telephone network which connects calls from one phone line to another, entirely by means of software running on a computer system. This work was formerly carried out by hardware, with physical switchboards to route the calls.

Sub Account

A sub-account can be used to break down an account into multiple smaller accounts. This could be for better tracking of detailed budgets and expenses, connect multiple PBX, connect devices from different locations or simply to connect multiple devices to our service without the need of a PBX. With the use of sub accounts, the customer doesn’t need to open multiple accounts and manage multiple balances to use the service.


Call Termination, also known as voice termination, refers to the handing off or routing of calls from one telephone company, also known as a carrier or provider, to another telephone company.

The terminating point is end point. The originating point is the party who initiates the call.

This term is highly used when referring to calls while using voip: a call initiated as a VoIP call is terminated using the PSTN The opposite of call termination is call origination, where a call initiated from the PSTN is terminated using VoIP.


the asterisk-based solution, trixbox, enables the home or small business user to quickly set up a VOIP Asterisk-based PBX. A web GUI makes configuration and operation easy. http://www.trixbox.com

Sources: whatis.com, wikipedia, voip-info.org, Asterisk