February 4, 2007

Is HD VOIP a New Trick or Old Trick?

Iristel is the first to bring HD VOIP to Canada.

Whoa…I don’t think I can handle this. We just recently enduring a long and arduous process of researching and purchasing our first “big” screen TV with…heaven help me…HD-high definition. You see in Canada it’s important to actually see the puck when the Canucks are playing. Apparently this is impossible on a 21″ tube TV with a pink stripe across the top of the screen. (Who knew?)

Maybe fellow bloggers can help me out here, but is HD VOIP something that consumers should consider when choosing a VOIP service, or is it more marketing lingo to work through? As I understand it, HD, high definition, or wideband VOIP refers to voice sampling at 16 kHz rather than at the measley 8 kHz supported by the PSTN and just about everyone else. If you capture voice with a wider spectrum of frequencies, the quality is better. They say it’s like comparing the quality of AM and FM radio. (See this article from voip-info.org).

Sounds good to me, but the catch is you have to have HD end to end in a VOIP call. It’s no good having 16 kHz at one end and 8 kHz at the other. In fact, 16 kHz downsampled to 8 kHz (which happens if you are calling a landline) may sound worse than 8 kHz from start to finish.

So I guess you can make the argument that HD or wideband VOIP is wasted on the masses when most calls travel partially over the PSTN or use VOIP networks using an 8 kHz voice capture process.

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September 12, 2006

Bandwidth and How it Affects VOIP

Reprinted with permission from: www.quickstartvoip.com 

You may have heard this before: Voice Over IP (VOIP) needs a certain amount of bandwidth to be usable. In other words, VOIP needs bandwidth so that voice conversations are intelligible by both parties. Your broadband Internet connection (provided through a cable or DSL service) provides that bandwidth.

Basically, bandwidth is the amount of digital data that can be transmitted in a certain time period. Internet connections are usually measured in KiloBits Per Second (kbps). Your connection may be rated at 512/128 kbps. This means that the upload stream (leaving your computer) has a maximum bandwidth of 128 kbps and download stream (coming to your computer) has a maximum bandwidth of 512 kbps. Find out the speed of your broadband connection…

For example, a web page that is 3000 bytes can theoretically viewed in 0.05 seconds – almost instantaneously - at a download speed of 512 kbps. However, in the real world web pages don’t usually load that fast.  That’s because data is held up, or delayed by various factors. Latency is the delay caused by the physical limitations of each piece of hardware the data is passed through. Delays are also caused by pathway congestion, error checking, transmission negotiations and extra data sent with the web page to identify its type, origin and destination.

What this means for VOIP is that enough bandwidth must be supplied to allow for the transmission of the actual voice data in real time as well as extra bandwidth for the overhead required for any data transmission.

So how much bandwidth does voice data use up? Well that depends on the codec (enCOder/DECoder) used to compress the data. Anywhere from 16 to 64 kbps is normal. Add onto that the extra overhead of about 10 to 24 kbps. In total, VOIP uses 26 to 88 kbps. As a general rule, assume that you will need at least 88 kbps to use VOIP reliably. Note that VOIP service providers will provide you with their broadband requirements.

You will need an Internet connection that can handle at least 128 kbps on the upload side. Extra bandwidth will allow others in the household to surf the net or download files while you are talking on the phone. If you wish to use conference calls even more bandwidth will be required.

While there is no set rule for the amount of bandwidth that a household will need, count on 128 kbps as a minimum.  If you have several computers and other devices connected to the Internet, consider going up to the maximum available.

Note that these speed are for uploads – the download speeds are usually quite a bit faster and present no problem for VOIP.

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August 11, 2006

Cast Off! The VOIP Girl is Underway…

Hi and welcome to my blog about VOIP, voice over IP, Internet phone, broadband telephone, or whatever you feel inclined to call it. This is NOT a blog about women, or even necessarily about women who use VOIP services–I mean how boring is that. I called it The VOIP Girl because well I’m a gal. And I want to write about this industry. That’s about it. What really prompted me to get going however, is the fact that there are no women commentators on this subject. I’m curious to see if my perspective will be any different.

I want to write about how consumers (that’s you and me) are doing using VOIP.  I don’t necessarily care about IPOs, who’s buying who, and all that insider industry stuff. There are tons of blogs out there talking about all that. In fact I would say that is mostly what you’ll find when you look for VOIP commentary in the blogosphere. It’s great stuff, but really I’m interested in how ordinary folks are using Vonage, Skype, Gizmo, etc., and how is the free stuff stacking up against heavy hitter paid services.

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September 11, 2006

Comparing VOIP Services Before You Buy

From: www.quickstartvoip.com

Voice over the Internet phone service is no longer an exclusive preserve of techies. Scores of telecom companies, cable companies and young entrepreneurs have set up VoIP services.

Each one claims to offer extra features and benefits. So how do you decide?

If price/cost is your most important deciding factor, VOIP comparison sites are great. They extract all the rate info and display it in a handy chart. Voipreview.org is a good example. Click here to compare the VOIP phone service and prices being charged by different providers.

The services that you should take a close look at are basic services, advanced services, voice mail, faxes, call blocking, web management, special calling, and customer service.

The most common services being offered by VoIP providers under the different service heads are:

  1. Basic Features: Call forwarding, international call forwarding, caller id by number, caller id by name, call-waiting, call-waiting caller id, disable call waiting, distinctive ringing, repeat dialing, return dial and three-way calling.
  2. Advanced Features: Call Transfer, conference bridging, simultaneous ringing, sequential ringing, secondary virtual phone number, additional service lines, toll-free numbers (incoming), MS Outlook integration and softphone support.
  3. Voice Mail: Retrieval of voicemail from telephone handset, phone number for external retrieval, retrieval via web interface and receiving of voice mail via e-mail
  4. Fax Functionality: Support outgoing/incoming faxes, receive faxes via voice mail, and a dedicated fax line.
  5. Call blocking/filtering: Block outgoing international calls, block outgoing 1-900 calls, block incoming anonymous calls, do not disturb notice. This also includes blocking of telemarketing calls or selective blocking and selective forwarding via e-mail of filtered calls
  6. Web Management: Modify basic/advanced features, obtain detailed call logs, activate order/cancel features/services, activate click to call facility, and provision of web interface that is compatible with non-IE browsers.
  7. Special Calling: 911 Emergency calling, 411 Information, free in-network calls, free calls to external VoIP networks, Cable box/SatTV/Tivo compatibility
  8. Customer service: Technical support via telephone, technical support via email, web-based technical support and account management by telephone.

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October 11, 2006

My wish list: fixing voice quality on the fly

One thing I wish for in my quest for perfect Internet phone service the ability is to fix voice quality problems on the fly or at least be able to understand “is it me or the network?”

When I had Vonage service, I usually started the converation with “So how do I sound, can you hear me?” Many times there was such a delay that practical converation was impossible. If I was at my computer, I’d try quitting applications to see if that helped (usually not). After hanging up in frustration, I’d try messing with the dreaded ”Bandwidth Saver” feature. But in my opinion, AFTER the call is way too late.

Is it possible to build intelligent applications that KNOW when they’re not performing optimally and can tell you what the problem is, or better yet advise how to fix it?

No sooner had I written this when Peter Csathy, CEO of SightSpeed, let me in on a little secret (okay well it’s not actually a secret…), SightSpeed detects when your video or voice call isn’t going so well and automatically makes adjustments to improve the quality.

For example, a video call that appears jumpy or out of sync with the audio could be caused by network congestion. SightSpeed can downsize the bandwidth usage for video so that at least the voice aspect of the call is preserved or improved. SightSpeed lets you know it’s doing this by popping up a dialog box that says it’s making these adjustments. I don’t know if you can you make adjustments yourself DURING a video call, but you can easily see your upload and download speeds while talking to someone.

To see SightSpeed statistics: 

With your mouse hovering over the SightSpeed title bar,  press CTRL + S (or right mouse-click followed by CTRL + S). A statistics dialog box appears showing your current, peak and average bandwidth usage.

Gizmo Project has a Call Quality Assistant (click the bar graph in the bottom left of the Gizmo Window) that gives you an idea of your network conditions. While it shows you at a glance the quality of your connection, it doesn’t say what you should do about it and at this point in my life, that’s what I need.

 

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September 15, 2006

Quick Guide to VOIP Phone Terminology

Reprinted with permission from: www.quickstartvoip.com

When shopping around for a VOIP phone, no doubt you’ll come across buzzwords like: softphone, SIP phone, IP phone, and Internet phone.

Do all these terms mean the same thing, or is there a difference? Let’s take a look.

VOIP phone or Broadband phone: A handset used to make VOIP calls. A VOIP phone has an Ethernet port (RJ-45) instead of regular phone jack (RJ-12), and they connect directly to a broadband Internet modem. A VOIP phone does not require a computer or softphone application to make or receive calls. A high speed Internet connection and subscription to VOIP phone service is all that is required. A SIP phone is a type of VOIP phone.

SIP Phone: A VOIP handset that complies with the SIP standard for voice over IP. SIP is the latest open standard, succeeding H323 standards. Bottom line, SIP compatibility is good. Phones that compete with SIP use proprietary signaling protocols for voice over IP. SIP softphones are VOIP software applications that comply with SIP standards.

IP phone: An umbrella term but used to mean any phone that can be used on an IP network (like the Internet). An IP phone may comply with either proprietary or open standards for voice signaling. An IP phone doesn’t have a RJ-12 connector like regular phones.

Internet phone: This term is used pretty loosely and depending on the context can mean the same as IP phone.

Softphone: A software application that lets you make calls over the Internet using the mouse or keyboard to dial phone numbers. To use a softphone, your computer must have an sound card, plus a speakers or headset, and a microphone. A USB phone can take the place of headset and microphone. Softphones are often free to download. Free VOIP software such as Skype and Free World Dialup are two popular choices.

USB phone: A handset that connects to the USB port on your computer. It is used for convenience when dialing from your computer, but it requires that a softphone application be installed first. Instead of using the softphone with a headset or microphone, the USB phone looks and acts like a regular phone and keypad. USB phones require driver software to be installed on the computer.

WiFi phone or WLAN phone: A handset used for making wireless VOIP calls. It has a  built-in WiFi transceiver unit instead of an Ethernet port. When you talk over WiFi, the phone connects wirelessly to a WiFi base station and from there to the Internet and a remote VoIP server. A computer or softphone is not required to make and receive VoIP phone calls. All that is required is access to a WiFi base station. Many cell phone companies are developing handsets with WiFi capability. This means you can make calls on the regular cell phone network and make VOIP calls on a local WiFi network (called a WiFi hotspot).

Skype phone: A handset that is much like a USB phone, except that the softphone application used to make the calls is Skype. A Skype phone can only make calls using the proprietary Skype phone software running on your computer. Linksys recently announced the release of its CIT200 wireless Skype handset which makes using Skype convenient from anywhere in the home.

Web phone: A marketing term that has been used to mean many things, both softphone and IP phone-related.

Net phone: Same as IP phone

Computer phone: See softphone, PC phone or USB phone.

PC phone: See computer phone, softphone or USB phone.

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September 11, 2006

VOIP Transmission in a Nutshell

Reprinted with permission from: www.quickstartvoip.com

In order to use VoIP, both parties need a broadband connection. This is a high-speed Internet, or broadband connection usually provided by a cable or DSL modem. Broadband modems are usually used to connect computers to the Internet, but in the case of VoIP, computers are not necessary.

The simplest form of VoIP is a computer-to-computer voice connection. All that is required for this type of connection is a computer, a headset consisting of earphones and microphone, and VoIP software. Most software packages are free and allow you to connect to any computer running the same software. There is no charge for this type of connection and calls can be made to anywhere in the world.

VoIP software can also be used to connect to land-line phones — that is, phones which are not connected directly to the Internet. This type of call is usually not free but the cost is quite a bit lower than what your telephone company charges. Some VoIP services also allow you to make calls to cellular phones.

The only time that both parties need a particular VoIP software package is when they are making computer-to-computer calls. Parties receiving land-line or cellular calls do not need any extra equipment or software.

VoIP Transmissions

VoIP is based on digital data transmission. The first step in any VoIP call is to convert the analog signal of the human voice into digital data. This is done within an Analog-to-Digital Converter (ADC) that divides an analog signal into discrete steps which are represented by numbers. The next step is to compress the audio data using a codec (enCOder/DECoder) which significantly reduces the amount of digital data while maintaining audio quality.

The compressed digital data can now be sent over the Internet. The data stream must be divided into packets which, besides containing the audio data, also have information concerning their destination and their place in the data stream.

All data that is sent over the Internet is encapsulated in ‘layers’ which aid in its proper delivery. For example, web pages may use the Internet Protocol (IP) network layer to specify destination and origin addresses, the Transmission Control Protocol (TCP) transport layer to create a connection between two computers and the Hyper Text Transfer Protocol (HTTP) as an application layer to allow the Web browser to display the web page correctly.

Most VoIP uses a transport layer called User Datagram Protocol (UDP) which is faster than TCP. A commonly used application layer is Real-time Transmission Protocol (RTP) — originally developed for delivering audio and video over the Internet. RTP provides information about the sequence of the data packets so they can be reconstructed in the correct order at their destination.

RTP also has the ability to drop packets if they do not arrive within a certain amount of time. This is necessary for telephone conversations because if the telephone software waited for every packet of information to arrive before reassembling it there would be unacceptable delays in the audio stream.

Even though some of the packets are dropped, there is usually still enough information to make the conversation legible. The number of packets that will be dropped depends on the speed of your Internet connection in the distance between the two parties.

Once the voice data has arrived at its destination, it is reassembled in the correct order and converted back from digital to analog.

 

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September 12, 2006

PhoneBoy Reviews GotVoice

TheVOIPGirl.com’s had another nice welcome from PhoneBoy (no, we’re not related), a prolific VOIP, telecom and technology blogger with an affable writing style. He explains things…

Check out his review of GotVoice and you’ll see what I mean. GotVoice is an interesting service that takes voicemail messages from different voicemail services (including VOIP ones) and sends them to your email inbox. You get convenient access to all your voice messages in one place. He also points out a few shortcomings of TheVoipGirl.com that I hasten to address. Thanks PhoneBoy!

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February 1, 2007

VOIP Spam–How bad is it?

I’ve been reading this week about NECs VOIP Seal anti-spam tool. VOIP Seal can detect if a call is computer generated and block it based on voice patterns. Alec Saunders also touched on it and pointed out that iotum is already anticipating the need for SPAM blocking with simple call screen capabity. GrandCentral does the same; they’ve got a SPAM registry where you can nominate the worse offenders so that everyone can block their caller IDs.

While I’m sure we’ll be hearing much more about the woes of phone SPAM in the future, I hope it’s something we won’t look back on and think, “If only we had done something back then…” In today’s report on M&C Tech News, they report the following statement from NEC:

“the existing infrastructure for producing spam e-mails (so called “botnets”) can easily be modified to also produce spam telephone calls. Today, the number of spam emails is higher than the number of regular emails produced jointly by all of the users in the internet. If unsolicited marketing and spam calls become as frequent as spam email, constantly-ringing VoIP phones may hinder the spread of their use.”

Alec talks about receiving annoying SPAM calls at the iotum office and we fielded one today at our house. A computer generated “Win a free trip” thing.

I’d be interested to hear from you if you’ve been getting spammed more than usual.

Let me know!

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September 11, 2006

VOIP Glossary

Analog audio signals: Analog audio signals are used to transmit voice data over telephone lines. This is done by varying or modulating the frequency of sound waves to accurately reflect the pitch of the sound. The same technology is used for radio wave transmissions.

ATA: ATA or the analog telephone adaptor is the hardware device that connects the conventional telephone to the Internet through a high speed bandwidth line, provides the interface to convert the analog voice signals into IP packets, delivers dial tone and manages the call setup.

Bandwidth: Bandwidth is the volume of data that can be transmitted over a communication line in a fixed amount of time. It is expressed in bits per second (bps) or bytes per second for digital devices and in cycles per second, or Hertz (Hz) for analog devices. Bandwidth can also be defined as the difference between a band of frequencies or wavelengths.

Broadband: It is a term used to define high speed Internet connection, generally provided by cable TV, DSL or dedicated telecom lines. The high speeds are achieved by the carrying capacity of the cable that can carry multiple messages simultaneously.

Cable modem: The cable modem is a device that is used to connect a computer to the high speed coaxial cable run by cable TV companies to provide access to the Internet. The connection is made through an Ethernet port, which is a shared medium and can affect download speeds if too many users log on simultaneously to the Internet on that particular cable segment. However, despite this cable modems provide extremely fast access to the net.

Circuit switched networks: These networks have been used for making phone calls since 1878. They use a dedicated point-to-point connection for each call. This reduces their utility because no network traffic can move across the switches that are being used to transmit a call.

Client (Softphone client): The software installed in the user’s computer to make calls over the Internet.

Codec: Codec is a term that arises from the Compressor-Decompressor or enCOder/DECoder process. It is used for software or hardware devices that can convert or transform a data stream. For instance, at the transmitting end codecs can encode a data stream or data signal for easy transmission, storage or encryption. At the receiving end, they can decode the signal in the appropriate form for viewing. They are most suitable for videoconferencing and streaming media solutions.

Compression: This is a term that is used to indicate the squeezing of data in a format that takes less space to store or less bandwidth to transmit. It is very useful in handling large graphics, audio and video files.

Data compression: This is the process that is used to compress large data files into mall files so that they use less bandwidth during transmission and less disk space when stored. The compression depends upon the repeatable patterns of binary 0s and 1s. The higher the number of repeatable patters, the higher is the compression. The right compression codes can compress data files to 40% of their original size. The graphics files can be compressed even more – from 20% to 90%.

DSL modem: A DSL modem is a device that is used to connect one or more computers to the high speed DSL line provided by a DSL operator to gain access to the Internet. The customers use these modems to log on the net to download or transmit data. Since the DSL lines have high bandwidth capacity the data transfer speeds are very high.

E911: E911 is the short form of the term Enhanced 911, and is used for providing emergency service on cellular and Internet voice calls.

Emergency 911 calls: This is an emergency telephone number that handles all calls related to police, fire or medical emergencies. The number, which is allotted under the North American Numbering Plan (NANP), is answered by either a telephone operator or an emergency service dispatcher, who, in turn, alerts the appropriate emergency service.

H.323: An ITU standard that lays down guidelines for real time voice and videoconferencing utilities on the Internet. The H.323 standard supports voice, video, data, application sharing and whiteboarding and defines media gateways for conversion to packets.

Internet congestion: Internet congestion occurs when a large volume of data is being routed on low bandwidth lines or across networks that have high latency and cannot handle large volumes. The result is slowing down of packet movement, packet loss and drop in service quality.

IP address: An IP address, also known as Internet Protocol address, is the machine number used to identify all devices that are connected to the net. Each device has its own unique number which it uses to communicate. This number is fixed in the case of those computing devices that have a fixed IP address. The rest are allotted a dynamic IP address, which is valid for the period they are connected to the net. The numbers range from 0.0.0.0 to 255.255.255.255.

IP mapping: IP mapping is the process of identifying IP addresses on the basis of their geographical locations. The mapping enables web administrators to pinpoint the location of any computing device connected to the Internet.

IP Phone: An IP phone is one that converts voice into digital packets and vice versa to make phone calls over Internet possible. It has built-in IP signaling protocols such as H.323 that ensure that the voice is routed to the right destination over the net. The IP phones come with several value added services like voicemail, e-mail, call number blocking etc.

IP telephony: IP telephony refers to the two-way transmission of voice over Internet. The voice is transmitted in real time by using the packet-switched technology over the IP network. Some of the applications that use IP telephony are IP-based phone services, voice over instant messaging and videoconferencing.

IP: IP, which is the acronym for Internet Protocol, defines the way data packets, also called datagrams, should be moved between the destination and the source. More technically, it can be defined as the network layer protocol in the TCP/IP communications protocol suite.

ITU: ITU, which is the acronym of International Telecommunication Union, is a telecommunications standards body based in Geneva. It works under the aegis of the United Nations and makes recommendations on standards in telecommunications, information technology, consumer electronics, broadcasting and multimedia communications.

Jitter: It is a term used to indicate a momentary fluctuation in the transmission signal. This happens in computing when a data packet arrives either ahead or behind a standard clock cycle. In telecommunication, it may result from an abrupt variation in signal characteristics, such as the interval between successive pulses.

Kbps: Kbps is the acronym for kilobits per second and is used to indicate the data transfer speed. If the modem speed, for instance, is 1 Kbps then it means that the modem can route data at the speed of one thousand bits per second.

Lag: Lag is the term used to indicate the extra time taken by a packet of data to travel from the source computer to the destination computer and back again. The lag may be caused by poor networking or by inefficient or excessive processing.

Latency: Latency is the time that elapses between the initiation of a request for data and the start of the actual data transfer. This delay may be in nanoseconds but it is still used to judge the efficiency of networks.

Mapping: The process of identifying all related data fields or data streams and putting them in an easily identifiable context. For example, IP mapping enables users to pinpoint the geographical location of any computing device on the Internet.

MGCP: Acronym of Media Gateway Control Protocol. Used for a Voice over IP system. It consists of a Call Agent and a set of gateways, of which at least one works as the “media gateway” and performs the conversions.

NANP: Stands for North American Numbering Plan. A telephone numbering system that has evolved the way area codes and numbers are allotted. The system was established in 1947 and covers the United States, Canada and a few neighboring areas. It uses a three-digit area code and seven-digit telephone numbers. Its fiat is, however, limited to the public switched telephone networks only.

Net Phone: A net phone uses the Voice over IP technology to make voice calls. These calls are made by converting analog sound signals into digital data packets, and then moving the packets to their destination over the net.

Packet loss: Packet loss is the term used to indicate the loss of data packets during transmission over a computer network. This may happen on account of high network latency or on account of overloading of switches or routers that are unable to process or route all the incoming data.

Packet switched networks: These are networks that break messages into small digital packets, stamp each packet with the destination IP address, and route them across different channels to their destination where they are reassembled in their proper sequence. This is done to avoid network congestion and speed up data movement from multiple sources.

Packet: A packet is a unit of data transmitted over the network in a packet-switched system. It consists of a header that stores the destination address, a data area which carries the information that is being transmitted, and a trailer which contains information to prevent errors during transmission.

Peer-to-Peer (P2P): The term peer-to-peer is used to indicate a form of computing where two or more than two users can share files or CPU power. They can even transmit real time data such as telephony traffic on their highly ad hoc networks. Interestingly, the peer-to-peer network does not work on the traditional client-server model but on equal peer nodes that work both as “clients” and “servers” to other nodes on the network.

POTS: POTS is the short form of plain old telephone service. It transmits voice as analog data on communication lines that are much slower when compared to today’s ISDN or FDDI lines. However, not long ago POTS, which is also known as the public switched telephone network, was the standard telephone system across the world.

Processor drain: This is a term used to indicate a drop in the quality of VoIP phone service when a user opens several applications on his computer simultaneously.

Protocol: It is a convention or standard that defines the procedures to be adopted regarding the transmission of data between two computing end points. These procedures include the way the sending device should sign off a message or how the receiving device should indicate the receipt of a message. Similarly, the protocols also lay down guidelines for error checking, data compression, and other relevant operational details.

PSTN: PSTN, which stands for Public Switched Telephone Network, refers to the telephone system that transmits analog voice data. Till recently, PSTN was the heart of all phone systems worldwide. However, most of the developed world is now switching to or has switched to telephone networks that are based on digital technologies, such as ISDN and FDDI. RJ45: RJ45, which is the acronym of Registered Jack-45, is a telephone connector that is used in Ethernet and Token Ring Type 3 devices. It has eight “pins” or electrical connections.

Router: A router is a network device that that handles message transfer between computers that form part of the Internet. The messages, which are in the form of data packets, are forwarded to their respective IP destinations by the router. A router can also be called the junction box that routes data packets between computer networks.

Sampling: This is a methodology used to measure the value of an analog signal at regular intervals, and encoding it into a digital format for VoIP phone services.

Service provider: A service provider is a business entity that provides a communication, storage or processing service for a fee. Some of the service providers in the digital world are the Internet service provider (ISP), application service provider (ASP), storage service provider, mobile phone service provider, web hosting provider, and of course, VOIP service provider.

SIP phone: A SIP phone is a telephone that uses the SIP (Session Initiation Protocol) standard to make a voice call over the Internet. The SIP phones come with several value added services like voicemail, e-mail, call number blocking etc. There are no charges for making calls from one SIP phone to another, and negligible charges for routing the call from a SIP phone to a PSTN phone.

SIP: SIP, which is the acronym of Session Initiation Protocol, is an IP telephony signaling protocol. It is primarily used for voice over IP (VoIP) calls, though with some extensions it can also be used for instant messaging. It is less complex than H.323, the other IP telephony protocol.

Skype: Skype is a peer-to-peer Internet telephony company that revolutionized the way voice calls are made by using VoIP technology. The company, which has been acquired by eBay, was founded by Niklas Zennström and Janus Friis. Skype users can speak to other Skype users for free, but have to pay a small fee for calling or receiving calls from conventional phones.

Soft switch: It is a software application that is used to keep track of, monitor or regulate connections at the junction point between circuit and packet networks. This software is loaded in computers and is now replacing hardware switches on most telecom networks.

Softphone: This is a software application that is installed in the user’s PC. It uses the Voice over IP technology to route voice calls over the net and provides several value added features, such as call forwarding, conference calling, and integration with applications such as Outlook for automatic dialing The audio is provided through a microphone and speakers plugged into the sound card. The only limitation of a Softphone is that the phone call has to made through a PC. Many softphone are free VOIP software downloads.

Voice chat: This is an application that enables two or more than two individuals to carry on a verbal conversation over the Internet. Voice chat is also known as audio-conferencing or telephone conferencing on the net.

Voice over IP (VOIP): VoIP or Voice over IP is the technology that is used to transmit voice over the Internet. The voice is first converted into digital data which is then organized into small packets. These packets are stamped with the destination IP address and routed over the Internet. At the receiving end the digital data is reconverted into voice and fed into the user’s phone.

Voicemail: It is a telephone messaging system that digitizes the analog voice signals and stores them on disk or flash memory in a central computer. These messages can then be retrieved by users by logging on to the server or forwarded to another voice mailbox. Most voice mail systems have auto attendant capabilities, that is they can use prerecorded messages to route callers to the appropriate person or mailbox. Voicemail is usually a free feature in VOIP service plans

IM: IM, which stands for Instant Messenging, is a software that allows users to exchange messages in real time. However, to do so both the users must be logged on to the instant messaging service at the same time. Some of the popular IM services are: MSN Messenger, AOL Instant Messenger, Yahoo! Messenger, Google Talk and ICQ.

VOIP Gateway: This device provides the conversion interface between the public switched telephone network (PSTN) and an IP network for voice and fax calls. Its primary functions include: voice and fax compression/decompression, packetization, call routing and control signaling. It also provides an interface to Gatekeepers or Softswitches, billing systems, and network management systems.

VOIP PBX: VoIP PBX, which stands for Voice over Internet Protocol Private Branch eXchange, is a telephone switch that converts IP phone calls into traditional circuit-switched TDM connections. It also supports traditional analog and digital telephones.

VOIP Phone: A VoIP phone is one that uses the Internet to route voice calls by converting the voice data into IP packets and vice versa. The phones come with built-in IP signaling protocols such as H.323 or SIP that help in the routing of data to the right destination. A VoIP phone can also be a software application that is installed in the user’s PC. In this case it is known as the Softphone. Also, the calls in this case have to be made from the PC, and not through a telephone instrument.

VOIP services: The VoIP services are packet-based services that use the Internet to move voice data. These services are much cheaper than the traditional PSTN services because the investment in infrastructure is low. They also come with several value added features which make them more lucrative than the conventional landline phone services.

Web phone: A web phone is a device that allows users to make voice calls over the Internet.

WiFi Hotspot: An area where a wireless access point enables users carrying wireless-enabled laptops to log on to the Internet. The limiting condition is that the access point is configured to broadcast its presence and does not require authorization for access. Generally, WiFI hotspots are located in public places like airports, train stations, libraries, marinas, convention centers, coffee shops and hotels.

WiFi phone: A WiFI phone is one that enables users to make phone calls from public WiFi hotspots or residential WiFI network environments. Besides voice calls, these phones can be used to send e-mails wirelessly.

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