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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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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October 3, 2006

Guide to all those Skype icon things…part 2

Well, I started Guide to all those Skype icon things…part 1, as a cheat sheet for myself. Here’s a few more.

Click the History tab to see:

skype-outgoingcallhistory.jpg Outgoing calls

skype-incomingcallhistory.jpg Answered incoming calls

skype-missedincomingcallhistory.jpg Missed calls (unanswered incoming calls)

skype-chathistory1.jpgChats

skype-newvoicemailhistory.jpgNew voice messages (unplayed)

skype-voicemessagehistory.jpgVoice messages (played)

skype-transferfileshistory.jpg Transferred files

skype-smshistory.jpg SMS messages sent

 

 

 

 

 

 

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February 8, 2007

More on MobileSTICK

Today BridgePort Networks announced the commercial availability of MobileSTICK, a USB device that transforms your PC or laptop into an extension of your cell phone. When you plug MobileSTICK into a PC or laptop with a fixed line or Wi-Fi broadband connection, the device launches a PC softphone, secured by a SIM card that utilizes the your existing mobile phone number to make and receive phone calls, SMS messages and multimedia (MMS) messages. Your existing mobile remains active and continues to utilize its existing SIM.

Ken Camp had the opportunity a few months ago to do a podcast with BridgPort’s Sanjay Jhawar that explains the MobileSTICK technology in detail. It’s really good and worth a listen.

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December 6, 2006

GrandCentral and TalkPlus, Tell Me the Difference

I’ve been wondering about the GrandCentral “One number for life” mantra, and TalkPlus‘ second number for your mobile phone. While I understand these products target different markets, I find it hard to keep them distinct in my mind. It dawned on me that this could be a problem that consumers have as well.

There’s a lot of chatter about Voice 2.0 applications–the user in control not the network–but that is the VOIP blogosphere talking. Me, I think okay, I get another number, it’s free for now, and it’s solving what problem exactly? 

GrandCentral:

Provides a phone number not tied to a device or location. When people call this number, the phones you have “attached” to it ring, like your cell phone, home phone, and office phone. Up to six phones can be linked to your GrandCentral number. The service is all about giving you control over how people reach you (inbound calling) as opposed to how you place calls. Callers can leave voice messages that can be checked by phone, email or online. You’re notified of a voice mail via email or text message to a cell phone. And you can flag unwanted callers as spam. It’s also free.

You need it if people have a hard time tracking you down. You find yourself playing telephone tag. You WANT to be found but heck, you’re always bouncing between work, home, and on the road.

TalkPlus:

Provides a second number for your cell phone so you can separate personal life and work life. The number can be used as a second line for business, dating, classified ads, online auctions, social groups, or a second residence. Like GrandCentral, unwanted callers can be blocked while priority numbers ring through. When making outbound calls, you can specify which caller ID to use so that the person you are calling doesn’t know how or where you’re calling from.

You need it if your mobile phone is your primary means of communication but the  separation of work and play is important to you. You want people to know that you’re calling from the office (the caller ID says this is a work-related call) even though you’re calling from home or the beach in Maui. You are also concerned about personal privacy and want to make sure that your personal number is only available to the people you want to have it.

***

In talking with Craig Walker, GrandCentral CEO, he tells me they couldn’t be more different than TalkPlus.

“Our philosophy is that we don’t need MORE numbers for people to reach us at…we need less.  As long as I have the control over my inbound calling that GrandCentral gives me, there’s no reason that I would want to juggle different personas. I don’t want phone numbers that identify me as being located at a certain place or doing a certain thing, I want a phone number that is personal to me.  If you want to reach me, call my ONE NUMBER.  I’ll be able to answer it wherever I want…I will be able to know who’s calling every time, and I’ll even be able to listen in on the voicemail as its being left if I’m still unsure whether I want to take the call.  When somebody calls me, they shouldn’t be able to “figure out” or know where I am based on the number…if I’m in the office, working from home or on the beach, that’s my business.” 

Obviously there’s more to both GrandCentral and TalkPlus than what I describe here, and the enthusiasm voiced by many in the VOIP blogging community seems to be not so much what these services are doing today, but what we can expect from them in the future.

To set up a free GrandCentral account, click here. To sign up for a TalkPlus sneak peak beta (for Cingular, T-Mobile, and Sprint customers in select U.S. states only), click here. TalkPlus won’t be a free service like GrandCentral and pricing is to be determined.

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November 30, 1999

About

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 voice over ip. That’s about it. What really prompted me to get going however, is the fact that there are few 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.

Thanks for stopping by.

Leanne Tremblay
VOIPGirl

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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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April 12, 2007

Battle of the Js: Jaxtr, Jangl, and JaJah

Did you know that for the year 2006, “J” was the most popular letter of the alphabet? Okay, I made that up. I have no idea if it’s true, but three out of ten most popular baby names last year started with J….

I had a comment a few days ago from Eric C. wanting to know about Jaxtr, Jangl and JaJah. What’s the difference? What indeed! People (like me) are starting to ask questions like this because…who can remember what these guys do! They sort of sound alike, they’re in the same industry space, they appeal to the same type of folks. Eric then wanted to know how the Js relate to GrandCentral. Oiy!

Jaxtr:

  • Provides a click-to-call widget (button) for social networking sites like MySpace, web sites, and blogs. Visitors to your page have the option to call you, send you a text message, or leave you a voice message. They don’t need a microphone or headset. They call you by entering their phone number in the widget, Jaxtr then provides a local number for them to dial.
  • Free registration, then you buy jax credits to forward calls to your phone. 100 free credits per month.  You can send unlimited calls to voice mail for free and unlimited text messages for free. Calls to other Jaxtr users are also free.
  • No software download required.
  • Unique feature is Voiceblast. You can record your own message or greeting that’s played automatically or on mouse-click when a someone visits your web page. I’ve added my voiceblast to my About page here.
  • Like Jangl, Jaxtr has privacy options that allow you and the person calling to hide your phone numbers or email addresses.
  • Like GrandCentral, you can block incoming calls or forward calls based on caller ID. Unlike GrandCentral, Jaxtr does not give you a 10-digit phone number that can be dialed from any phone. People calling you are given a special 10-digit number they can use, but they must use the same phone every time. If they call from a different phone, Jaxtr provides a different number.
  • I like it.

Jangl:

  • Provides a click-to-call widget (let’s call this widget dialing) for social network sites, web sites and blogs that masks the incoming and outgoing numbers. It’s a way for people to make and receive calls without giving out phone numbers. I guess the advantage here is privacy. For example, you can post the Jangl widget on MySpace without revealing your personal phone number.
  • When you register, you receive a Jangl ID that people use to call you. Someone enters your Jangl ID in the online widget, Jangl then gives them a special number to call you on a regular phone. Your number and the calling person’s number are never exchanged.
  • Registration is free, then you pay per call based on telephone company charges.
  • You can choose from several cool looking widgets
  • Not really like GrandCentral at all, except in that they both have widget dialing (GrandCentral recently introduced web buttons).
  • Sounds a lot like Jaxtr but I still found Jangl a bit confusing and not a lot of info or online support. I haven’t used it much.

JaJah

  • Provides web-based dialing, or dialing from a web page, without a microphone or headset. All calls are made phone to phone, whether landline or mobile. This means the person you’re calling does not have to be connected to the internet.
  • You log in to your Jajah account, enter your friend’s number, and click the CALL button. Your phone will ring, you pick up, then your friend’s phone will ring.
  • No software download required.
  • Other Jajah services and tools include conference calls, call scheduling, access to Jajah phone book from the web browser on your mobile phone, Jajah plugins for Google, Outlook, Mac OS X Address Book, Firefox, and Plaxo
  • Registration is free, you get 5 minutes free calling anywhere, then you pay as you go. Calls to Jajah users are free.
  • I have an account but haven’t used it.

I’ll leave TalkPlus, Talkster, and Talk-Now for another time….”Tango of the Ts” perhaps? 

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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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May 15, 2007

Fring, Talkster, Nimbuzz and Barablu Reviewed

Laptop magazine has a review of Fring, Talkster, Nimbuzz and Barablu. These four apps turn your good old cell phone into an internet phone and more. Of the four, Nimbuzz is new to me and Barablu I haven’t tried yet. Fring and Talkster I’ve used and quite like, although I think having a great handset makes all the difference. I wouldn’t bother with any of these if I didn’t have the Nokia N80i to test with.

The review does a good job of capturing the typical setup process for all applications, and gives both the pros and cons of usability, call quality and the like. Author Joanna Stern points out that “regardless of which mobile VoIP service you use, you’re going to need an unlimited data plan, which costs anywhere from $5.99 per month (T-Mobile) to $24.99 (Cingular)”.

Edited May 17, 2007:

James Wanless of Talkster clarified for me that their service in fact does not require a data plan:

One thing that I wanted to point out to your readers though is that Talkster doesn’t require an unlimited data plan. In fact, quite the opposite. Once you have selected who you want to talk to, the voice portion of the call travels over the regular cellular voice channels and uses your “in plan” minutes. You can always count on the voice channel to be available and the quality to be consistent or at least a known quantity which we feel is the right approach given the point of evolution of cellular networks today.

Thanks James

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