Tuesday, May 30, 2006
The managing director of South Africa based internet service provider Posix Systems, Mark Elkins revealed at the AfrISPA forum in Kenya that South Africa has started issuing 087.. numbers for VoIP services. As a result of increasing use of VoIP technology in South Africa the SA regulators decided to create numbering type specifically for VoIP. They are also looking into the areas of interconnection and SIP access provision. VoIP has been gaining momentum in the African continent and not a few see it as an opportinuty to brigde the communication devide.
JaJah integrates PayPal
For users of JaJah who does not like using their credit card online there is now several payments options. The latest inclution is Paypal, users can now pay for their calls using their paypal account which is a secure way of making online payments. The figure to left shows all the payments option available to JaJah users.
Monday, May 29, 2006
Jajah include text messaging and Call Scheduling
Users of JaJah can now send SMS text messages directly from Jajah web activated telephony web interface. All they have to do is click on the sms button, enter number they want to send text message, enter the text and click send. Also its now possible to schedule your calls too. Enter the number you want to call, select the date and time you want the call to be made and Jajah will intitate a call from your phone to the destination at the specified time. With this exciting features Jajah will continue increase in popularity and i am not suprised that some other voip providers are allready adopting the web activated telephony approach to voip.
Voipbuster, Voipcheap, Voipstunt , Internetcalls and other Finarea voip clients now with text messaging, Direct Callback and Web activated telephony
Finarea has added exciting new features to all their voip clients and services. The latest version of voipbuster, voipcheap, voipstunt, internetcalls and others now come with sms sending and Direct callback capability. Its now possible to send text messages to mobile phones in countries with country code starting with +1.., +3.. and +4.. at the cost of 5 cents per message. Just enter your home or mobile phone number in the Direct Callback field of the application. You can now decide with each call if you want to place it via the application OR take your call on your regular phone. With the Direct Callback feature checked, the application will first dial your regular phone number. As soon as you have picked it up, a connection with the number you want to call will be made. This have the advantage that you can now make voip calls directly using your regular phone just like you use jajah web activated telephony. Also much like Jajah this direct callback feature is available directly from your account page. You just need to enter your phone number, the number you want to call and click call. Web activated telephony is becoming more popular and the era of making voip calls siting in from of a PC is quickly becoming a thing of the past.
Labels: Free International Calls, VoIP News
Tiger NetCom's GATE 104 - The VoIP Adapter with Fax function
China based Tiger Netcom Ltd has developed GATE 104 a VoIP adapter with fax function. GATE 104 is a SIP based voip phone adapter that also support fax function. It not only serve as an adapter for regular phone but also as an adapter for fax machines enabling sending of fax over ip networks. It has a very lightweight design thereby very suitable for travelling. It supports SIP 2.0 (RFC 3261), TCP/UDP/IP, RTP/RTCP, ICMP, ARP/RARP, DNS, DHCP, NTP, PPPoE, STUN and TFTP. One can make both VoIP-to-PSTN and PSTN-to-VoIP calls. It also supports remote configuration via TFTP server with encrypted configuration files. All major phone features such as caller ID, conference calling, call forwarding, call blocking, call hold, call waiting and call transfer are supported. More information about GATE 104 can be found at http://www.tigernetcom.com/Labels: Conference Calls, Fax-over-IP
Saturday, May 27, 2006
All Finarea voip clients now with 300 minutes per week limitation
Finarea, the company behind the popular voip applications such as voipbuster, voipcheap, voipdiscount, internetcalls etc are increasingly showing that they can not continue to substain the unlimited free calls they offer to several countries. This they are doing my constantly putting more and more limitations to the amount of free calls a user can make. Before now users of such applications are limited to 20 hours a month but just after few weeks of implementing the 20 hours limitation they reduced it futher to 300 minutes per week without any change of carrying over and unused minutes to the next week. This is not unexpected anyway as many of us has wondered how Finarea will be able to substain the unlimited free calls on a long time basis. Personaly i think the best business approach that is substainable on long time is to charge a fixed amount per month for unlimited free calls to popular destinations. I have proposed this approach on several of my post on this blog and AOL has taken this approach with the lauch of AIM Phonline with gives unlimited free calls to about 30 countries for a monthly subscription of $9.95.
Labels: Free International Calls
Friday, May 26, 2006
AIM Phoneline VoIP Service - Get free US phone, make free unlimited calls and more!
With the AIM Phoneline voip service the popular AOL Instant Messenger application has been transformed to an advanced voip application with lot of great features. The AIM Phoneline is the latest furray into the voip market by AOL. The AIM Phoneline voip application is very easy to install and configure, much thesame with AOL instant messenger. One very interesting feature about this voip service from AOL is that they are given away what others are selling, for example when you sign up for AIM Phoneline you get a free US phone line (nicknamed AIM Digits ) on which people can call you from any phone in the world. Receiving call on this number is free and unlimited. Also you get a free subscription to AIM phoneline voicemail which automatically answers your call and record audio messages when you are not available. It took me just some few minutes to download, install and setup AIM Phoneline. Both the voicemail and phone-in number worked very fine and the audio quality for both pc-to-pc and pc-to-phone calls were excellent. You can also subscribe to an interesting monthly plan for just $9.95 per month. With this monthly plan you can make unlimited free calls to landline and mobile phones in US, Canada, China, Hong Kong and Singapore. Also calls to landlines in about 30 other countries are free and unlimited with this monthly plan subscription. You can download and sign-up for AIM Phoneline at www.aimphoneline.com.
iBasis to provide International call termination for Yahoo Messenger with Voice
Yahoo has announced that the Bullington, Mass. based leading global VoIP company iBasis will be providing international call termination for the Yahoo! Messenger with Voice.. iBasis, founded in 1996 is a leading provider of high-quality termination services for calls from the U.S to international markets. iBasis network span over 180 countries and its customers include many of the largest telecommunications carriers in the world, including AT&T, Cable & Wireless, China Mobile, China Unicom, MCI, Sprint, Skype, and Telefonica. This will enable Yahoo messenger with voice to offer its customers high quality PC-to-Phone calling over iBasis global VoIP network at very competitive rates."Like the Internet itself, Yahoo! is ubiquitous," said Ofer Gneezy, president and CEO of iBasis. "By delivering innovative services and compelling features to their enormous customer base, Yahoo! is playing a major role in the transformation of how people communicate. They did it with Yahoo! Messenger, and now they will do it with Yahoo! Messenger with Voice. We're very excited to be an integral part of the service."
More Yahoo Links :
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Yahoo! Personals with 7 Day FREE Trial offer
Get quotes from Yahoo! Autos
Yahoo!Music Unlimited Free Trial
Yahoo! Top Games
Google partners Nokia - Add Google Talk to Internet Tablet Nokia 770
Many of us expected that Nokia will add a voip application to their new Internet Tablet Nokia 770 but no one expected its going to be Google Talk. The Wall Street Journal has reported that Google is teaming up with Nokia to add google talk to Nokia 770 and this is the VoIP and Instant Messaging software upgrade that Nokia is planning to release in the coming months. Users of Nokia 770 can now make free PC-to-PC and low cost PC-to-Phone calls anywhere they find themself using Google Talk.Labels: VoIP News
Thursday, May 25, 2006
JaJah add Click-to-Call to Plaxo Smart Address Book
JaJah has teamed up with Plaxo to add click to call capability to Plaxo smart address book. Plaxo users can now click on phone numbers of their contacts stored in their address book and directly initial a phone call from their regular or mobile phone to the desired destination phone. With this partnership Plaxo Members Can Now Enjoy the Benefit of Simple, Cheap Calls Without Headphones, Microphones, or Software Downloads and it will also increase the user base of JaJah web activated telephony.“We are excited to partner with Plaxo for two reasons,” said JAJAH co-founder Roman Scharf. “First, through the click-to-call integration with Plaxo, we can offer users even greater convenience. And second, partnering with Plaxo gets our offering in front of a user base that’s 10 million and growing rapidly.”As part of the partnership, Plaxo also will power the address books of all JAJAH members. Later this year, JAJAH will build on Plaxo’s Application Program Interface (API) to introduce the capabilities of the Plaxo smart address book to their members. JAJAH members will soon have direct and integrated access to the Plaxo smart address book, providing JAJAH members with a way to seamlessly transition between their work, home, and mobile devices, and have accurate information wherever they may be.The JAJAH integration is available in Plaxo Online, the web based version of the Plaxo service, and in the latest version of the Plaxo Toolbar for Outlook and Outlook Express. Members wishing to have JAJAH integration in their Outlook or Outlook Express address book can download the latest Plaxo Toolbar at http://www.plaxo.com/downloads. Plaxo, Inc., is used more than 10 million people in over 83 countries, and enjoys a wide following in academia, government, and both small and large businesses, including 98 of the Fortune 100 companies. Plaxo makes it easy for members to connect to their personal information, friends and colleagues in the manner that is most appropriate. The company provides members with a single, smart address book that can be used virtually anywhere; across multiple accounts and on numerous applications, including Outlook, Outlook Express, Mac, Thunderbird, Yahoo, Palm, and various mobile applications. Plaxo has created a network of people whose contact information is always up-to-date through seamless address book updating that occurs between linked members. Plaxo Premium Services expand the functionality of Plaxo and include Plaxo Address Book Optimizer, Plaxo eCards, Plaxo Mobile Access, and Plaxo VIP Support.Founded in 2001, Plaxo is a privately held company funded by leading investment and technology firms – Sequoia Capital, Globespan Capital Platforms, Harbinger Venture Management and Cisco Systems – as well as individual investors Ram Shriram and Tim Koogle.
Wednesday, May 24, 2006
Gizmo project 2.0 out with new features.
SIPphone, Inc has released new version of gizmo project softphone, the gizmo project 2.0. This version 2.0 comes with two very exiting new features, the first one is that it now works with Asterisk, an open source PBX software with growing popularity. Gizmo project 2.0 can now be configured to communicate with your Asterisk PBX. Incoming calls from your PBX will be clearly identified, as they are when a call comes from Gizmo. The second interesting features is that you can configure gizmo project to work with other SIP-Based services such Free World Dialup and others. This means you can now log into your other SIP-based service directly from gizmo project, making it work like a universal softphone such as eyebeam. Another good news about this latest version is that it will work on Nokia 770 Tablet as well. More info about gizmo project 2.0 is available at http://www.gizmoproject.com/asterisk.html
Labels: VoIP News
Tuesday, May 23, 2006
JaJah Firefox Extension
JaJah has just released a jajah extension for firefox browser. This gives a true click to call capability to your browser. Phone numbers in websites are automatically detected and highlighted thereby ready for click to call. Clicking on such highlighted numbers cause Jajah to initiate a phone call from your landline or mobile phone to the desired destination. Numbers can also be entered directly into the toolbar and at the click of enter jajah initiates the call. I have used it and it worked really fine. The latest feature has even made a very easy to use web activated telephony sytem even easier. Check out https://addons.mozilla.org/firefox/2577/ to download and install this wonderfull Jajah firefox extension.
Monday, May 22, 2006
Skype Outlook Integration
Microsoft Outlook has been there for a very long time and has become the primary e-mail client for many people and businesses. Most if not all the contacts for most people and businesses are stored in outlook. Skype also has become so popular as a means of making free internet calls with superb audio quality. With skypeout we can also call regular phone numbers anywhere in the world at very low rates. Lots of individuals and businesses are now using skype to reduce the cost of there long distance calls or to enhance their business. Integration of Skype and outlook gives an all in all application. Skype or Skypeout calls can be made to an outlook contact directly. No need to transfer your outlook contacts to skype and vice versa. With this post i will review several applications out there that enables us to integrate skype and outlook and also give neccessary links to obtain the applications.
Skylook
This is a very great application that intelligent integrates Skype with outlook. I have been using skylook for some time now and it has really made my communication more efficient. Skylook has transformed my outlook into a complete communication tool by efficiently integrating it with skype. With skylook you can do the following:
- Make Skype and Skypeout calls to your outlook contacts and answer incoming calls directly from outlook
- Skylook answering machine answers incoming calls automatically and record audio messages directly to your outlook inbox, thereby saving you the cost of subscribing to skype voicemail service.
- Record voice calls into MP3 and auto archive them to outlook folders.
- Send instant messages to outlook contacts. Skylook put unanswered instant messages in to outlook inbox and you manage them as you manage unanswered e-mails.
- Make Skype and Skypeout calls to your outlook contacts
- Send Instant Message to your contacts
- Access all your skype contact directly from the drop-down menu in the skype toolbar in outlook
- Instanly see presence information of your contacts
- Email respondents in your Skype buddy list directly from outlook.
- Make free Skype to skype calls from your outlook contact folder.
- Make cheap Skypeout calls to any landline or mobile phones of your outlook contacts.
- Start a chat session with a contact
- Keep track of the calls you made.
Labels: VoIP News
Saturday, May 20, 2006
VoIP to steal $100B in PSTN revenues
According to a U.K research house Informa Telecoms & Media, Over the next five years VoIP usage will drain about $100 billion from carriers PSTN revenues. It is also believed that by 2010 PSTN revenue will no longer be the main revenue source for the carriers. Informa predicts that by 2011 the carriers revenue from PSTN will fall by as much as 16.7 per cent. The good news for the carriers is that their revenue source from VoIP will also increase considerably because they have made huge investement in broadband. But carrier's revenue from VoIP is expected to be much lower than that from PSTN because voip services are generally priced lower and due to competition from dedicated VoIP services like Skype and vonage.
Labels: VoIP News
Thursday, May 18, 2006
T-Mobile UK Bans Use of VoIP and IM Over Its Web 'n' Walk service
T-Mobile UK has ban the use of VoIP and IM over its Web 'n' Walk service. The Web 'n' Walk service enables users to have broadband internet connection over T-mobile network. The clause in the contract says "Use of Voice over Internet Protocol and Messaging over Internet Protocol is prohibited by T-Mobile. If use of either or both of these services is detected T-Mobile may terminate all contracts with the customer and disconnect any SIM cards and/or web ‘n’ walk cards from the T-Mobile network." This is a move that i believe will backfire as many people wants to use such service because it enables them to save on high cost of mobile communication especially in europe. T-mobile said they are aiming business network and claimed that VoIP and IM is not key to their customers but its really difficult to conceptualize how low-cost voice and instant messaging would be unimportant to businesses. So if you are interested in signing up for a mobile internet access service and would like to use VoIP and IM you should beware of such clauses in the contract.
VoXaLot Launches Web Activated Telephony Service
VoXaLot has launched a jajah like web activated telephony services. This service enables you to enjoy the cost sayings of voip without the need for ip phones, adapters or a broadband internet connection. Users of this service can initiate calls between two phones anywhere around the world. The phones at either end can be landline, mobile or voip phone. This service also have an added advantage that users can choose their own provider for the calls thereby controlling the cost of the calls. Its also possible to use different providers for each leg of the call. After signing up for the service all you have to do to make a call is to enter your number and the number you want to call. Enter the providers you want to use for the calls and click call and your call will be completed in seconds. For more information visit www.voxalot.com.
Phonegnome interoperates with Skype
TelEvolution has just released an extension to phonegnome nicknamed GnomeLink which enables making and receiving skype calls using the phonegnome box. Phonegnome previously enable users to call any voip and pstn numbers but now users can use their phonegnome to call skype contacts and skypeIn numbers.
The PhoneGnome VoIP Box from TelEvolution
The PhoneGnome voip service from TelEvolution is a very interesting solution that cleverly blend your existing PSTN with voip allowing you to enjoy advantages of voip service while still keeping your regular phone line for services such as E911 etc. The phonegnome is an easy to install and use system. It comes in form of a box that plugs into your existing phone line and your broadband internet connection (DSL modem or Cable modem). After properly setting up phonegnome which normally takes some minutes because its plug and play you can start making voip calls using your normal telephony. No need to change the way you dail numbers, just dial as you use to dial before and your calls will be routed through the internet at very low rates and very good audio quality. Calls to other phonegnome enabled phones are free no matter how long you call. Also phonegnome enables you to call other voip phones such as FWD, gizmo etc for free. When your internet connection is down calls can be routed through you regular phone line.The figure below (source phonegnome.com) shows how phonegnome works.
Phonegnome also come with interesting phone features such as Seven-digit Dialing,Emergency 911,Three-way Calling,Speed Dial,Call Forwarding,Call Transfer,Telemarketer Screening and internet features such as Online management of Settings and Features,Online Contact Management,White Pages,Click-to-Dial,Voicemail to E-mail,Find-Me/Follow-Me,Online Call Logs. Phonegnome can be configured for any SIP based voip service that meet your needs. Phonegnome has a price tag of $119 and can be ordered online. For more information about phonegnome check out www.phonegnome.com

Phonegnome also come with interesting phone features such as Seven-digit Dialing,Emergency 911,Three-way Calling,Speed Dial,Call Forwarding,Call Transfer,Telemarketer Screening and internet features such as Online management of Settings and Features,Online Contact Management,White Pages,Click-to-Dial,Voicemail to E-mail,Find-Me/Follow-Me,Online Call Logs. Phonegnome can be configured for any SIP based voip service that meet your needs. Phonegnome has a price tag of $119 and can be ordered online. For more information about phonegnome check out www.phonegnome.com
Skype Announces Free calling for US and Canada until 2007
In other to improve the use of Skype in North America, Skype has announce that calls to landlines and mobiles in US and Canada will be free till year 2007. Before now one need to buy skypeout credits to be able to call regular phones in US, Canada and other parts of the world. This free calls is only available for calls originating from US and Canada so skype users in other part of the world still require skypeout credits to call pstn numbers in US and Canada. Skype has greater adoption in Europe and voip subscribers US and Canada tend to favour ISP branded services. So skype believe this move will grow US and Canadian usage of the application and eventually create more paid clients.
Tuesday, May 16, 2006
Introducing VoIP eBook , "Internet Telephony Secrets"
The voip ebook tittled Internet Telephony Secrects written by Gobala Krishnan of Liberty Straits Solutions is a very good ebook for businesses wishing to use skype and voip in general as a marketing tool. It shows in easy steps how businesses can use the power of skype and voip technology not only to reduce the cost of their long distance and international calls but to use it as a tool to grow their business and expand to markets that otherwise could have been impossible to reach. I have read this eBook and i really recommend it. Click Here for more details about Internet Telephony Secrects.
What are Codecs - Speex
Speex is a variable bitrate codec, which means that it si able to dynamically modify its bitrate to respond to changing network conditions. It is offered in both narrowband and wideband versions, depending on whether you want telephone quality or better. Speex is a totally free codec, licensed under the Xiph.org variant of the BSD license. Speex can operate at anywhere from 2.15 to 22.4 kbps due to its variable bitrate.
Related post: What are Codecs?
What are Codecs - The iLBC
Written in full iLBC means Internet Low Bitrate Codec. iLBC provides an attractive mix of low bandwith usage and quality, and it is especiallly well suited to sustaining reasonable quality on lossy network links. It is not as popular as the ITU codecs and thus may not be compatible with common IP telephones and commercial voip systems. Because iLBC uses complex algorithms to achieve its high levels of compression, it has a fairly high CPU cost. While you are allowed to use iLBC without paying royalty fees, the holder of the patent, Global IP Sound (GIPS), wants to know whenever you use it in a commercial application. The way you do that is by downloading and printing a copy of the iLBC license, signing it, and returning it to them. iLBC operates at 13.3 kbps (30-ms frames) and 15.2 kbps (20-ms frames).
Related post: What are Codecs?
Why Fax-over-IP? Why not email?
Labels: Fax-over-IP
What are Codecs - GSM
The GSM codec does not come encumbered with a licensing requirement the way that G.723.1 and G. 729A do, and it offers outstanding performance with respect to the demand it places on the CPU. The sound quality is generally considered to be of a lesser grade than that produced by G.729A, but as much of this comes down to personal opinion. You should try out the codec in your voip applications.
Related post: What are Codecs?
What are codecs - The G.729A
In this post i will describe the features and relevance of the G.729A codec. Considering its low bandwith requirement, the G.729A delivers impressive sound quality. It does this through the use of Conjugate-Structure Algebraic-Code-Excited Linear Prediction (CS-ACELP). Because of patents, you can't use G729A without paying a licensing fee; however, it is extremely popular and is thus well supported on many different phones and systems. To achieve its impressive compression ratio, this codec requires an equally impressive amount of effort from the CPU. G.729A uses only 8 kbps of bandwith.
Related post: What are Codecs?
What are codecs - The G.723.1
This post is for the purpose of describing the features of the G.723.1 codec. This codec is designed for low-bitrate speech. It has two data bitrate settings: 5.3 kbps and 6.3 kbps. G.723.1 is one of the codecs required for compliance with the H.323 protocol ( although other codecs may be employed by H.323). It is currently encumbered by patents and thus requires licensing if used in commercial applications. What this means is that while you can switch two G.723.1 calls through an Asterisk system for example, you are not allowed to decode them without a license.
Related post: What are Codecs?
Saturday, May 13, 2006
UPYLE4 - Express USB Phone
This USB Phone connect to your PC through a USB port and enable you to make both PC-to-PC and PC-to-phone directly from it using a SIP based voip software of your choice. It was developed by Australia based NCH Swift Sound and it has the following features, Commercial grade high quality speakerphone, Large LCD display with backlight, Selectable ring style and volume for incoming calls, Caller ID display,Echo cancellation, noise reduction, full duplex communication, PC-to-PC, PC-to-phone, Phone to Phone operation, Comes with free Express Talk VoIp Phone Software but works with other SIP software, Free VoIP to VoIP calls and much cheaper calls to ordinary phone if your telco has a as Sip VoIP Gateway Service. The USB phone can be used with voipbuster, voipcheap and other Finarea voip applications with very good audio quality. It require no external power and has built in driver and Sound card. System requirement for this phone are: Windows 2000, XP or later,400 MHz processor or higher,128 MB RAM or higher, Free USB port,Internet Connection (33.6 Kbps modem, or any broadband: cable, ADSL.)Labels: Free International Calls
InteleFone Will Introduce E-911 Emergency Service and Virtual Numbers!
The subscription based voip service provider InteleFone will begin New E-911 Emergency Services and a selection of Area Codes for InteleFone Virtual Phone Numbers. This will be part of their May-2006 Phase III Roll Out. They believe that adding these features to their best selection of calling plans with New Extensive Calling Features will give InteleFone Distributors the Best Possible Suite of Products, Features and Calling Plans to Market to the World. The details of these new features will be discussed at monday night conference call of InteleFone staffs and worldwide distributors on 15th May 2006 and a public announcement will be made shortly after. For more information about InteleFone visit www.intelefone.com.
Labels: Conference Calls, VoIP News
More "Goodies" coming from JaJah
Jajah will soon be adding lots of cool features to their highly popular web activated telephony. These new features include but not limited to conference calls, text messaging and scheduled calls. Just imagine having access to all this wonderful features without even having a broadband internet connection. Click here to register with JaJah now and enjoy this cool features soon.
Labels: VoIP News
Friday, May 12, 2006
What are codecs - The G.726
This codec has been around for some time (it used to be G.721, which is now obsolete), and it is one of the original compressed codecs. It is also known as Adaptive Differential Pulse-Code Modulation (ADPCM), and it can run at several bitrates. The most common rates are 16 kbps, 24 kbps, and 32 kbps. G.726 offers quality nearly identical to G.711, but it uses only half the bandwidth. This is possible because rather than sending the result of the quantization measurement, it sends only enough information to describe the difference between the current sample and the previous one. G.726 fell from favor in the 1990s due to its inability to carry modem and fax signals, but because of its bandwith/CPU performance ratio it is now making a comeback. G.726 is especially attractive because it does not require a lot of computational work from the system.
Labels: Fax-over-IP
What are Codecs - The G.711
This is the second post of the "What are Codecs series" and it describes the G.711 codec. G.711 is the fundamental codec of the PSTN. In fact, if someone refers to PCM with respect to a telephone network, you are allowed to think of G.711. Two companding methods are used: micro-law in North America and A-law in the rest of the world. Either one delivers an 8-bit word transmitted 8,000 times per second. If you do the math, you will see that this requires 64,000 bits to be transmitted per second. Many people will tell you that G.711 is an uncompressed codec. This is not exactly true, as companding is considered a form of compression. What is true is that G.711 is the codec from which all others are derived.
What are Codecs ?
I am writing this piece in response to mails from some of my readers who wrote in to ask for explanation about the significance and effect of the different codec options they have in their various voip applications. This post is to give the definition of codecs, explain the different codecs in use in the voip ecosystem and explain how choosing one or the other could affect the performance of your voip application. Codecs are the various mathematical models used to digitally encode and compress analog audio information. Whenever you talk into your microphone while using your favourite voip application to call someone, the analog audio signal generated by your microphone can not be sent over the internet directly but has to be digitised and compressed before it can be sent over the internet. The mathematical models the enable the coding and compression of such analog audio signal is what we refer to as codecs. Originally, the term CODEC referred ti a COder/DECoder: a device that converts between analog and digital. Now, the term seems to relate more to COmpression/DECompression.One of the important features of a codec is its bitrate which refers to the number of bits that is required to be transmitted per second. The higher the bitrate the higher is the bandwith requirement, so the purpose of the various codecs is to strike a balance between efficiency and quality. The table below gives the various codecs that are used in voip applications and their bitrate.
In the following posts i will describe each codec in detail and the table above will be link to those posts so that by clicking on any codec you can read about all the features of that codec. This is just the first post in the "what are Codecs" series and by the end of the series you will be have a proper understanding of what codecs are, their features and which one is the best for you to select in your voip application for best performance.
Codec Bitrate (kbps) G.711 64 G.726 16, 24, or 32 G.723.1 5.3 or 6.3 G.729A 8 GSM 13 iLBC 13.3 Speex Variable
In the following posts i will describe each codec in detail and the table above will be link to those posts so that by clicking on any codec you can read about all the features of that codec. This is just the first post in the "what are Codecs" series and by the end of the series you will be have a proper understanding of what codecs are, their features and which one is the best for you to select in your voip application for best performance.
Thursday, May 11, 2006
All Finarea voip clients are now with only 20 hours free per month
All voip clients from Finarea i.e voipbuster, voipcheap, voipstunt, voipdiscount, poivy etc are now only limited to 20 hours of free calls per month. The days of unlimited free calls seems to be over but this clients still have cheapest rates in the market.
Labels: Free International Calls
Voipbuster new superdeal call rates
Voipbuster has introduced lots of new superdeal call rates for several destinations drastically reducing the cost of calling several countries. Below are the break down of the new superdeal destinations:
4 cents per min destinations
Aruba, Bahrain, Benin mobile,Botswana, El salvador, Guatemala, Jordan, Kenya, Kuwait, Mali, Mayotte, Namibia, Niger, NL antilles, Togo and Uganda 3 cents per min destinations
Albania, Iran, Malawi, Reunion 2 cents per min destinations
Burundi, Dominican Rep., Gabon, Iraq, Kyrgyzstan, Laos, Paraguay, Serbia and Montenegro, Uruguay, Zimbabwe.
Aruba, Bahrain, Benin mobile,Botswana, El salvador, Guatemala, Jordan, Kenya, Kuwait, Mali, Mayotte, Namibia, Niger, NL antilles, Togo and Uganda 3 cents per min destinations
Albania, Iran, Malawi, Reunion 2 cents per min destinations
Burundi, Dominican Rep., Gabon, Iraq, Kyrgyzstan, Laos, Paraguay, Serbia and Montenegro, Uruguay, Zimbabwe.
Labels: VoIP News
Certified Voipbuster Hardware
Voipbuster now has a certified hardware webpage. This webpage has a list of all certified hardwares that can be used with voipbuster for best performance. The is to make it easier for voipbuster users to choose the best available hardware to use with voipbuster. Listed are Computer headsets,USB phones,USB wireless phones,SIP phones,Wi-Fi phones,ATA,SIP ATA,Routers WLAN,ADSL router and ADSL router WLAN. So if you are looking for hardware to use with voipbuster check out the certified voipbuster hardware webpage.
Wednesday, May 10, 2006
Connect Skype to your PBX with Skype2PBX
For the small and medium enterprise who do not want to completely migrate to IP-PBX but still want to enjoy the cost savings of voip technology Skype2PBX is one of the ways to go. Skype2PBX is one of the series of skype API based products coming into the market on daily basis nowadays. Skype2PBX is developed by Italy based Micronet S.r.l and it enable an enterprise to connects its PBX system to Skype thereby making it possible to make and receive skype calls on the PBX extensions. Skype2PBX is a self installing software and it enables you to manage all skype services with a traditional telephone. To be able to use Skype2PBX you require a PC with Linux operating system on which skype2PBX will be installed and a USB phone adaptor for each skype account you wish to connect to the PBX. The phone adaptor connects to the PC through a USB port and to the PBX with RJ-11 connector.The figure below (source www.skype2pbx.com) shows a typical setup for skype2pbx:
The minimum system requirement includes Intel Pentium III 800 MHz, 256 MB RAM, 8 GB Hard drive, USB 1.1 ports. If properly installed the Skype2PBX works fine and all the skype accounts can be managed from within the software. Up to 99 skype lines can be connected to your PBX but the number of lines connected to Skype2PBX will affect the overall computer performance.I am working at the moment on connecting skype to my asterisk PBX using Skype2PBX. For more information and download visit www.skype2pbx.com.
The minimum system requirement includes Intel Pentium III 800 MHz, 256 MB RAM, 8 GB Hard drive, USB 1.1 ports. If properly installed the Skype2PBX works fine and all the skype accounts can be managed from within the software. Up to 99 skype lines can be connected to your PBX but the number of lines connected to Skype2PBX will affect the overall computer performance.I am working at the moment on connecting skype to my asterisk PBX using Skype2PBX. For more information and download visit www.skype2pbx.com.
100 free JaJah Minutes for your Mom - JaJah's Mothers day gift.
JaJah is giving away 100 free Jajah minutes to mother's of Jajah users on mother's day. If you are a jajah user all you have to do is to register your Mom (or grandmother) with Jajah and send her e-mail address to mothersday[at]jajah.com, and she will get 100 free minutes with JaJah to call you. Jajah is a fast growing voip calling and their click to call internet activated service is very easy to use by everybody including our mothers and grandmothers.“All of us at JAJAH love Mothers,” said JAJAH co-founder Roman Scharf. “And since JaJah is just like making a typical phone-to-phone call, everyone’s Mom can now call their children anytime they want and save a ton of money.”Visit JaJah and give your Mom a Jajah's mother's day present now.
Sunday, May 07, 2006
FINAREA launched another voipcheap with more than 50 free destinations.
Finarea, the company behing voipbuster and other voip clients seems to be short of names and has decided to call there latest voip client again voipcheap. The differences between this latest voipcheap and the old one are, the domain name for the new voipcheap is voipcheap.com as against voipcheap.co.uk for the old one, rates are in euros and not pounds, the GUI design is slightly different and there are more free destinations. However free calls is limited to 20 hours per month in the new voipcheap. Allthough the voice quality of Finarea's voip applications are really good and their call rates are still the lowest in the market, but the way they keep changing their rules and business model couple with the fact they have no custormer support is annoying many subscribers. Not a few believe that the behaviour of Finarea is far from acceptable business ethics.
Labels: Free International Calls, VoIP News
Skype now with 100-person conferencing feature
Skype has release a "early preview" of a new skype feature which allows up to 100 person to talk in a conference call. Nicknamed Skypecast, it allows users to participate in conversation with up to 100 other skype subscribers and they will be moderated by a host with the authority to manage who speaks when and for how long. Before now skype only allow conferences with up to 10 participants. Skypecast will be listed publicly on skype website for easy access. I believe skypecast has the potencial to change the way people communicate over the web and to become a popular venues for people with similar interest and skills to meet each other and network.
Labels: Conference Calls, VoIP News
Tuesday, May 02, 2006
PowerDsine Announces Next-Generation Power over Ethernet Midspan Series
PowerDsube announced the launch of its next generation 6500 PoE Midspan series. The new IEEE 802.3af-compliant 6500 PoE Midspan series is reportedly the first PoE midspan to offer advanced secured network management system and the only PoE Midspan to ship with a lifetime warranty. The PowerDsine 6500 PoE Midspans will be available in 6-, 12-, 24- and 48-port versions. The 6500 Midspans are designed for enterprises, which are deploying IP phones, wireless LAN access points or IP surveillance cameras. MIdspans allow networks the added functionality of PoE without the need to replace existing Ethernet switches. This optimizes PoE port count, improves the ROI of networks and saves costs associated with these deployments. "The 6500 PoE Midspan Family provides enterprises a new level of security with this advanced remote management solution," said Igal Rotem, PowerDsine's CEO. "Moreover, we have such confidence in the 6500 Midspans' reliability that we are offering customers a lifetime warranty."
Labels: VoIP News
Monday, May 01, 2006
Freshtel firefly softphone gives really good audio quality and advanced voicemail features
I have been testing firefly softphone from Australia based freshtel for some time and i am really impressed with the crystal clear voice quality. The softphone GUI is simple but beautifully designed and its very easy to install and use. To use freshtel voip service you need to sign up for an account on their website, download and install the firefly softphone with your account details. Installing firefly takes just few minutes with no tedious configuration required. Both PC-to-PC and PC-to-phone calls are available and pc-to-pc calls between firefly users is free. PC-to-phone calls are at low rates (rates are given in Australian dollars) though rates are a bit higher than some other services i have written about on this blog before but the voice quality and the fact that calls are charged per second compesates for that. Firefly also has instant messaging capability and advanced voicemail system which is completely free. You can record both unavailable and busy messages to be played back when somebody call you when you are not available or busy. Your voicemail is stored on their server and password protected so that only you can listen to it whenever you want. You are informed through a text message whenever you have a voicemail. Freshtel also have attractive subscription based service with cost ranging from $5.95 dollars per month to $29.95 dollars per month. These subscription based services are targeted at residents of Australia and includes call credits, flat rates calling Australia wide, free Standard Australian phone number etc. You can visit Freshtel website to download firefly and to try their this amazing voip service.
VXI Parrott TalkPro TP100 USB Headset - The VoIP Headset
The TalkPro TP100 USB Headset is designed specifically for voip applications. It has an integrated Digital Signal Processor (DSP) which convertes the analog audio signal from the microphone to digital signal which is more suited for the computer environment thereby eliminating the problem of voice echo within voip applications. The TalkPro TP100 USB Headset has a superior, voice band optimized, noise-canceling microphone. It connects to the computer through USB interface removing problems of PC system sound interference and hardware incompatibility by effectively bypassing the sound card. This features are really great for voip applications and will surely improve your call experience. Its compatible with Windows 98SE, ME, 2000, XP, and Mac OS 9.2 and higher.

