Voip Technology, Info for those admire Voip Engineer

It is very easy to get into a discussion that is very technical and confusing to most readers. The purpose of this section will be to provide a very high-level overview of Voice over IP (▲VoIP) aimed at those who do not consider themselves experts in the subject and hopefully with enough clarity that it serves as a good introduction to most readers.

Many people have used a computer and a microphone to record a human voice or other sounds. The process involves sampling the sound that is heard by the computer at a very high rate (at least 8,000 times per second or more) and storing those "samples" in memory or in a file on the computer. Each sample of sound is just a very tiny bit of the person's voice or other sound recorded by the computer. The computer has the wherewithal to take all of those samples and play them, so that the listener can hear what was recorded.

VoIP is based on the same idea, but the difference is that the audio samples are not stored locally. Instead, they are sent over the IP network to another computer and played there.

Of course, there is much more required in order to make VoIP work. When recording the sound samples, the computer might compress those sounds so that they require less space and will certainly record only a limited frequency range. There are a number of ways to compress audio, the algorithm for which is referred to as a "compressor/de-compressor", or simply ▲CODEC. Many CODECs exist for a variety of applications (e.g., movies and sound recordings) and, for VoIP, the CODECs are optimized for compressing voice, which significantly reduce the bandwidth used compared to an uncompressed audio stream. Speech CODECs are optimized to improve spoken words at the expense of sounds outside the frequency range of human speech. Recorded music and other sounds do not generally sound very good when passed through a speech CODEC, but that is perfectly OK for the task at hand.

Once the sound is recorded by the computer and compressed into very small samples, the samples are collected together into larger chunks and placed into data packets for transmission over the IP network. This process is referred to packetization. Generally, a single IP packet will contain 10 or more milliseconds of audio, with 20 or 30 milliseconds being most common.

Vint Cerf, who is often called the Father of the Internet, once explained packets in a way that is very easy to understand. Paraphrasing his description, he suggested to think of a packet as a postcards sent via postal mail. A postcard contains just a limited amount of information. To deliver a very long message, one must send a lot of postcards. Of course, the post office might lose one or more postcards. One also has to assemble the received postcards in order, so some kind of mechanism must be used to properly order to postcards, such as placing a sequence number on the bottom right corner. One can think of data packets in an IP network as postcards.

Just like postcards sent via the postal system, some IP data packets get lost and the CODECs must compensate for lost packets by "filling in the gaps" with audio that is acceptable to the human ear. This process is referred to as ▲packet-loss concealment (PLC). In some cases, packets are sent multiple times in order to overcome packet loss. This method is called, appropriately enough, redundancy. Another method to address packet loss, known as forward-error correction (FEC), is to include some information from previously transmitted packets in subsequent packets. By performing mathematical operations in a particular FEC scheme, it is possible to reconstruct a lost packet from information bits in neighboring packets.

Packets are also sometimes delayed, just as with the postcards sent through the post office. This is particularly problematic for VoIP systems, as delays in delivering a voice packet means the information is too old to play. Such old packets are simply discarded, just as if the packet was never received. This is acceptable, as the same PLC algorithms can smooth the audio to provide good audio quality.

Computers generally measure the packet delay and expect the delay to remain relatively constant, though delay can increase and decrease during the course of a conversation. Variation in delay (called jitter) is the most frustrating for IP devices. Delay, itself, just means it takes longer for the recorded voice spoken by the first person to be heard by the user on the far end. In general, good networks have an end-to-end delay of less than 100ms, though delay up to 400ms is considered acceptable (especially when using satellite systems). Jitter can result in choppy voice or temporary glitches, so VoIP devices must implement jitter buffer algorithms to compensate for jitter. Essentially, this means that a certain number of packets are queued before play-out and the queue length may be increased or decreased over time to reduce the number of discarded, late-arriving packets or to reduce "mouth to ear" delay. Such "adaptive jitter buffer" schemes are also used by CD recorders and other types of devices that deal with variable delay.

Video works in much the same way as voice. Video information received through a camera is broken into small pieces, compressed with a CODEC, placed into small packets, and transmitted over the IP network. This is one reason why VoIP is promising as a new technology: adding video or other media is relatively simple. Of course, there are certain issues that must be considered that are unique to video (e.g., frame refresh and much higher bandwidth requirements), but the basic principles of VoIP equally apply to ▲video telephony.

Of course there is much more to VoIP than just sending the audio/video packets over the Internet. There must also be an agreed protocol for how computers find each other and how information is exchanged in order to allow packets to ultimately flow between the communicating devices. There must also be an agreed format (called payload format) for the contents of the media packets. We will describe some of the popular VoIP protocols in the next section.

Through this section, we have focused on computers that communicate with each other. However, VoIP is certainly not limited to desktop computers. VoIP is implemented in a variety of hardware devices, including IP phones, ▲analog terminal adapters (ATAs), and ▲gateways. In short, a large number of devices can enable VoIP communication, some of which allow one to use traditional telephone devices to interface with the IP networks: one does not have to throw out existing equipment to migrate to VoIP.

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Free Call To USA

New today [08 March 2009]

15. www.voipdiscount.com

16. www.voipraider.com

17. www.voipstunt.com

18. www.voipwise.com

19. www.voipzoom.com

20. www.webcalldirect.com

Edited 07 March 2009.

More new website that provide free call to USA

6. www.nonoh.net

7. www.poivy.com

8. www.sipdiscount.com

9. www.smartvoip.com

10. www.smslisto.com

11. www.voipbuster.com

12. www.voipbusterpro.com

13. www.voipcheap.co.uk

14. www.voipcheap.com

That all for today updated.. Stay tune...


Here are suggestion of web that provide FREE CALL TO USA

1. www.12voip.com

2. www.freecalls.com

3. www.justvoip.com

4. www.lowratevop.com

5. www.netapel.com

All this voip provide free call to mobile and fixed line to USA using softphone in computer with T & C apply.

You can buy this voucher from me. IT cost you only RM 60 per euro 10. It may reduce to RM 57 for bulk.

This voip all give good quality voice over IP.

Suitable for :-
1. Student whose study oversea.

2. Businessmen.

3. Those working OS.

4. Have alot of friend in OS

5. Want to reduce calling cost.

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Peer-to-peer, Virtual PBX, hosted, managed business VOIP

So yu hae heard talk about VoIP, but wat exactly does it mean? Well, the answer is that VoIP means different things to different people; it all depends on who you ask. Since the term VoIP can be applied in many unique ways, there is usually another word in front of VoIP to accurately explain the service. Usually the term VoIP is applied in one of the following ways : peer-to-peer VoIP, Virtual PBX VoIP, hosted VoIP, managed business VoIP etc. Below is a short run down of these services providing you with a quick look at each niche market to help you better understand this service.

Peer-to-Peer VoIP

What it is : Two people who have downloaded the same software can chat directly for free. Example : Skype, YM

What you should know : It is free but works best for individuals ( when you want to chat with your kids or spouse from a business trip, for instance ), not business.

Residential VoIP. In general this means that you use your analog phone to chat with other people except in this case the call is going over the Internet instead of a phone line. AN example of this is Vonage. An important thing to keep in mind is you get what you pay for. The price is better than a regular phone but service is often spotty.

Virtual PBX. The benefit of virtual PBX is that it makes a small company look bigger. If you have a one-person firm, this is a good ption. By setting up a virtual PBX you can have callers press one for sales, two for marketing and three for technlogy. Then you can have all calls routed to your cell phone. An example of virtual PBX is GotVMail. Keep in mind however that while this is expensive, its functionality is limited.

Hosted. This simply means that there are phones but no central piece of equipment at your office. This central technology is hosted by the provider. Example : Packet 8

What you should know : You get more functionality than with a virtual PBX, yet you don't have to spend as much as you would on a managed business VoIP.

Managed Business.

This service is nearly identical to hosted VoIP, except everything runs on private lines, instead of the public Internet. This is ideal for a very large company that will be in truble if its phone s system goes down, even on a very rare occasion and even if only for a few minutes. Examples : MS< CBeyond, Cisco.

What you should know : Better quality control but most expensive of all VoIP options.

About the Author :
Frank Newman is a VoIP expert, providing in depth analysis of IP phones, VoIP providers, and more on his industry leading blog.

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Most Cheap Voip In World

Definitely i suggest all user voip use Betamax product. There provide the most cheap and the best quality voip i have experience.

You don't need to maintain monthly payment - no monthly fees.

You don't need to hurry up finish ur credit - credit not expired.

You can use phone to phone - does't need to use your mic or headphone..

You also can use call back - some country especially europe.

You also can sms using Betamax product.

You also can save a lot of money. Compare to the PSTN cost.. It will charge you Euro 1 / minute to call Middle East from Malaysia.. BUt using VOIP BETAMAX, it onlt cost you Euro 0.05 - euro 0.10 .. That is more than 20 times save than using PSTN... This voip technology can help you in reducing your calling cost because the network is already there..

I hope this voip is more advanced in year 2009. Maybe we can use video conference or 3G technology combain with voip or the rates is more cheap... or maybe we can send mms via voip..

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Talk Becomes Cheap - VoIP The Alternative

More than for million people worldwide now use VOIP to route heir calls over the internet and private networks instead of over dedicated phone lnes. All you need is an adaptor that sits between your phone and your high-speed connection and magically transforms your conversation into byte-size digital packets and back to voice again. The quality is better than your cell and often as good as your landline, with a gazallion include features, for less money than you pay now. You can even keep your phone and number. So why aren't we all using VoIP?

For starters, you need broadband. Also, local area codes aren't yet available everywhere. Other hurdless : VoIP lines die when the power does, not all provider support 91, and there are theoritical security risks ( call taping, voicemail spam ), although most providers have protection against these threats

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VoIP Gains Traction in Slower Economy

As if we needed any more proof, rumors swirling this week of massive layoffs at Microsoft Corp. show us that no company is immune from the recession that’s taken hold.

Yet, just as mobility is emerging as an attractive option in this economy, with its inherent cost-savings, so VoIP, with its lower costs and increasingly reliable and high quality, is poised to gain a healthy share of the voice communications space.

Experts already have told us that VoIP will gain a greater appreciation from the federal agency that regulates communications in the United States under President-elect Barack Obama’s administration. They’ve also said that VoIP stands to get better carrier interconnection rights and recognition among policy-makers that the telecommunications world is evolving toward Internet telephony.

As Sam Li, chief executive officer and co-founder of Clearsight Networks says, demand for VoIP test solutions remains very healthy. Li’s Fremont, California-based company provides network monitoring and analysis tools for real-time application troubleshooting.

Interestingly, in our interview with Li, printed below, we learn that for many of Clearsight’s customers, unified communications haven’t yet reached a point where solutions are compelling enough to draw clients.

Our exchange follows.

TMCnet: As it has for many industries, the recession has thrown a wrench into plans for companies in the telecommunications space. At the same time, VoIP is emerging as an increasingly attractive way for many businesses navigating the slower economy to cut costs. How has demand for ClearSight Networks’ VoIP analysis tools changed since the slowdown, if at all, and what does the company anticipate for 2009?

Sam Li (pictured left): While it is true that today’s economic situation is hitting the telecommunications space hard, there is a silver lining. VoIP’s big draw is, and always has been, that it is more cost-effective than traditional telephony. As a result, there’s good reason to believe that the VoIP industry may not be as impacted as other businesses, as companies looking to save money continue to dedicate more attention to VoIP solutions than traditional telecom services. To ensure the ongoing health of their network and these VoIP services, these organizations require management tools such as the kind ClearSight provides.

ClearSight has received quite a few awards for our VoIP test solutions. These awards help us position ClearSight as a leader in this space and help boost customer confidence when selecting one of our solutions.

Demand for our tools is still relatively healthy – all things considered. We continue to push our products into new markets and verticals. For example, ClearSight is currently working with one of largest telecom organizations in Japan for the third phase of a large triple play project; the end-users are residential and home customers.

As far as the outlook for 2009, ClearSight is on track to grow our business and make our revenue targets, despite the challenging economy.


TMCnet: Though the market hype for some time has been around “unified communications,” some would say that companies must get VoIP right first – that is, of a high and consistent quality – before they consider moving into a unified messaging environment. What do you hear from customers who are pursuing UC options such as fixed and mobile voice, e-mail, instant messaging, desktop and advanced business applications, IP-PBX (News - Alert), VoIP, telepresence, voicemail, fax, audio video and Web conferencing?

SL: UC has not reached a critical point yet with our customers and I think this is mainly due to the lack of compelling solutions. ClearSight is closely researching the market, especially for the UC infrastructure solutions such as Cisco UC Manager orMicrosoft ( News - Alert) Office Communication Server.

Ultimately, however, UC requires a higher quality network experience that any organization looking to invest in the technologies needs to closely consider. ClearSight Networks’ quality of service monitoring and trouble-shooting functions are major differentiators to any competitive offering, and as demand for UC increases, we anticipate so will demand for our solutions to properly monitor and manage them. Currently, ClearSight offers sophisticated and in-depth application analysis features for managing VoIP, video, fax and Web applications. As you mentioned, most of our customers would like to get VoIP deployed and working first, and so that is were we are focusing much of our efforts.

TMCnet: How often do the problems that ClearSight sees emerge in a VoIP system – whether it’s packet loss or something else – have to do with the IP network itself?

SL: When it comes to system problems, I would say it is 50/50 for the VoIP deployment. Fifty percent of the problems in a VoIP system are caused by the network, which is perhaps not as well-designed and engineered as it should be. Some companies are still running the applications other than VoIP in their VoIP backbone. Such practices should be discouraged as packet loss, jitter and delay are the major culprit to good VoIP quality and running such applications can only detrimental to achieving good VoIP. Lastly, VoIP equipment and applications are also major culprits.

TMCnet: Your company’s ClearSight Analyzer and ClearSight Distributed products monitor VoIP transmission through sets of audio and video statistics and reports. TMCnet has talked to companies that offer similar services in the past, such asPsytechnics ( News - Alert), which provides software solutions that improve video communications and the quality of IP telephony. In a market that’s expected to become even more crowded, what specific value proposition does ClearSight bring?

SL: Since VoIP is becoming increasingly important, there will be more and more companies entering the market and the total market size will grow significantly for the next few years. ClearSight enjoys several clear advantages in the area of VoIP and video monitoring and management. As a pioneer in application monitoring, analysis and reporting, we not only have years of experience strengthening our solutions, but we are also committed to continuously enhancing our products and solutions so that we may provide innovative and cutting-edge tools to our loyal customers.

There are a handful of unique, key innovations at work within ClearSight solutions that gives us advantages over the competition. For example, in the area of unified communications, ClearSight solutions are adept at VoIP, video, fax and Web applications, ensuring organizations can proactively protect the health of their cutting-edge networks. ClearSight has also built up intuitive VoIP application level analysis features on top of ClearSight Network Time Machine product line, which will continue to be a focus of ours for the next three years.

ClearSight delivers the only solutions that can accurately identify problems at the application layer. This enables ClearSight to more effectively capture, report and store data for real-time, proactive monitoring and management of your most critical applications as well as new, emerging technologies—better than any competitive solution available today.

TMCnet: What about video? Many IT insiders, such as Cisco CEO John Chambers (News - Alert), have said that video-based communications are the wave of the future. The United States is lagging behind Europe as far as mobile video communications go, yet we hear every day about new telepresence and similar systems proliferating. What does ClearSight see as the future of those technologies and the future of its services that support them?

SL: Video is becoming more and more important as Cisco (News - Alert) is pushing its MXE or media engine switch. When ClearSight was building a VoIP engine for our tools, the video component was part of the architecture, so our solutions are already prepared to support this fast growing technology. In fact, that support is one of the reasons why ClearSight’s VoIP and video features are so highly regarded today. We developed a video QoS measurement called VQ-Factor which is the equipment part of Audio MOS. We will be able to go into Video CODEC to provide much deeper Video QOS measurement and alarming. Additionally, as 3G service is getting popular, ClearSight is also anticipating adding our audio and video features into PDA or mobile devices.

By Michael Dinan
TMCnet Editor

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Private Telephone Systems Reduce POTS Line Costs

Computer data networks and circuit-switched voice networks are completely
separate, with individual staffing, billing, maintenance, and accounting systems.
Although the maintenance costs of computer networks are affordable
for most companies, the recurring charges for traditional forms of telephony
are huge for small, medium, and large multilocation companies.

VoIP is designed to converge (integrate) a company’s voice needs onto the company’s
existing computer network. If a company does this, they can eliminate
most (if not all) recurring circuit-switched telephony charges.

In the past, the POTS world had only two types of services: local and long distance.
Local service covered the entire metropolitan area, with no distinctions
for the various levels of toll service that we have today. In the early
days of the telephone, long-distance cost customers dearly. A call from New
York to the west coast might have cost $3 to $4 per minute. Today, that same
call might cost a consumer $.02 to $.05 per minute and a corporate caller $.01
to $.03 per minute. The corporate customer is most likely on some sort of
dedicated private network consisting of a phone system connected to the
PSTN.

It might appear that the cost of telephony today is dirt cheap in historical
terms. This would be a mistaken conclusion. In addition to the carriers getting
more organized and the government increasing its regulation of the
telecommunications industry, many changes have evolved. These changes
have increased your bottom-line telephone bill and increased the number of
line items on that bill.

Now, instead of just two types of phone service offered on the PSTN (local
and long distance), we have five: local, intralata, intrastate, interstate, and
international. These five services
are based on the origin and destination of a call, using the LATA and
NPA-NXX to determine those locations. In addition, the same system is used
by the government to place various surcharges and fees on each telephone
access line.

No one would argue that the quality of carrier-switched telephony is excellent.
However, the system that has evolved for charging telephony customers
leaves much unsaid and a lot to be desired. Except for local calling, VoIP can
reduce or eliminate the charges of the other four categories.

To lessen the burden of newer and diverse telephone costs, many companies
have acquired their own POTS-based telephone systems. Company-sponsored
telephone systems can reduce the monthly bill that consumers and companies
pay for telephony services. Four different telephony system models have
evolved in the past three decades.

The first model, POTS, has already been described; it is the use of telephony
access lines and carrier services over the PSTN through a carrier. The other
models are the Centrex, KTS, and PBX models. Each of these are discussed in
this sections.

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Eye for IP Telephony

VoIP also makes possible other services that older telephony systems can’t

provide. VoIP telephony services are interoperable, meaning that they work
well over all kinds of networks. They are also highly portable, which means
they will work with any IP-enabled device such as an IP telephone, a computer,
or even a personal digital assistant (PDA).
IP telephony works by taking traditional voice signals and converting them to
a form that can be easily transmitted over a local area network. Thus, the
heart of IP telephony is the same as traditional data networking with computers.
IP-enabled phones handle the voice-to-data conversion well, but don’t be
misled — implementing VoIP doesn’t mean that everyone has to use IP-enabled
phones. The best VoIP providers implement IP telephony in a manner that
protects your investment in existing telephone equipment, even if you have
analog telephone stations. (You’ll find more on this topic in Chapter 10.)
All IP phones have one important thing in common: a built-in network interface
card (NIC), just like a computer uses. The NIC is critical for any network
device because it provides the device with a physical address and a way to
communicate over the network.
The physical address supplied by a NIC is called a MAC address. MAC stands
for media access control. The MAC address uses a standardized address and
is usually represented by six hexadecimal numbers separated by dashes. For
example, the following is a valid MAC address: 00-0A-E4-02-7B-99.
To support IP telephony, a server is typically dedicated to run the software
used to manage calls. Servers are just like personal computers, except they
have more memory, speed, and capacity. The server stores the database that
contains all the MAC addresses corresponding to all the IP telephone extensions
assigned to users. Depending on the size of the LAN and the number of
users, you may use more than one server. For example, some LANs running IP
telephony dedicate a server just for handling voice mail.
Depending on the size of the LAN, one or more devices known as switches
are installed. These switches are boxes that have a series of ports into which
all LAN-addressable devices ultimately connect. (Examples of LAN-addressable
devices include computers, printers, wireless access devices, gateways, and
storage devices.) Usually the switches are set up in the communications closets
around the LAN, and they operate 24/7. All the switches are interconnected,
often with fiber-optic cable.
In a nutshell, all network devices, including your IP telephone, must physically
connect to the LAN through a port on a switch.

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