How Computers Work (EMMA) Orientation



EMMA HS2 Outline Week #9

Web Cameras and Video Conferencing

Use web cameras and video conferencing applications PowerPoint

How VoIP Works

Videoconferencing

Skype

Install Web Cams & Headsets

Test Web Cams & Headsets

Videoconferencing in Class

Documentation Online

Web Cams & Video Conferencing PowerPoint, B203-02, B203-03, How VoIP Works, Videoconferencing, Skype

Homework - B203-02, B203-03, VoIP/Videoconferencing/Skype Quiz

How VoIP Works

by Robert Valdes and Dave Roos

If you've never heard of VoIP, get ready to change the way you think about long-distance phone calls. VoIP, or Voice over Internet Protocol, is a method for taking analog audio signals, like the kind you hear when you talk on the phone, and turning them into digital data that can be transmitted over the Internet.

How is this useful? VoIP can turn a standard Internet connection into a way to place free phone calls. The practical upshot of this is that by using some of the free VoIP software that is available to make Internet phone calls, you're bypassing the phone company (and its charges) entirely.

VoIP is a revolutionary technology that has the potential to completely rework the world's phone systems. VoIP providers like Vonage have already been around for a while and are growing steadily. Major carriers like AT&T are already setting up VoIP calling plans in several markets around the United States, and the FCC is looking seriously at the potential ramifications of VoIP service.

Above all else, VoIP is basically a clever "reinvention of the wheel." In this article, we'll explore the principles behind VoIP, its applications and the potential of this emerging technology, which will more than likely one day replace the traditional phone system entirely.

The interesting thing about VoIP is that there is not just one way to place a call. There are three different "flavors" of VoIP service in common use today:

• ATA -- The simplest and most common way is through the use of a device called an ATA (analog telephone adaptor). The ATA allows you to connect a standard phone to your computer or your Internet connection for use with VoIP. The ATA is an analog-to-digital converter. It takes the analog signal from your traditional phone and converts it into digital data for transmission over the Internet. Providers like Vonage and AT&T CallVantage are bundling ATAs free with their service. You simply crack the ATA out of the box, plug the cable from your phone that would normally go in the wall socket into the ATA, and you're ready to make VoIP calls. Some ATAs may ship with additional software that is loaded onto the host computer to configure it; but in any case, it's a very straightforward setup.

• IP Phones -- These specialized phones look just like normal phones with a handset, cradle and buttons. But instead of having the standard RJ-11 phone connectors, IP phones have an RJ-45 Ethernet connector. IP phones connect directly to your router and have all the hardware and software necessary right onboard to handle the IP call. Wi-Fi phones allow subscribing callers to make VoIP calls from any Wi-Fi hot spot.

• Computer-to-computer -- This is certainly the easiest way to use VoIP. You don't even have to pay for long-distance calls. There are several companies offering free or very low-cost software that you can use for this type of VoIP. All you need is the software, a microphone, speakers, a sound card and an Internet connection, preferably a fast one like you would get through a cable or DSL modem. Except for your normal monthly ISP fee, there is usually no charge for computer-to-computer calls, no matter the distance.

VoIP phone users can make calls from anywhere there's a broadband connection.

Chances are good you're already making VoIP calls any time you place a long-distance call. Phone companies use VoIP to streamline their networks. By routing thousands of phone calls through a circuit switch and into an IP gateway, they can seriously reduce the bandwidth they're using for the long haul. Once the call is received by a gateway on the other side of the call, it's decompressed, reassembled and routed to a local circuit switch.

Although it will take some time, you can be sure that eventually all of the current circuit-switched networks will be replaced with packet-switching technology (more on packet switching and circuit switching later). IP telephony just makes sense, in terms of both economics and infrastructure requirements. More and more businesses are installing VoIP systems, and the technology will continue to grow in popularity as it makes its way into our homes. Perhaps the biggest draws to VoIP for the home users that are making the switch are price and flexibility.

With VoIP, you can make a call from anywhere you have broadband connectivity. Since the IP phones or ATAs broadcast their info over the Internet, they can be administered by the provider anywhere there's a connection. So business travelers can take their phones or ATAs with them on trips and always have access to their home phone. Another alternative is the softphone. A softphone is client software that loads the VoIP service onto your desktop or laptop. The Vonage softphone has an interface on your screen that looks like a traditional telephone. As long as you have a headset/microphone, you can place calls from your laptop anywhere in the broadband-connected world.

Most VoIP companies provide the features that normal phone companies charge extra for when they are added to your service plan. VoIP includes:

• Caller ID

• Call waiting

• Call transfer

• Repeat dial

• Return call

• Three-way calling

There are also advanced call-filtering options available from some carriers. These features use caller ID information to allow you make a choice about how calls from a particular number are handled. You can:

• Forward the call to a particular number

• Send the call directly to voice mail

• Give the caller a busy signal

• Play a "not-in-service" message

• Send the caller to a funny rejection hotline

With many VoIP services, you can also check voice mail via the Web or attach messages to an e-mail that is sent to your computer or handheld. Not all VoIP services offer all of the features above. Prices and services vary, so if you're interested, it's best to do a little shopping.

Now that we've looked at VoIP in a general sense, let's look more closely at the components that make the system work. To understand how VoIP really works and why it's an improvement over the traditional phone system, it helps to first understand how a traditional phone system works.

VoIP: Circuit Switching

Existing phone systems are driven by a very reliable but somewhat inefficient method for connecting calls called circuit switching.

Circuit switching is a very basic concept that has been used by telephone networks for more than 100 years. When a call is made between two parties, the connection is maintained for the duration of the call. Because you're connecting two points in both directions, the connection is called a circuit. This is the foundation of the Public Switched Telephone Network (PSTN).

Here's how a typical telephone call works:

1. You pick up the receiver and listen for a dial tone. This lets you know that you have a connection to the local office of your telephone carrier.

2. You dial the number of the party you wish to talk to.

3. The call is routed through the switch at your local carrier to the party you are calling.

4. A connection is made between your telephone and the other party's line using several interconnected switches along the way.

5. The phone at the other end rings, and someone answers the call.

6. The connection opens the circuit.

7. You talk for a period of time and then hang up the receiver.

8. When you hang up, the circuit is closed, freeing your line and all the lines in between.

Let's say you talk for 10 minutes. During this time, the circuit is continuously open between the two phones. In the early phone system, up until 1960 or so, every call had to have a dedicated wire stretching from one end of the call to the other for the duration of the call. So if you were in New York and you wanted to call Los Angeles, the switches between New York and Los Angeles would connect pieces of copper wire all the way across the United States. You would use all those pieces of wire just for your call for the full 10 minutes. You paid a lot for the call, because you actually owned a 3,000-mile-long copper wire for 10 minutes.

Telephone conversations over today's traditional phone network are somewhat more efficient and they cost a lot less. Your voice is digitized, and your voice along with thousands of others can be combined onto a single fiber optic cable for much of the journey (there's still a dedicated piece of copper wire going into your house, though). These calls are transmitted at a fixed rate of 64 kilobits per second (Kbps) in each direction, for a total transmission rate of 128 Kbps. Since there are 8 kilobits (Kb) in a kilobyte (KB), this translates to a transmission of 16 KB each second the circuit is open, and 960 KB every minute it's open. In a 10-minute conversation, the total transmission is 9,600 KB, which is roughly equal to 10 megabytes (check out How Bits and Bytes Work to learn about these conversions). If you look at a typical phone conversation, much of this transmitted data is wasted.

On the next page, we'll talk about packet switching.

VoIP: Packet Switching

VoIP phone users can make calls using their Internet connection.

A packet-switched phone network is the alternative to circuit switching. It works like this: While you're talking, the other party is listening, which means that only half of the connection is in use at any given time. Based on that, we can surmise that we could cut the file in half, down to about 4.7 MB, for efficiency. Plus, a significant amount of the time in most conversations is dead air -- for seconds at a time, neither party is talking. If we could remove these silent intervals, the file would be even smaller. Then, instead of sending a continuous stream of bytes (both silent and noisy), what if we sent just the packets of noisy bytes when you created them?

Data networks do not use circuit switching. Your Internet connection would be a lot slower if it maintained a constant connection to the Web page you were viewing at any given time. Instead, data networks simply send and retrieve data as you need it. And, instead of routing the data over a dedicated line, the data packets flow through a chaotic network along thousands of possible paths. This is called packet switching.

While circuit switching keeps the connection open and constant, packet switching opens a brief connection -- just long enough to send a small chunk of data, called a packet, from one system to another. It works like this:

• The sending computer chops data into small packets, with an address on each one telling the network devices where to send them.

• Inside of each packet is a payload. The payload is a piece of the e-mail, a music file or whatever type of file is being transmitted inside the packet.

• The sending computer sends the packet to a nearby router and forgets about it. The nearby router sends the packet to another router that is closer to the recipient computer. That router sends the packet along to another, even closer router, and so on.

• When the receiving computer finally gets the packets (which may have all taken completely different paths to get there), it uses instructions contained within the packets to reassemble the data into its original state.

Packet switching is very efficient. It lets the network route the packets along the least congested and cheapest lines. It also frees up the two computers communicating with each other so that they can accept information from other computers, as well.

Advantages of Using VoIP

VoIP technology uses the Internet's packet-switching capabilities to provide phone service. VoIP has several advantages over circuit switching. For example, packet switching allows several telephone calls to occupy the amount of space occupied by only one in a circuit-switched network. Using PSTN, that 10-minute phone call we talked about earlier consumed 10 full minutes of transmission time at a cost of 128 Kbps. With VoIP, that same call may have occupied only 3.5 minutes of transmission time at a cost of 64 Kbps, leaving another 64 Kbps free for that 3.5 minutes, plus an additional 128 Kbps for the remaining 6.5 minutes. Based on this simple estimate, another three or four calls could easily fit into the space used by a single call under the conventional system. And this example doesn't even factor in the use of data compression, which further reduces the size of each call.

Let's say that you and your friend both have service through a VoIP provider. You both have your analog phones hooked up to the service-provided ATAs. Let's take another look at that typical telephone call, but this time using VoIP over a packet-switched network:

1. You pick up the receiver, which sends a signal to the ATA.

2. The ATA receives the signal and sends a dial tone. This lets you know that you have a connection to the Internet.

3. You dial the phone number of the party you wish to talk to. The tones are converted by the ATA into digital data and temporarily stored.

4. The phone number data is sent in the form of a request to your VoIP company's call processor. The call processor checks it to ensure that it's in a valid format.

5. The call processor determines to whom to map the phone number. In mapping, the phone number is translated to an IP address (more on this later). The soft switch connects the two devices on either end of the call. On the other end, a signal is sent to your friend's ATA, telling it to ask the connected phone to ring.

6. Once your friend picks up the phone, a session is established between your computer and your friend's computer. This means that each system knows to expect packets of data from the other system. In the middle, the normal Internet infrastructure handles the call as if it were e-mail or a Web page. Each system must use the same protocol to communicate. The systems implement two channels, one for each direction, as part of the session.

7. You talk for a period of time. During the conversation, your system and your friend's system transmit packets back and forth when there is data to be sent. The ATAs at each end translate these packets as they are received and convert them to the analog audio signal that you hear. Your ATA also keeps the circuit open between itself and your analog phone while it forwards packets to and from the IP host at the other end.

8. You finish talking and hang up the receiver.

9. When you hang up, the circuit is closed between your phone and the ATA.

10. The ATA sends a signal to the soft switch connecting the call, terminating the session.

11.

|VoIP Terms |

|The central call processor is a piece of hardware running a specialized database/mapping program called a soft switch. See the "Soft Switches" section to learn |

|more. |

Probably one of the most compelling advantages of packet switching is that data networks already understand the technology. By migrating to this technology, telephone networks immediately gain the ability to communicate the way computers do.

It will still be at least a decade before communications companies can make the full switch over to VoIP. As with all emerging technologies, there are certain hurdles that have to be overcome. We'll look at those in the next section.

Disadvantages of Using VoIP

The current Public Switched Telephone Network is a robust and fairly bulletproof system for delivering phone calls. Phones just work, and we've all come to depend on that. On the other hand, computers, e-mail and other related devices are still kind of flaky. Let's face it -- few people really panic when their e-mail goes down for 30 minutes. It's expected from time to time. On the other hand, a half hour of no dial tone can easily send people into a panic. So what the PSTN may lack in efficiency it more than makes up for in reliability. But the network that makes up the Internet is far more complex and therefore functions within a far greater margin of error. What this all adds up to is one of the major flaws in VoIP: reliability.

• First of all, VoIP is dependant on wall power. Your current phone runs on phantom power that is provided over the line from the central office. Even if your power goes out, your phone (unless it is a cordless) still works. With VoIP, no power means no phone. A stable power source must be created for VoIP.

• Another consideration is that many other systems in your home may be integrated into the phone line. Digital video recorders, digital subscription TV services and home security systems all use a standard phone line to do their thing. There's currently no way to integrate these products with VoIP. The related industries are going to have to get together to make this work.

• Emergency 911 calls also become a challenge with VoIP. As stated before, VoIP uses IP-addressed phone numbers, not NANP phone numbers. There's no way to associate a geographic location with an IP address. So if the caller can't tell the 911 operator where he is located, then there's no way to know which call center to route the emergency call to and which EMS should respond. To fix this, perhaps geographical information could somehow be integrated into the packets.

• Because VoIP uses an Internet connection, it's susceptible to all the hiccups normally associated with home broadband services. All of these factors affect call quality:

▪ Latency

▪ Jitter

▪ Packet loss

Phone conversations can become distorted, garbled or lost because of transmission errors. Some kind of stability in Internet data transfer needs to be guaranteed before VoIP could truly replace traditional phones.

• VoIP is susceptible to worms, viruses and hacking, although this is very rare and VoIP developers are working on VoIP encryption to counter this.

• Another issue associated with VoIP is having a phone system dependant on individual PCs of varying specifications and power. A call can be affected by processor drain. Let's say you are chatting away on your softphone, and you decide to open a program that saps your processor. Quality loss will become immediately evident. In a worst case scenario, your system could crash in the middle of an important call. In VoIP, all phone calls are subject to the limitations of normal computer issues.

Videoconferencing

From Wikipedia, the free encyclopedia

A videoconference or video conference (also known as a videoteleconference) is a set of interactive telecommunication technologies which allow two or more locations to interact via two-way video and audio transmissions simultaneously. It has also been called 'visual collaboration' and is a type of groupware.

Videoconferencing differs from videophone calls in that it's designed to serve a conference rather than individuals. It is an intermediate form of videotelephony, first deployed commercially by AT&T during the early 1970s using their Picturephone technology.

Videoconferencing uses telecommunications of audio and video to bring people at different sites together for a meeting. This can be as simple as a conversation between two people in private offices (point-to-point) or involve several sites (multi-point) with more than one person in large rooms at different sites. Besides the audio and visual transmission of meeting activities, videoconferencing can be used to share documents, computer-displayed information, and whiteboards.

Simple analog videoconferences could be established as early as the invention of the television. Such videoconferencing systems usually consisted of two closed-circuit television systems connected via cable.

During the first manned space flights, NASA used two radiofrequency (UHF or VHF) links, one in each direction. TV channels routinely use this kind of videoconferencing when reporting from distant locations, for instance. Then mobile links to satellites using specially equipped trucks became rather common.

This technique was very expensive, though, and could not be used for applications such as telemedicine, distance education, and business meetings. Attempts at using normal telephony networks to transmit slow-scan video, such as the first systems developed by AT&T, failed mostly due to the poor picture quality and the lack of efficient video compression techniques. The greater 1 MHz bandwidth and 6 Mbit/s bit rate of Picturephone in the 1970s also did not cause the service to prosper.

It was only in the 1980s that digital telephony transmission networks became possible, such as ISDN, assuring a minimum bit rate (usually 128 kilobits/s) for compressed video and audio transmission. During his time, there was also research into other forms of digital video and audio communication. Many of these technologies, such as the Media space, are not as widely used today as videoconferencing but were still an important area of research. The first dedicated systems started to appear in the market as ISDN networks were expanding throughout the world. One of the first commercial Videoconferencing systems sold to companies came from PictureTel Corp. who had an Initial Public Offering in November, 1984. Videoconferencing systems throughout the 1990s rapidly evolved from very expensive proprietary equipment, software and network requirements to standards based technology that is readily available to the general public at a reasonable cost.

Finally, in the 1990s, IP (Internet Protocol) based videoconferencing became possible, and more efficient video compression technologies were developed, permitting desktop, or personal computer (PC)-based videoconferencing. In 1992 CU-SeeMe was developed at Cornell by Tim Dorcey et al. In 1995 the First public videoconference and peacecast between the continents of North America and Africa took place, linking a technofair in San Francisco with a techno-rave and cyberdeli in Cape Town. At the Winter Olympics opening ceremony in Nagano, Japan, Seiji Ozawa conducted the Ode to Joy from Beethoven's Ninth Symphony simultaneously across five continents in near-real time.

In the 2000s, videotelephony was popularized via free Internet services such as Skype and iChat, web plugins and on-line telecommunication programs which promoted low cost, albeit low-quality, videoconferencing to virtually every location with an Internet connection.

In May 2005, the first high definition video conferencing systems, made by LifeSize Communications were displayed at the Interop trade show in Las Vegas, Nevada, able to provide 30 frames per second at a 1280 by 720 display resolution. Polycom introduced its first high definition video conferencing system to the market in 2006. High definition has now become standard, with all serious players in the videoconferencing market offering it.

The core technology used in a videoconferencing system is digital compression of audio and video streams in real time. The hardware or software that performs compression is called a codec (coder/decoder). Compression rates of up to 1:500 can be achieved. The resulting digital stream of 1s and 0s is subdivided into labeled packets, which are then transmitted through a digital network of some kind (usually ISDN or IP). The use of audio modems in the transmission line allow for the use of POTS, or the Plain Old Telephone System, in some low-speed applications, such as videotelephony, because they convert the digital pulses to/from analog waves in the audio spectrum range.

The other components required for a videoconferencing system include:

• Video input : video camera or webcam

• Video output: computer monitor , television or projector

• Audio input: microphones, CD/DVD player, cassette player, or any other source of PreAmp audio outlet.

• Audio output: usually loudspeakers associated with the display device or telephone

• Data transfer: analog or digital telephone network, LAN or Internet

There are basically two kinds of videoconferencing systems:

1. Dedicated systems have all required components packaged into a single piece of equipment, usually a console with a high quality remote controlled video camera. These cameras can be controlled at a distance to pan left and right, tilt up and down, and zoom. They became known as PTZ cameras. The console contains all electrical interfaces, the control computer, and the software or hardware-based codec. Omnidirectional microphones are connected to the console, as well as a TV monitor with loudspeakers and/or a video projector. There are several types of dedicated videoconferencing devices:

1. Large group videoconferencing are non-portable, large, more expensive devices used for large rooms and auditoriums.

2. Small group videoconferencing are non-portable or portable, smaller, less expensive devices used for small meeting rooms.

3. Individual videoconferencing are usually portable devices, meant for single users, have fixed cameras, microphones and loudspeakers integrated into the console.

2. Desktop systems are add-ons (hardware boards, usually) to normal PCs, transforming them into videoconferencing devices. A range of different cameras and microphones can be used with the board, which contains the necessary codec and transmission interfaces. Most of the desktops systems work with the H.323 standard. Videoconferences carried out via dispersed PCs are also known as e-meetings.

Echo cancellation

A fundamental feature of professional videoconferencing systems is acoustic echo cancellation (AEC). Echo can be defined as the reflected source wave interference with new wave created by source. AEC is an algorithm which is able to detect when sounds or utterances reenter the audio input of the videoconferencing codec, which came from the audio output of the same system, after some time delay. If unchecked, this can lead to several problems including:

1. the remote party hearing their own voice coming back at them (usually significantly delayed)

2. strong reverberation, rendering the voice channel useless as it becomes hard to understand and

3. howling created by feedback. Echo cancellation is a processor-intensive task that usually works over a narrow range of sound delays.

Multipoint videoconferencing

Simultaneous videoconferencing among three or more remote points is possible by means of a Multipoint Control Unit (MCU). This is a bridge that interconnects calls from several sources (in a similar way to the audio conference call). All parties call the MCU unit, or the MCU unit can also call the parties which are going to participate, in sequence. There are MCU bridges for IP and ISDN-based videoconferencing. There are MCUs which are pure software, and others which are a combination of hardware and software. An MCU is characterised according to the number of simultaneous calls it can handle, its ability to conduct transposing of data rates and protocols, and features such as Continuous Presence, in which multiple parties can be seen onscreen at once. MCUs can be stand-alone hardware devices, or they can be embedded into dedicated videoconferencing units.

Some systems are capable of multipoint conferencing with no MCU, stand-alone, embedded or otherwise. These use a standards-based H.323 technique known as "decentralized multipoint", where each station in a multipoint call exchanges video and audio directly with the other stations with no central "manager" or other bottleneck. The advantages of this technique are that the video and audio will generally be of higher quality because they don't have to be relayed through a central point. Also, users can make ad-hoc multipoint calls without any concern for the availability or control of an MCU. This added convenience and quality comes at the expense of some increased network bandwidth, because every station must transmit to every other station directly.

Skype

From Wikipedia, the free encyclopedia

Skype (pronounced /skaɪp/) is a software application that allows users to make voice calls over the Internet. Calls to other users within the Skype service are free, while calls to both traditional landline telephones and mobile phones can be made for a fee using a debit-based user account system. Skype has also become popular for its additional features which include instant messaging, file transfer, and video conferencing. The network is operated by a company called Skype Limited, headquartered in Luxembourg and partly owned by eBay.

Unlike other VoIP services, the Skype company does not run servers, but makes use of background processing on computers running Skype software—the original name proposed, Sky peer-to-peer (see below) reflects this.

Features

Registered users of Skype are identified by a unique Skype Name, and may be listed in the Skype directory. Skype allows these registered users to communicate through both instant messaging and voice chat. Voice chat allows calls between pairs of users and conference calling, and uses a proprietary audio codec. Skype's text chat client allows group chats, emoticons, storing chat history, offline messaging and (in recent versions) editing of previous messages. The usual features familiar to instant messaging users—user profiles, online status indicators, and so on—are also included.

The Online Number (aka SkypeIn) service allows Skype users to receive calls on their computers dialed by regular phone subscribers to a local Skype phone number.

Video conferencing between two users was introduced in January 2006 for the Windows and Mac OS X platform clients. Skype 2.0 for Linux, released on March 13, 2008, also features support for video conferencing. Version 5 beta 1 for Windows, released 13 May 2010, offers free video conferencing with up to 5 people.

Skype for Windows, starting with version 3.6.0.216, supports "High Quality Video" with quality and features, e.g., full-screen and screen-in-screen modes, similar to those of mid-range videoconferencing systems. Skype audio conferences currently support up to 25 people at a time, including the host.

History

Skype was founded in 2003 by the Swedish entrepreneur Niklas Zennström and the Dane Janus Friis. The Skype software was developed by Estonian developers Ahti Heinla, Priit Kasesalu and Jaan Tallinn, the same individuals who together with Niklas and Janus were also originally behind the peer-to-peer file sharing software Kazaa. In April 2003, and domain names were registered. In August 2003, the first public beta version was released.

One of the initial names for the project was "Sky peer-to-peer", which was then abbreviated to "Skyper". However, some of the domain names associated with "Skyper" were already taken. Dropping the final "r" left the current title "Skype", for which domain names were available.

In September 2005, SkypeOut was banned in South China. In October of the same year, eBay purchased Skype and in fall of 2009 sold a majority stake to an investor group. In December, videotelephony was introduced.

In April 2006, the number of registered users reached 100 million. In October, Skype 2.0 for Mac was released, the first full release of Skype with video for Macintosh, and in December, Skype announces a new pricing structure, with connection fees for all SkypeOut calls. Skype 3.0 for Windows was released.

Throughout 2007 updates (3.1, 3.2 and 3.5) added new features including Skype Find, Skype Prime, Send Money (which allowed users to send money via PayPal from one Skype user to another), video in mood, inclusion of video content in chat, call transfer to another person or a group, and auto-redial. Skype 2.7.0.49 (beta) for Mac OS X released adding availability of contacts in the Mac Address Book to the Skype contact list, auto redial, contact groups, public chat creation, and an in-window volume slider in the call window. In August, Skype users unable to connect to full Skype network in many countries because of a Skype system-wide crash which was the result of exceptional number of logins after a Windows patch reboot ("Patch Tuesday").

In 2008, Skype released various updates including versions for the Sony PSP hand-held gaming system, version 2.0 for Linux with support for video-conferencing, version 4 for Windows (with both a full screen and a compact mode). Many users and observers had commented on the high rate of dropped calls, and the difficulty of reconnecting dropped calls. Skype was also used in the seventh season of the U.S. syndicated game show Who Wants To Be a Millionaire, for a new Ask the Expert lifeline for video chat.

In 2009, Skype 4 was released, their Linux client was updated, and also launched Skype for SIP, a service aimed at business users. At the time of launch around 35% of Skype's users were business users. In April, 2009, eBay announced plans to spin off Skype through an initial public offering in 2010. In August, Joltid filed a motion with the U.S. Securities and Exchange Commission, seeking to terminate a licensing agreement with eBay which allows eBay (and therefore Skype) to use the peer-to-peer communications technology on which Skype is based. If successful, this may have caused a shutdown of Skype in its current form. In September, eBay announced the sale of 65% of Skype to a consortium of Index Ventures and Silver Lake Partners. Early in September, Skype had shut down the Extras developer program. In November, eBay completed the sale of 70% of Skype to a consortium comprising Silver Lake Partners, CPPIB, Andreessen Horowitz, and the original founders valuing the business at USD2.75 billion.

In 2010, a report by TeleGeography Research stated that Skype-to-Skype calls accounted for 13% of all international call minutes in 2009. Out of the 406 billion international call minutes a total of 54 billion were used by Skype calls. In May, Skype 5.0 beta was released, with support of group video calls with up to four participants. Also in May, Skype released an updated client for the Apple iPhone which allowed Skype calls to be made over a 3G network. Originally, a 3G call subscription plan was to be instituted in 2011, but the plan was eventually dropped by Skype. On August 9, 2010, Skype filed with the SEC to raise up to $100 million in an initial public offering.

System and software

Skype uses a proprietary Internet telephony (VoIP) network, called the Skype protocol. The protocol has not been made publicly available by Skype and official applications using the protocol are closed-source. Part of the Skype technology relies on the Global Index P2P protocol, belonging to the Joltid Ltd. corporation. The main difference between Skype and standard VoIP clients is that Skype operates on a peer-to-peer model (originally based on the Kazaa software), rather than the more usual client–server model (note that the very popular SIP model of VoIP is also peer to peer, but implementation generally requires registration with a server, as does Skype).

Many networking and security companies claim to detect and control Skype's protocol for enterprise and carrier applications. While the specific detection methods used by these companies are often private, Pearson's chi-square test and Naive Bayes classification are two approaches that were published in 2007. Combining statistical measurements of payload properties (such as byte frequencies and initial byte sequences) as well as flow properties (like packet sizes and packet directions) has also shown to be an effective method for identifying Skype's TCP and UDP based protocols.

Security and privacy

Skype is a secure communication; encryption cannot be disabled, and is invisible to the user. Skype reportedly uses non-proprietary, widely trusted encryption techniques: RSA for key negotiation and the Advanced Encryption Standard to encrypt conversations. Skype provides an uncontrolled registration system for users with no proof of identity. Instead, a free choice of nicknames permits users to use the system without revealing their identity to other users. It is trivial to set up an account using any name; the displayed caller's name is no guarantee of authenticity. A third party paper analyzing the security and methodology of Skype was presented at Black Hat Europe 2006. It analyzed Skype and found a number of security issues with the current security model.

Skype incorporates some features which tend to hide its traffic, but it is not specifically designed to thwart traffic analysis and therefore does not provide anonymous communication. Some researchers have been able to watermark the traffic so that it is identifiable even after passing through an anonymizing network.

Handout B203-02

Taking Photographs with a Webcam

Expected Outcome – Using your webcam, you will take a small 180x120 photo and a larger 360x240 photo and email them to your instructor.

1. Launch the Webcam software.

2. Study the settings and find the 180x120 setting for the picture size.

3. Look directly into the camera and smile.

4. Click the “Take a Picture” button and hold your pose momentarily.

5. Your photo should show in the gallery.

6. If you aren’t pleased with the outcome of the photo, select it in the gallery and hit your delete button on your keyboard to delete it. Then retake a new one until you are satisfied.

1. Now you are going to take a larger sized photo.

2. Study the settings and find the 360x240 setting for the picture size.

3. Look directly into the camera and smile.

4. Click the “Take a Picture” button and hold your pose momentarily.

5. Your photo should show in the gallery.

6. If you aren’t pleased with the outcome of the photo, select it in the gallery and hit your delete button on your keyboard to delete it. Then retake a new one until you are satisfied.

1. Select the 180x120 photo from your gallery.

2. From your menu, use the “SaveAs” option.

3. Save the photo to your hard drive using the file name iiismall(substituting your three initials for the iii.

1. Select the 360x240 photo from your gallery.

2. From your menu, use the “SaveAs” option.

3. Save the photo to your hard drive using the file name iiilarge(substituting your three initials for the iii.

Email the two files to your instructor.

Handout B203-03

Capturing Video with a Webcam

Expected Outcome – Using your webcam, you will take a small 180x120 video and email it to your instructor

1. Launch the Webcam software.

2. Study the settings and find the 180x120 setting for the picture size.

3. Look directly into the camera and smile.

4. Click the “Record a Video” button to start the camera.

5. Give a brief introduction of yourself for approximately one minute.

6. If you aren’t pleased with the outcome of the video, select it in the gallery and hit your delete button on your keyboard to delete it. Then retake a new one until you are satisfied with your video.

7. Select the video from your gallery.

8. From your menu, use the “SaveAs” option.

9. Save the video to your hard drive using the file name iiivideo(substituting your three initials for the iii.

Email the video file to your instructor.

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