NTC/360 - Voice over Internet Protocol (VoIP) 06/15/2005 Voice over Internet Protocol (VoIP) Definition of VoIP " Mr. Watson, come here, I want you!" These were the first words that were spoken over the phone back on March 10 1876. If you combine this invention with the same invention of the first computer that was completed nearly 70 years later in 1946, you would then be able to access VoIP. VoIP is also known as voice over internet protocol. VoIP is in essence the ability to talk with ones voice via computer to computer. In the next few pages you will learn about the history of VoIP, how it works, the requirements that are needed to use VoIP, the benefits and risks of this type of technology, and lastly you will get a glimpse at the future of VoIP. Not many people know what VoIP is or even that it exists.

VoIP was designed to help with the costs for long distance charges. The idea or premise of this type of technology was conceived to use ones phone line and make a call through the Internet. It was designed to be used for both local and long distance calls thus cutting phone bill costs. The VoIP way of communication was started back in 1995 and was the result of work done by some hobbyists in Israel (Interangent, 2005). These hobbyists as well as other great inventors saw a good idea. Their idea was to be able to communicate through the computer as appose to talking over the phone.

The idea was not so much to get away from the phone system but more to avoid long distance charges. In the beginning VoIP was only available when there was a direct personal computer to personal computer connection. Later in 1995 Vocal tec, Inc. released Internet Phone Software (Inter agent, 2005). In the beginning it was fairly difficult to use. Due to the fact that in order to really be able to communicate using VoIP both the computers that were trying to communicate to each other would be required to use the same equipment, have a sound card and use a microphone.

It was a good first effort, but as many firsts, the quality was very poor. VoIP would make great strides in the next three years. In 1998 VoIP had made such great strides that some companies were able to offer personal computer to phone services. The strides also were able to offer phone to phone contact. In the beginning VoIP did not charge their customers but chose to get payment through advertisements. When you made the phone call you would have to wait through an advertisement.

In the beginning, VoIP was still dependant on phone lines through internal or external modems and that kept the quality poor because of static and dropped connections. With the introduction of Ethernet services the quality was able to get better. There was better clarity but there were still some bugs due to static and difficulty making connections. In the beginning VoIP traffic represented rather less than 1% of voice traffic. In 1998 three IP switch manufacturers introduced equipment capable of switching (inter tangent, 2005). With these implementations by the year 2000 the voice traffic had risen only two more percent to 3%.

This is not a huge leap but there are many advances that are still being made and the numbers will continue to grow. The real breakthrough occurred when such systems like Cisco Systems and Nortel started producing VoIP equipment that was capable of switching (Vio preview, 2005). With the introduction of switching it was no longer the CPU's responsibility to switch the voice packets. With the introduction of the new switching this made it to where the computer did not require specific hardware to be able to use VoIP. When the switches were introduced this made the hardware cheaper and allowed larger companies to be able to implement their calls via the internet. VoIP has not really caught on with the general public, but has been primarily used by large corporations.

The general public does not really search out things such as this because they sometimes become complacent in their day to day life. On the other hand, large companies will search out any way to reduce costs. Large companies can reduce costs of business communications, which may include fax, conference calling, along with streaming video application (Try, 2004). Large companies such as call centers were able to implement VoIP to also cut costs. VoIP may not have been something in the past that was readily used by the public. The public has found other ways to cut their long distance phone calls and with the packages that the phone companies have offered it may not be feasible or even become widely used.

The main people entity that sees the value in VoIP is the large corporations and they have in recent years seen the benefits. VoIP is being used and has been noticed and will continue to be sought after by larger corporations whom wish to continue to cut their costs. How VoIP Works VoIP Packet Switching To fully grasp the concept of how Voice over Internet Protocol (VoIP) works, it is necessary to have a full comprehension of the differences between circuit switching and packet switching. The current method of routing phone calls on a public switched telephone network (PSTN) involves circuit switching. Phone calls are routed or switched at the local telephone exchange site's switching centers and directed to the location being called. This connection of the calling and called stations in both directions creates a continuous electrical circuit which is established until one the calling stations releases the transmission.

VoIP, on the other hand, utilizes packet switching to manage data transmissions. Packet switching works by taking packets (messages or fragments of messages), and individually routing them between nodes (network devices) which have no previously established communication path. Packet switching uses optimal bandwidth and minimizes latency during data transmissions. Since VoIP utilizes packet switching, it has become a more efficient way of transmitting data and it possesses several advantages over circuit switched PSTN phone networks. The main difference between packet switching and circuit switching is that packet switching allows numerous phone calls to fill in the area occupied by only one phone call in a circuit switched network. For example, a 15 minute phone call using PSTN would use the full 15 minutes of transmission time at about 128 kips.

However, this same call using VoIP may only take 3 to 4 minutes of transmission time at about 64 kips. The space which has been saved by the VoIP process can then be used to make 3 or 4 more phone calls. VoIP Packet Transmission VoIP takes audio signals and sends them over the Internet Protocol (IP) network to another node or device where they are played back. These analog signals are first digitalized, then transmitted, and finally reconverted to analog signals at the destination node. The way these samples are converted involves an algorithm which is referred to as a compressor / de -compressor or CODEC. VoIP utilizes specific CODECs which are optimized for compressing voice and using less bandwidth than a typical audio stream.

Some of the more common voice CODECs are listed below. o GSM (Global System for Mobile Communications, the most popular standard for mobile phones in the world) o G. 711 (an ITU-T standard for audio commanding) o G. 723.1 (an audio code for voice that compresses voice audio in chunks of 30 milliseconds) o G. 726 (an ITU-T speech code operating at bit rates of 16-40 kips) o G. 729 (an audio data compression algorithm for voice that compresses voice audio in chunks of 10 milliseconds) o HL IN (MPEG-4 Parametric audio coding) o PAC (Perceptual Audio Coding is an algorithm used to compress digital audio by removing superfluous information not recognized by most humans, used in satellite radio) Once the sound is recorded, it is compressed into small samples. These small samples are the compiled into larger chunks and placed into data packets for transmission over the IP network. These compressed data packets generally contain 10 - 30 milliseconds of audio. During packet transmission, some data packets do get lost.

When data packet loss occurs, the CODECs compensate for the losses by filling in the gaps with audio that is suitable for the human ear. This process of filling in the data loss gaps is referred to as packet loss concealment (PLC). In some cases, packets are sent numerous times in order to overcome packet loss (redundancy). Another way packet loss is addressed is through forward error correction (FEC).

This method includes information from previously transmitted packets in successive packets. In addition to packet loss, occasional packet delays may occur. This is a fairly common occurrence in VoIP systems. This is acceptable as the same PLC algorithms can be applied to smooth out the audio and ensure good quality. Another issue that VoIP must deal with during data transmissions is jitter. Jitter occurs when there is a variation in the delay of expected packets.

Jitter can result in choppy voice transmissions or temporary glitches in the VoIP network. To counter this problem, VoIP implements Jitter buffer algorithms. These algorithms queue certain numbers of packets before they are played. This reduces the number of discarded and late arriving packets and helps alleviate "mouth-to-ear" delays. Similar buffers are used by common CD recorders. VoIP Protocols In addition to packet transmission, there are several VoIP protocols which allow packets to flow between communicating devices.

There must be an agreed upon payload format for the contents of the VoIP packets. With the exception of the Inter-Asterisk exchange protocol (IAX), the majority of VoIP systems use Real-time Transport Protocol (RTP) to transmit VoIP traffic. RTP ensures consistent delivery order of voice data packets in an IP network. The services provided by RTP include: o Payload-type Identification Sequence Number ingo Time Stampin go Delivery Monitoring The most widely used signaling VoIP protocol is H. 323. H. 323 was originally created for local area networks (LANs) but has rapidly evolved to address VoIP networks. H. 323 provides specifications for real-time, interactive videoconferencing, data sharing and audio applications such as IP telephony.

Session Initiation Protocol (SIP) is an alternative to H. 323 developed specifically for IP telephony. SIP is smaller and more efficient than H. 323, and it takes advantage of existing protocols to handle certain parts of the process. Some other common VoIP signaling protocols include: o SCCP (proprietary protocol from Cisco) o Mega co / H. 248 (a signaling protocol, used between a Media Gateway and a Media Gateway Controller in a VoIP network) o MGCP (proprietary Cisco protocol used within a VoIP system) o Mine (a proprietary stimulus protocol from Mitel that carries keystroke information from a telephone set to a call control server) o IAX (Inter-Asterisk exchange protocol IAX which carries both signaling and voice data over a UDP stream) VoIP Communicating Methods There are four common methods of communication offered by VoIP. These methods are computer-to-computer, computer-to-telephone, telephone-to-computer, or telephone-to-telephone. The following diagram illustrates these methods. Each of these methods requires different types of hardware and software in order to process the voice / data transmissions.

The next section will go will explain this in greater detail. Hardware / Software Requirements Before one is able to make use of a VoIP service, there needs to be a few other things in place. First off would be to have some type of broadband Ethernet connection. VoIP could work with a dial up internet connection, but really, what would be the point? Depending on where the customer lives, there are three common broadband services to choose from, DSL, Cable, and Satellite.

Either DSL or Cable services are offered in most cities and suburbs these days, but if the customer lives in the country, satellite might be the only option. Any of the three types of services will provide the customer with an IP address which is required to communicate using VoIP. The next thing a home user should have is a LAN (Local Area Network) to share the IP address with multiple devices. Though it is possible, it is not practical to pay for a broadband service only to use VoIP.

There are many ways to set up and configure a LAN, this paper will only discuss using a router. A router is a hardware device that receives the IP address from the broadband provider, and assigns internal IP addresses to every other device connected to it. All of those devices connected together are called a LAN. At this point, there are three major ways to use VoIP and others in development. An analog telephone adaptor (ATA) could be connected to the LAN for use with a touch-tone telephone. An IP telephone could be connected directly into the LAN without the need for an ATA.

And lastly, one could make use of the microphone and speakers on a computer connected to the LAN. We will go into details about each of the three major ways and discuss the pros and cons. Analog Telephone Adaptor This method is the most common with the major VoIP service providers. Somewhere connected within the customers LAN needs to be an ATA.

This required piece of hardware is an analog-to-digital converter. A touch-tone telephone will also need to be connected to the ATA device. For the customer to continue using a touch-tone telephone, the analog signal from the telephone will need to be converted into a digital signal. The digital signal is then able to be transmitted anywhere over the internet. So, of course a touch-tone telephone is also required to use this method of VoIP.

Pros: can call to any phone, works with current phone wiring. Cons: pay service fee. (Tyson, 2005) IP Telephone IP telephones are able to connect directly into a LAN via Ethernet connection. These telephones are designed to convert analog to digital themselves.

Think of them like a touch-tone telephone and an ATA combined into one device. Just like every other network device, every IP phone connected to the network will have its own IP address. Pros: can call to any phone, easy setup, no need for an ATA. Cons: would have to buy the new phones. Computer microphone and speakers The majority of computers these days are equipped with a microphone and speakers. One could buy a head set to plug into the computer for a small price.

There are many software companies offering VoIP applications that make use of the computers ability to input and output sound. This way, the analog to digital conversion is handled by the computer. The most popular of these applications is called Skype and is available for free. Applications such as Skype can be used to call anybody else on the internet for free, that is if they are also using the same software application. Skype now offers advanced features which will let its users call regular telephone numbers for a small per minute fee.

Soon they will have the ability for its users to accept incoming calls made by people using a touch-tone telephone. (Skype, 2005) Pros: Cheap or free, easy, only needs the computer. Cons: Can not use a touch-tone telephone. Benefits / Risks of the technology The benefits and risks of any new technology must be considered when implementing them. There are risks in deploying VOIP technologies but the benefits far out weigh them. Early adopters of VoIP are already experiencing these benefits.

In general, these benefits can be classified into two main categories: lower transmission costs and reduced long-term network ownership costs. By directing voice calls over the corporate data network, rather than through a carrier, companies can significantly reduce their monthly phone bills. These savings are obviously dependent on several factors, including the volume of internal company calls and the distances between company offices. Companies with overseas offices, obviously, can experience the greatest savings, since they can eliminate a great deal of international long-distance charges (TechLand 2005). In addition to reducing a company's monthly phone bills, a converged network also reduces the ongoing costs of owning two separate networks; one for voice and one for data. These costs include the need to buy two separate sets of equipment, the staff time dedicated to the servicing that equipment, and the monitoring of traffic on the two networks.

Other benefits include Single network infrastructure, simple upgrade path and bandwidth efficiency. When installing VoIP in an office only a single cable is required to the desk, for both telephone and data. A simple upgrade path means it is easier to expand, upgrade and maintain. Finally, Bandwidth efficiency can be realized because of the ability to compress more voice calls into available bandwidth then legacy telephone systems. With this efficiency organizations can realize even more cost savings (TechLand, 2005). As stated previously, there are risks involved in the deployment of a converged VOIP network as well.

Any business considering a converged network must have a good understand of these risks in order to mitigate them. The risks which are of greatest concern are; the loss of voice quality, loss of reliability, being locked in to a specific vendor and security threats. Loss of voice quality by using VOIP for most companies would be unacceptable. On the data network packets bounce around somewhat indeterminately.

They can collide and get distorted or even lost. Error-correction mechanisms in Ethernet hardware and the IP protocol itself can compensate for these on the data side, since the millisecond delays that occur as the network don't affect most computer applications. But such problems can adversely affect voice calls; which require a good quality, real time flow of packets from one end of the network to the other. To solve this problem a digital buffer can be used so the flow of packets is not disrupted. An intelligent switch can also switch to the PSTN if network traffic is too congested which would reduce voice quality.

While this means that the company will be charged by its carrier for those calls temporarily, that is a small price to pay for avoiding any potential interruption to normal business operations (TechLand, 2005). In addition, a loss of reliability in a corporate environment will not be acceptable either. We all know what it is like to have our computer freeze or to be told that the network is 'down. ' But this rarely happens with our phones or our telephone carriers.

The average down time for the current telephone system is 5 minutes a year. Immediate and uninterrupted access to others over the phone is an essential aspect of conducting business and it is expected. With a VoIP network the so called single point of failure is where the reliability risk comes in. Any single point of failure in the network means no phones. An intelligent multi-path switch can be used to protect against loss of voice service in two ways.

First, it reroutes calls over the PSTN in the event that the failure of a data networking device. Second, it protects against its own failure by becoming a transparent 'wire' to the PSTN in the event of shutdown. Thus it provides 100% up time for voice services. Another concern which must be considered is idea of vendor lock-in.

If a company adopts a specific vendors approach to voice / data convergence it may cause a lock-in that extends beyond the VOIP solution, forcing a long-term commitment to that vendor's overall networking architecture. This is something that many companies have already experienced with their desktop applications and data network hardware, where use of a particular operating system or routing technology has narrowed their choices in many other areas; such as applications. No one wants their VoIP implementation to result in the same type of limitation of long-term choices. There are switches available that can be used with virtually any PBX and any router architecture. Because it is not directly tied to either device, it is highly flexible and can still be used during and hardware migration. Lastly, security of the VoIP network is growing concern as more and more organizations adopt the technology.

VoIP becomes vulnerable to disruptions caused by denial of service (DOS) attacks, viruses and worms, all of which are frequently targeted at IP infrastructure. Security concerns can be mitigated if the proper precautions are taken. Standards have been developed which if followed will give an organization solid footing to reduce security breach's. Standards such as the ISO 1779 which provides recommendations for information security and the International Telecommunications Union's X. 805 standard which defines security architecture for systems providing end-to-end communications are two examples.

Also, the NRI C provides best practices guidance in a number of areas that relate to VoIP (CSO, 2004). Future of VoIP So what is next for VoIP? Just under a year ago at Chicago's Super comm 2004 industry experts said that nothing was happening with the technology, but looking at it today, vendors have much to say about its advancement, promising that it is the next tech boom. (Wagner, 2004) While strides in the actual VoIP technology are not moving as other technical areas, convergence and quality tools around it are creating huge opportunities and changes across the board. Convergence while not a new term in VoIP, has taken on a new meaning with the world of wireless evolving around it.

Today, customer service based companies are facing new evolutions in keeping people happy. By way of wireless local area networks, voice over IP has integrated mobility and communications. A real world example is a sales representative gets a phone call, but is in the lunchroom on break, the call is forwarded to his PDA and without leaving his fast food combo meal, he can take the call, check inventory for the rush order request and log the order all over one device. Convergence, a conglomeration of voice, video and data over an IP connection should deliver speedy returns on investment and make people work faster, according to the vendors who pitch their product lines. (Wagner, 2004) These types of new applications are the future of VoIP, not the technology itself. Session Initiation Protocol (SIP) is another trend in VoIP.

It is a protocol developed by the IETF MUSIC Working Group and proposed standard for setting up sessions between one or more clients. It is currently the leading signaling protocol for VoIP, and is gradually replacing H. 323 in this role. SIP which is not limited to voice, can mediate any kind of communication session from voice to video to future, unrealized applications. (Wikipedia, 2004) A standard instant messaging protocol based on SIP, called SIMPLE, has been proposed and is under development. SIMPLE can also carry Presence Information, conveying a person's willingness and ability to engage in communications.

Presence information is most recognizable today as buddy status in IM clients such as MSN Messenger and AIM. AOL is offering a product called Internet Phone Service which would identify what buddies are available for calls and the recipient would get a pop-up screen that shows who is calling. AT&T also offers a new product called Enhanced VoIP Controller. Some of the features of the product consist of desktop and server software that can be used to set up and support conference calls. The product transcribes speech to text conversations with 85% accuracy and can be done without voice training and will understand accents as well as be able to translate through bad connections.

The product can be used to search for keywords in a call and a user can zero in on the part of the conference call they need to read. A user can also click on a specific section, and listen, replay, pause or slow down the sections they hear to ensure accuracy. While calls are being played back they appear on the computer screen, creating an instant closed captioning of the conversation. (Greene, 2005) When it comes to quality, unlike data networks, VoIP must work perfectly out of the gate. Upfront network assessments will only go so far in ensuring the quality of a VoIP implementation. Voice is revealing problems with the network that used to go unnoticed, because it is not a static environment, it can be interrupted by spikes in data traffic and surges causing hiccups in the flow.

Companies like Qo via and Em prix are working on new products that have been developed to show where problems occur in the network from point to point. The new quality management tools perform active and passive approaches and include installing a thin client on various end points such as phones and gateways and take local readings and return a quality metric. This recent evolution in testing is different from past quality applications as it is non-intrusive and does not eat up bandwidth. The new applications take an active approach, they generate less traffic and use software probes to listen, but do not require and appliances or software at the remote location, the only thing needed is the IP address at the other end, so in essence, you can assess hundreds of sites per hour. The real benefit to this technology is a proactive approach. The data for review is real time, not after the fact as in the past.

It is the monitoring of actual traffic stream vs. looking at the call report after the fact. Now a service center will know the problem before the calls start coming in and hopefully have it resolved before the impact is too great. Once VoIP call quality monitoring systems are in place, it is tempting to speculate about what a little artificial intelligence would add to the mix. Call systems could be self correcting; moving traffic around to more optimum paths, but the cultural obstacles may be larger than the technological hurdles as fears of "Big Brother" loom.

(Breidenbach, 2005) In conclusion, voice over internet protocol is a technology that has come a long way in its ten year life, creating an alternative to regular phone system by running over the internet. While the standards for its technology are still being developed, it has become a robust protocol that allows for convergence and quality to increase communication. VoIP is not only the future of voice; it is the future of communications.


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