The Future of Voice over Internet Protocol
Scott T. Martin
Department of Computer Science and Software Engineering
University Of Wisconsin-Platteville
Abstract
Advances in the telephone communication industry have led to the implementation of Voice over Internet Protocol (VoIP). VoIP makes it possible to route phone calls over the Internet,which has greatly decreased the cost and the amount of data being sent between call locations. Power concerns, network consistency, virus concerns, and emergency calls are all issues that are still being dealt with that continue to hold back its success. In this paper, I plan on discussing how VoIP works. I will also cover the benefitsof switching over to VoIP and issues that are currently preventing VoIP from becoming the dominant method of telephony.
Introduction
When Alexander Graham Bell made the first telephone call to his assistant Thomas Watson in 1876, his invention was a huge breakthrough in communications. However, in the 100 years that followed that monumental call, there were no massive advances in the telephone communication industry. In the late 20th century, researchers wanted to know if it would be feasible to send video and voice over IP networks. This technology would be especially useful for sending information across corporate intranets and across the Internet. The research done by these individuals has led to the creation of Voice Over Internet Protocol.
History
Voice over Internet Protocol’s roots can be traced back to one of the first computer network protocols for transporting human speech known as the Network Voice Protocol. NVP was first implemented in December of 1973 by Internet researcher Danny Cohen of the Information Sciences Institute at the University of Southern California. The project had several goals. One was to develop and demonstrate the feasibility of secure, high-quality, low-bandwidth, real-time, full-duplex digital voice communications over packet-switched computer communications networks. A second goal was to supply digitized speech which can be secured by existing encryption devices. This research project hoped to create a digital high-quality, low-bandwidth, secure voice handling capability as part of the general military requirement for worldwide secure voice communication. [1]
The protocol had two separate parts: control protocols and a data transport protocol. The control protocol part handles basic features such as origin and destination information, ring tones, and call termination. The data transport protocol handled the encoded speech.
Circuit Switching
Current phone systems are based on circuit switching. It’s a very reliable system that is somewhat inefficient. In the early phone systems, for two parties to have a conversation, they had to make a connection between the two locations that stayed open for the duration of the call. The problem is that the call has to have a dedicated line specifically for that call, which was rather expensive since the line was not available for other use. In today’s phone systems, the call would be less expensive due to the high bandwidth of the fiber optic cable system that calls are transmitted over.
Figure 1: Example of a Circuit Switched Connection
However, current phone calls are very inefficient. Each phone call today is transmitted over fiber optic lines at a fixed rate of 64 kilobits per second in each direction. So for every second that a conversation lasts, 128 kbps (or 16 kilobytes) are being transmitted. If a call lasts 10 minutes, this means that roughly 10 megabytes of data is being sent back and forth. For the most part, when one person is talking, the other is listening, meaning that the 10 megabytes of data could be cut in half to be a better use of resources. Better yet, this could be cut down even more if dead air were able to be removed, moments when neither party is talking. If data was only transmitted when one party was saying something, it would be much more effective. [2]
Packet Switching
Packet switching accomplishes the same as circuit switching but works in a completely different way. Instead of opening a constant connection between the two parties, packet switching opens up a short connection, just long enough to send a small chunk of data called a packet. The sending computer breaks the data into small pieces, addressing each packet with an address as to where its destination is. Each packet contains a payload which contains a piece of the original message. The sending computer sends each packet to the nearest router and then forgets about it. Each packet can be sent along any one of thousands of paths to the destination computer. Once it arrives at its destination, the packets are reassembled into the original message.
Figure 2: Example of a Packet Switched Connection
The benefit to this system is that the network is free to route the packets along the least congested and cheapest paths. The computers that sent the information are not tied up on one connection, allowing for the sending and receiving of other information from other computers. The theory and implementation of packet switching is the major idea behind Quality of Service (QoS). [2]
VoIP Implementation
VoIP uses this internet technology of packet switching to provide phone service. Packet switching allows for multiple calls to occupy the space of one call on a circuit switching network. In the previously used example, if the 10 MB call that was made on a circuit switching network had been on a packet switching network, it would be compacted enough to allow four or five calls in place of the one call.
A VoIP call from home works similarly to a regular call. A call is initiated by picking up a receiver, which sends a signal to an analog telephone adapter. The ATA sends back a dial tone alerting you that you have a working Internet connection and that you are ready to make a call. The sender then dials a phone number which is sent to a VoIP call processor. This call processor is called a soft switch, which will be covered more in depth a little later in this paper. The call processor translates the number entered into a destination IP address. Once this address is known, a signal is sent to the receiver’s phone, alerting it to start ringing. Once the connection is made, a session is created between the two computers. This alerts the two computers to expect packets from each other. The data transfers during the call are handled by the regular infrastructure of the Internet.
While the call is going on, the ATAs on each end convert the data back and forth between digital and analog formats. When the call is completed, one party hangs up, terminating the signal between the phone and the ATA. The ATA then sends a signal along to the call processor ending the session. One of the best advantages about this technology is that it uses technology already in common use. This technology will allow telephone networks to communicate the way the computers do, making the transition from the old phone system to VoIP considerably easier. [2]
Codecs
Coder-decoders (or codecs) are used to convert the analog audio signal into a digital signal for transmission over the internet. After the transmission is complete, a codec is again used to reassemble the digital signal into an uncompressed analog audio signal. This audio signal is what is played and is what the receiver of the call hears.
Codecs create the digital signal by sampling the original analog audio signal several thousand times per second. Each tiny sample is converted and compressed into digital data for transmission. When the tiny data samples are received, they are reassembled into nearly the original analog signal. Because there are so many pieces, the gaps between the reassembled pieces are inaudible to the human ear so it sounds like one continuous piece of audio.
The most commonly used codec in VoIP is the G.729A codec which has a sampling speed of 8,000 times per second. While it is one of the lower sampling rates, it is the best compromise of sound quality and transmission speed. Sampling rates of 64,000 and 32,000 times per second are both commonly used in VoIP technology.
Codecs are controlled by advanced algorithms that sort, sample, compress, and pack the data for transmission. The most commonly used algorithm in VoIP is the CS-ACELP (conjugate-structure algebraic-code-excited linear prediction) algorithm. This algorithm organizes the data packets and streamlines the available bandwidth for the call. This algorithm is responsible for the transmission rule that was previously mentioned in this paper that stated “if no one is talking, don’t send any data.” This rule creates the large gap in performance for packet switching compared to circuit switching that makes VoIP possible. [2]
Soft Switches
The central call processor that was mentioned earlier is a specially designed database mapping program known in the industry as a soft switch. This switch connects the caller to the destination phone. The caller and the hardware making the call are referred to as endpoints. The soft switch uses information about the endpoint to connect the call. The location of the endpoint, the phone number attached to the endpoint, and the IP address of the endpoint are all needed to make the connection.
When a call is made via VoIP, the sending endpoint sends a request to the soft switch. This request is trying to locate the IP address the call needs to be routed to. If the soft switch doesn’t contain the information about the needed endpoint, the soft switch passed the request off to other soft switches until one can find the number needed. When the correct endpoint is located, the information about the destination endpoint is sent back to the sender. Finally, a connection can be made and data can be exchanged between the two locations.
This soft switch is what allows all the different devices that currently support VoIP to talk to each other. By forcing all VoIP connection to communicate the same way, all phones, computers, and WiFi phones can work together interchangeably. [2]
Industry Protocol Standards
Many different protocols exist to manage the way the software and hardware for VoIP work together. The most common and most widely used protocol is H.323. This standard was originally created by the International Telecommunication Union (ITU) for use in video conferencing. It provides specifications for real-time, interactive videoconferencing, data sharing and audio applications. The H.323 protocol is actually a collection of previously created protocols designed for other systems. This collection can be seen at the bottom of the previous page. The major flaw with the H.323 protocol is that it was not specifically designed for use with VoIP.
Figure 3: Protocols used in the H.323 Protocol Suite
Another major protocol in use in VoIP systems is the Session Initiation Protocol (SIP). SIP is a smaller, more efficient protocol when compared to H.323 as it was created for use with VoIP. SIP handles five major parts of the connection process for VoIP: user location, user capabilities, user availability, call setup, and call handling. Other protocols in use today include Megaco H.248, Media Gateway Control Protocol, Remote Voice Protocol over IP Specification, Session Announcement Protocol, Simple Gateway Control Protocol, and Skinny Client Control Protocol. [3]
VoIP Service Types
Voice over Internet Protocol can be used in several different ways, via a home phone with an adapter, a specially designed Internet phone, or through a computer.
In order to convert home phone service to VoIP, an analog telephone adapter (ATA) is needed. The ATA changes sounds from an analog signal to a digital signal. After the signal is converted, it is sent out over the Internet to a routing station. Most VoIP service providers include an ATA free of charge when subscribing to package deals.
IP phones come in two varieties: Ethernet phones and WiFi phones. Ethernet phones look just like the phones in homes today. These phones eliminate the need for an ATA by incorporating all the hardware and software the ATA provides allowing the phone to plug directly into a router bypassing the computer completely. WiFi phones work in a similar fashion to cell phones, allowing the owner to make a call from anywhere in the world that has a WiFi hotspot.
Figure 4: Connection of multiple types of VoIP services
Computer-to-computer calls are the simplest way to make a VoIP call. All it takes is a microphone, a speaker, a sound card, and an Internet connection, all things most computers already have. Cable and DSL connections work the best for VoIP calls. The cost is one of the most appealing things about computer-to-computer calls. There is the cost for the necessary software for your computer, which is usually very small or nonexistent. Also, the cost of Internet provider service is still there. There is no cost for calls made however, no matter the distance, making this service extremely appealing to those making lots of long distance or international calls. [4]
Benefits
VoIP has several very appealing benefits to its use. Primarily, the cost of using VoIP compared to conventional phone systems is considerably lower. Calls from computer to computer are basically free and most long distance calls could be made at almost no cost. VoIP would have extremely high or no limits on call volumes, making it ideal for use in the corporate world. International calls are not free but when compared to regular international phone companies, the rate is considerably better, as shown above.
Table 1: Cost Compassion of Calls Made From the United States
Additionally, most VoIP companies bundle many services together in one low-priced package. These bundles usually cost less than basic phone service and include several services that would cost you extra through a regular phone company, such as Soft Phone, Caller ID, Voicemail, Call Waiting, Call Forwarding, and many others. As an added benefit, many of these services can be managed online. Soft Phone is a feature that allows for the owner to turn a PC into a working phone. With this service, calls can be made or received and the user can access their voicemail from their computer, making a laptop into a phone that works anywhere in the world where there is an Internet connection. [10]
Finally, with VoIP making connections through the internet, it allows for considerably higher bandwidth data transfers. This will allow VoIP service providers to offer their customers streaming movies, televisions, and high access speeds than anything a cellular phone service provider could offer.
Drawbacks
As with any new technology, there are still many flaws in the system that need to be worked out for VoIP to dominate the phone service market. One of the biggest advantages that cell phones and home phones have over VoIP continues to be the availability of 911 emergency services. 911 calls are identified by the number being called from which makes it possible for emergency services to locate the source of the call if the caller cannot give their location. This service is vitally important, but early VoIP services had no way to locate the caller. Since VoIP calls are routed and identified by IP number, it was extremely difficult to route calls to the correct emergency call center. In 2005, the FCC sent a warning to VoIP providers requiring them to fix the problem; E911 is now available through most VoIP service centers but it still experiences problems routing emergency calls to the correct location.
Another major problem that VoIP has is its dependence on wall power. Current home phones (not cordless sets) will still work if the power goes out; VoIP needs its own stable power source. With no power, VoIP phones cannot connect to the Internet and become better paperweights than phones.
As VoIP works through the internet, it only works as well as the network it is connected works. Latency issues, jitter, and packet loss are all issues that can cause phone calls to become distorted, garbled or lost because of transmission errors. Stability in Internet data transfers need to be guaranteed before VoIP could replace traditional phones systems completely. Additionally, VoIP is vulnerable to viruses, worms, and hackers. These attacks are rare, but the threat still exists. VoIP companies are continually developing more advanced encryption to counteract these risks.