Voice over IP (VoIP)
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`VoIP
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`Table of Contents
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`Introduction
`What is VoIP?
`How it works
`Benefits of VoIP
`Conclusion
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`Introduction
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`Dating back over 100 years, traditional voice networks and the telephone have become a
`integral part of modern society. In fact, it is not unusual, even in remote parts of the world, for
`people to feel that they are entitled to basic telephone service. Obviously, the telephone and the
`associated networks are a large part of modern communications and technology. However, in
`recent years, data networks have been growing at a tremendous rate, largely due to the growing
`Internet. According to some experts, data traffic is predicted to soon exceed traditional voice
`traffic. As a result, more and more companies have become interested in implementing VoIP.
`But what exactly is VoIP and how does it work? Also, what are the benefits of VoIP?
`
`What is VoIP?
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`VoIP, or Voice over IP, is an application that enables data packet networks to transport real time
`voice traffic. It consists of hardware and software that allows companies and persons to engage
`in telephone conversations over data networks. According to an article written by techguide.com,
`"VoIP can be defined as the ability to make telephone calls (i.e., to do everything we can do
`today with the PSTN) and to send facsimiles over IP-based data networks with a suitable quality
`of service (QoS) and a much superior cost/benefit." It is also known as Internet Telephony.
`However, the latter term is often used in reference to calls made over the public Internet, and
`VoIP is often used to refer to calls made on a private network.
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`The traditional voice network, or POTS (plain old telephone system), uses circuit switching
`techniques. This means that a particular communications uses a dedicated path for the duration
`of the call. Although this provides a very reliable connection for voice transmissions, it makes
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`VoIP
`very inefficient use of bandwidth. On the other hand, data networks generally uses packet or cell
`switching technologies. These use Statistical Time Division Multiplexing (STDM) in order to
`dynamically allot bandwidth to a particular stream of data, based on its requirements and the
`requirements and demands of other data on the network. This provides for much more efficient
`use of available bandwidth but can create problems for voice traffic, which is very sensitive to
`delay. Because each packet is individually routed across the network, this makes packet
`switching networks inherently less efficient in dealing with voice traffic and poses a number of
`challenges to a quality voice transmission. These include: packet loss, delay(echo),
`jitter(variable delay) and unreliable and out of order packet delivery due to the connectionless
`nature of packet networks. So, then, how does VoIP work, and how does it overcome these
`obstacles in order to provide reliable, quality telephone conversations?
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`How does it work?
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`In order to deal with these issues and provide a voice service with a reasonable measure of
`quality, there are many techniques that are employed in order to deal with network congestion
`and delay by making better use of bandwidth. These bandwidth saving schemes include
`prioritization, fragmentation, jitter buffering, voice compression, silence suppression and echo
`cancellation. This is where the various protocols, such as H.323 come in, as standards are being
`set to control the quality of voice transmissions on a data network.
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`Prioritization techniques are related to QoS (quality of service), which is a method of
`guaranteeing throughput for certain traffic on the network. This can ensure that voice traffic on a
`data network is given high priority. This prioritization can be based on location, protocol or
`application type. Protocols used to ensure this QoS are RTP (Real Time Protocol) and RSVP
`(Resource Reservation Protocol).
`
`Fragmentation divides packets into smaller fragments so that their priority can be ensured. This
`can help reduce the overall delay of voice delivery. However, on IP based networks, this can
`create extra overhead because of the large size of IP headers (20 bytes). So although
`necessary, fragmentation alone cannot ensure the reliable delivery of real time voice
`applications.
`
`This is why compression is also necessary. Various codecs (coder/decoder) standards have
`been implemented. ITU G.723, which provides for 3.1 kbps bandwidth over 5.3 and 6.3 kbps
`channels has been adopted for use with VoIP. The ITU G.729 standard has been adopted for
`VoFR (Voice Data Convergence Glossary).
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`In IP-based networks, packets that belong to the same transmission (whether voice or data) do
`not always arrive with the same amount of delay. For example, packets 1-5 of a given data
`stream may all arrive with a consistent amount of delay between each packet, but the delay
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`between packet 5 and 6 may be twice as long. This variation in delay is referred to as "jitter." As
`a result, voice transmissions will sound unnatural. When the next packet in a voice stream does
`not arrive in time, the previous packet is usually replayed. However, this can create
`conversations that lack a natural quality. In order to handle this delay variability, a jitter buffer is
`established. This allows packets to be collected into a buffer and held there long enough for the
`slower packets to arrive so that they can all be played in proper sequence and in a natural voice
`flow. Although this can remove packet delay, this creates additional overall delay. According to
`Gil Biran, Vice President of Research and Development for RAD Data Communications, "the
`jitter buffer should fit the network's differential delay." This will provide for the necessary balance
`between packet delay and overall delay, allowing for voice quality transmissions.
`
`In human telephone conversations, generally only about 50% of the full duplex bandwidth is
`used at any given time. This is because one person is generally listening while the other is
`talking. When you couple this with the fact that there are natural pauses, pauses for breath and
`between words, the total required bandwidth for a conversation is reduced an additional 10%.
`This means that there is between 50-60% of the available bandwidth that is not being used.
`Silence suppression techniques take advantage of this by detecting when there is a gap and
`then suppresses the transmission of these silences. This can result in more bandwidth being
`available for other transmissions. However, because these silences are necessary for the
`conversation to sound natural, the receiving device must interpret the lack of packets and re
`insert the silent spots into the output.
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`When the total end to end delay of a voice transmission is greater than 50 milliseconds, echo
`becomes a problem that can detract from the quality of the conversation. An echo cancellation
`unit solve this problem by performing echo cancellation on the signals. ITU G.165 or G.168
`provide the standards and requirements for echo cancellation.
`
`When dealing with data transmission on IP networks, TCP (Transmission Control Protocol)
`handles any packets that may be lost due to congestion or link failures by issuing
`acknowledgements and requesting retransmittal of lost packets. Although this works well for
`data, this method is not efficient for time sensitive information such as voice. In order to help
`ensure a quality voice conversation, packet losses greater than 10% are not tolerable
`(Techguide). For any packet losses under 10%, interpolation (playback of the last packet) can
`help maintain a continuous flow of voice with minimal distraction to the quality.
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`In order for different manufacturers to implement these various techniques and maintain
`interoperability, various standards have been recommended and approved. The ITU H.323
`standard is a sort of umbrella standard that includes in its family the various standards for
`compression and call control. H.323 describes equipment that provide multimedia
`communications on networks that do not ensure a guaranteed QoS (such as IP based
`networks). There are also other protocols designed for VoIP client applications. These include
`SGCP, SAP, SIP, RTSP and SDP. You can find additional information on these protocols by
`using the links to Additional Resources on the left. In addition, I plan to do further research and
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`provide a tutorial on these in the near future as well.
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`Obviously, enhancements to equipment and standards for VoIP are constantly being improved.
`You may have heard about VoIP but may have wondered whether or not it is a good solution for
`your company. So, then, what are the benefits of implementing VoIP?
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`Benefits of VoIP
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`Many companies are seeing the value of implementing VoIP in their data networks for many
`reasons. These include:
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`Cost reduction - low cost phone calls
`Convergence of data/voice networks - unification
`Simplification and consolidation - centralized management
`
`As data networks continue to grow, implementing VoIP can be a very appealing option that can
`allow for reduced costs and provide for greater flexibility. In addition to replacing internal voice
`networks at large corporate offices, VoIP can be used to connect various branch offices through
`existing WAN links. This gives companies an alternative to the PSTN that can continue to grow
`and be scaled to fit their needs.
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`Conclusion
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`As data traffic continues to increase and surpass that of voice traffic, the convergence and
`integration of these technologies will not only continue to improve, but also will pave the way for
`a truly unified and seamless means of communication. Implementing VoIP can provide
`significant benefits and savings to your company. To find out more about this exciting new
`technology and to see how it can be implemented to fit your needs, please feel free to contact
`Donald G.W. Lau, an Applications Engineer at ComTest Technologies.
`
`ComTest Technologies proudly represents a number of VoIP manufacturers. These include, but
`are not limited to, the following:
`
`Oki
`Alcatel
`Vive
`more...
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`To learn more about VoIP, visit these links:
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`VoIP
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`Bandwidth Management and QoS
`Voice over IP Overview at Protocols.com
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`References
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`This page was written by Will Twiggs, an associate with ComTest Technologies, Inc. For more information or if you
`have questions about this material, you can contact the author at william@comtest.com
`
`Voice over Frame Relay, IP and ATM by Gil Biran
`
`Voice over IP by Techguide.com
`
`This page created by Will Twiggs
`Best viewed at "800x600" using MS IE 4+
`Last updated on 12/18/2000
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