Lesson 13: Voice Technology Basics

Welcome to the Voice Technology Basics lesson. Combined voice and data networks are definitely a hot topic these days. In this module, we’ll start by discussing the convergence of voice and data. We’ll present a bit of history as well so that you understand how this all came about.

We’ll then move into discussing actual voice technology. There’s a lot to cover here and a lot of vocabulary you’ll need to be familiar with. We’ll start with understanding the traditional telephony equipment. We’ll also discuss voice quality issues as well as enabling technologies such as compression that are making voice/data networks possible.

After we cover the technology, we’ll discuss Voice over IP, Voice over Frame Relay, and Voice over ATM. We’ll then cover some of the new applications that are possible on combined voice/data networks.
Finally, we’ll look at how a company might migrate from traditional telephony to an integrated voice/data network.

The Agenda

- Convergence of Voice and Data

- Voice Technology Basics

- Voice over Data Transports

- Applications

- Sample Migration

Convergence of Voice and Data

Today, voice and data typically exist in two different networks. Data networks use packet-switching technology, which sends packets across a network. All packets share the available network bandwidth. At the same time, voice networks use circuit switching, which seizes a trunk or line for dedicated use. But this is all changing...

Data/Voice Convergence—Why?

There is a lot of talk today about merging voice and data networks. You may hear this referred to as multiservice networking or data/voice/video integration or just voice/data integration. They all refer to the same thing. Merging multiple infrastructures into one that carries all data, regardless of type.

In this new world order, voice is just plain data. The trends driving this integration are cost initially--saving money. Significant amounts of money can be saved by doing away with parallel infrastructures. In the long run, though, new business applications are what will drive the integration of data and voice. Applications such as:

- Integrated messaging
- Voice-enabled desktop applications
- Internet telephony
- Desktop video (Intel ProShare, Microsoft NetMeeting, etc.)

So, how does a combined network save money?

Data, Voice, and Video Integration Benefits

The place where you can realize the greatest savings is in the wide-area network (WAN), where the bandwidth and services are very expensive.

The concept here is that at some point, you want voice data “to ride for free.” If you look at the overall bandwidth requirements of voice compared to the rest of the network, it is miniscule. If you had to charge per-packet or per-kilobit, voice is basically “free.”

Companies should experience several kinds of cost savings. Traditionally, the overall telecom budget includes three basic sections: capital equipment, support overhead such as wages and salaries, and facilities. The majority of costs are incurred in the facilities. Facilities charges are recurring, such as leased-line charges which occur every month, as opposed to capital equipment, which can be amortized over a couple of years.

Because facilities are the largest expense, this can also be the place where the most money can be saved. The largest part of the facilities charge is the telecom budget. If the telecom budget can be reduced, money can be leveraged out of that to pay for network expansion.

People tell Cisco, “We have to leverage our budget to converge data, voice, and video. We have exponential applications that demand growth and we don’t know how to finance that.” Cisco advises customers to look at their established budgets and see if there is any way to squeeze money out of them by putting in a more efficient infrastructure with features such as compression, and move all traffic over a single transport mechanism. On average, users can expect a 30 to 50 percent reduction in their IT budgets with convergence.

New applications that include voice are becoming increasingly important as they drive competitive advantage.

Before we get into the nuts and bolts of voice technology, let’s take a look at just a couple of these applications that multiservice networks enable.

Example: Personal Telephony Services

One of the greatest advantages of the new world IP telephony system is the ease of intelligent integration with existing applications.

End users can use their Web browsers to graphically define a “personal rules engine” to create filters on incoming calls, or scan and organize voice mail with the same ease as organizing e-mail; creating personal phone configurations, such as speed dial, and building a valet service that could scan your personal calendar to intelligently route your call.

Traditional PBX call routing and embedded features are based on proprietary applications that are specific to that particular system. Traditional PBXs were an island independent of all the other applications running on the corporate network. Voice mail and e-mail have traditionally been separate because they have been developed on separate systems. In the new world, IP PBX voice mail and e-mail are all part of the same application running in a distributed fashion across the entire corporate network. A single mailbox can now hold your voice messages, e-mail, fax, and video clips.

Example: Integrated Web Search and Calls

Integrated Web search and call applications can be very powerful. If a user is on the Web looking for a product or service and has a question, they can click an icon on the Web browser to speak to an agent. The agent can recognize the user by a cookie file and see the Web page the user is looking at, so the agent is ready to help without having to ask for account information each time the user calls. The call is also intelligently routed based on the information the network has retrieved from the user’s computer. This “click-to-talk” application is one place where convergence is helping to differentiate one service from another.
With integrated Web search and call features, a user can click on a button and talk specifically to an agent who is qualified to address the user’s specific question. The user does not have to go off-line to use the phone, but can actually do it live while on-line.
What this means for e-commerce applications is that sales cycles that could not be completed using just the Web page because a user had a question can now be completed, increasing revenue.
This is not just “pie in the sky.” Cisco does 80 percent of its business through its Web page—that’s $8 billion a year. More importantly, 90 percent of technical support questions are answered through the Web. This not only reduces the number of agents, it gets the information to the customers quickly and increases customer satisfaction. There is a click-to-talk feature in place. To do this, you need to have all your services—data, voice, and video—on a single infrastructure. Now, whether customers are on the Web or on the phone, agents have access to them.

Voice Technology Basics

There is a lot of technology and a lot of issues that are important to understand with voice/data integration. There’s also a lot of jargon and vocabulary. Pace yourself as we move through this section.

We’ll start by looking at TDM versus packet-based networks. Then we’ll cover the traditional telephony equipment. Voice quality issues are essential and we’ll discuss these, along with the technologies that are making voice/data convergence a possibility.

Traditional Separate Networks

So let’s go back to looking at where most companies are today?

Many organizations operate multiple separate networks, because when they were created that was the best way to provide various types of communication services that were both affordable and at a level of quality acceptable to the user community.

For example, many organizations currently operate at least three wide-area networks, one for voice, one for SNA, and another for LAN-to-LAN data communications. This traffic can be very “bursty.”
The traditional model for voice transport has been time-division multiplexing (TDM), which employs dedicated circuits.

Dedicated TDM circuits are inefficient for the transport of “bursty” traffic such as LAN-to-LAN data. Let’s look at TDM in more detail so that you can understand why.

Traditional TDM Networking

TDM relies on the allocation of bandwidth on an end-to-end basis. For example, a pulse code modulated (PCM) voice channel requires 64 kbps to be allocated from end to end.
TDM wastes bandwidth, because bandwidth is allocated regardless of whether there is an actual phone conversation taking place.

So again, dedicated TDM circuits are inefficient for the transport of “bursty” traffic because:

- LAN traffic can typically be supported by TDM in the WAN only by allocating enough bandwidth to support the peak requirement of each connection or traffic type. The trade-off is between poor application response time and expensive bandwidth.

- Regardless of whether single or multiple networks are involved, bandwidth is wasted. TDM traffic is transmitted across time slots. Varying traffic types, mainly voice and data, take dedicated bandwidth, regardless of whether the time slot is idle or active. Bandwidth is not shared.

After: Integrated Multiservice Networks—Data/Voice/Video

With a multiservice network, all data is run over the same infrastructure. We no longer have three or four separate networks, some TDM, some packet. One packet-based network carries all the data. How does this work? Let’s look at packet-based networking.

Packet-Based Networking

As we have just seen, TDM networking allocates time slots through the network.

In contrast, packet-based networking is statistical, in that it relies on the laws of probability for servicing inbound traffic. A common trait of this type of networking is that the sum of the inbound bandwidth often exceeds the capacity of the trunk.

Data traffic by nature is very bursty. At any instant in time, the average amount of offered traffic may be well below the peak rate. Designing the network to more closely match the average offered traffic ensures that the trunk is more efficiently utilized.

However, this efficiency is not without its cost. In our effort to increase efficiency, we run the risk of a surge in offered traffic that exceeds our trunk.

In that case, there are two options: we can discard the traffic or buffer it. Buffering helps us reduce the potential of discarded data traffic, but increases the delay of the data. Large amounts of oversubscription and large amounts of buffering can result in long variable delays.

Traditional Telephony

You can’t really understand voice/data integration unless you understand telephony. This section covers that.

Voice Systems Rely on Public Switched Telephone Networks

In a typical voice/analog telephone network, users make an outside phone call from the phone on their desk. The call then connects to the company’s internal phone system or directly to the Public Switched Telephone Network (PSTN) over a basic telephone service analog trunk or a T1/E1 digital trunk. From the PSTN, the call is routed to the recipient, such as an individual at home.

If a call connects to a company’s internal phone system, the call may be routed internally to another phone on the corporate voice network without ever going through a PSTN.

The PSTN may contain a variety of transmission media, including copper cable, fiber-optic cable, microwave communications, and satellite communications.

Traditional Telephony Equipment

A telephone set is simply a telephone.

KTS: Key telephone systems, found commonly in small business environments, enhance the functionality of telephone sets. The telephones have multiple buttons and require the user to select central-office phone and intercom lines.

EKTS: Electronic key telephone systems improve upon KTS systems. EKTSs often provide switching capabilities and impressive functionality, crossing into the PBX world.

PBX: A private branch exchange system allows the sharing of pooled trunks (outside lines) to which the user typically gains access by dialing an access digit such as “9.” Software in the PBX manages contention for pooled lines. The PBX system has many features, including simultaneous voice call and data screen, automated dial-outs from computer databases, and transfers to experts based on responses to questions rather than phone numbers.

The historical differences between a PBX and a key system have blurred, and both product lines offer comparable feature sets for station-to-station calling, voice mail, and so on. Either the customer owns the PBX or it can be owned and operated by a third party as a service to the end customer. To blur things further, key systems are beginning to offer selected trunk interfaces.

The major differences between a PBX and a key system are the following:

- A PBX looks to the network like another switch—it connects via trunk (PBX-to-PBX) interfaces to the network.
- A key system looks like a phone set (station) and connects via lines (station to PBX).
- PBXs serve the high end of the market.
- Key systems serve the low end of the market.

CO: The central office is the phone company facility that houses the switches.

Switch: An electromechanical device, a switch performs the central switching function of a traditional telephony network. Today, it can include both analog and digital hardware and software.

Toll switch: This switch is used to handle long-distance traffic.

Traditional TelephonySignaling, Addressing, and Routing

We will now consider how phone calls are created and sent through the traditional telephone network Signaling

- Off-hook signaling - how a phone call gets started
- Signaling paths
- Signaling types

Addressing

- Very different from data network schemes
- These differences must be resolved in order to implement integrated data/voice/video (DVV)

Routing

- Dependent on the resolution of the addressing issue

Signaling in a Voice System Sets Up and Tears Down Calls

In any telephone system, some form of signaling mechanism is required to set up and tear down calls. When a caller from an office desk calls someone across the country at another office desk, many forms of signaling are used, including the following:

- Between the telephone and PBX
- Between the PBX and CO
- Between two COs

All of these signaling forms may be different. Simple examples of signaling include ringing of a telephone, dial tone, ringing, and so on.

There are five basic categories of signals commonly used in a telecommunications network:

Supervisory—Used to indicate the various operating states of circuit combinations. Also used to initiate and terminate charging on a call.

Information—Inform the customer or operator about the progress of a call. These are generally in the form of universally understood audible tones (for example, dial tone, busy, ringing) or recorded announcement (for example, intercept, all circuits busy).

Address—Provides information about the desired destination of the call. This is usually the dialed digits of the called telephone number or access codes. Typical types of address signals are Dial Pulse (DP), DTMF, and MF.

Control—Interface signals that are used to announce, start, stop, or modify a call. Controls signals are used in interoffice trunk signaling.

Alert—Ringing signal put on subscriber access lines to indicate an incoming call. Signals such as ringing and receiver off-hook are transmitted over the loop to notify the customer of some activity on the line.

Signaling Between the Telephone and PBX

A telephone can be in one of two states: off-hook or on-hook. A line is seized when the phone goes off-hook.

Off-hook—A telephone is off-hook when the telephone handset is lifted from its cradle. When you lift the handset, the hook switch is moved by a spring and alerts the PBX that the user wants to receive an incoming call or dial an outgoing call. A dial tone indicates “Give me an order.”

On-hook—A telephone is on-hook when its handset is resting in the cradle and the phone is not connected to a line. Only the bell is active, that is, it will ring if a call comes in.

The phone company can provision a Private Line, Automatic Ringdown (PLAR) between two devices. A PLAR is a leased voice circuit that connects two single instruments. When either handset is lifted, the other instrument automatically rings. Typical PLAR applications include a telephone at a bank ATM, phones at an airport that ring a selected hotel, and emergency phones.

Signaling Between the PBX and CO

A telephone system “starts” (seizes) a trunk, or the CO seizes a trunk by giving it a supervisory signal. There are three ways to seize a trunk:

- Loop start—A signaling method in which a line is seized by bridging through a resistance at the tip and ring (both wires) of a telephone line.

- Ground start—A signaling method in which one side of the two-wire line (typically the “ring” conductor of the tip and ring) is momentarily grounded to get dial tone.

- Wink—A wink signal is sent between two telecommunications devices as part of a handshaking protocol. It is a momentary interruption in the single frequency tone indicating that one device is ready to receive the digits that have just been dialed.

With a DID trunk, a wink signal from the CO indicates that additional digits will be sent. After the PBX acknowledges the wink, the DID digits are sent by the CO.

PBXs work best on ground start trunks, though many will work on both loop start and ground start. Normal single-line phones and key systems typically work on loop start trunks.

Signaling Between Switches

Common channel signaling (CCS) is a form of signaling where a group of circuits share a signaling channel.

Signaling system 7 (SS7) provides three basic functions:

- Supervisory signaling
- Alerting
- Addressing

SS7 is an ITU-T standard adopted in 1987. It is required by telecommunications administrations worldwide for their networks. The major parts of SS7 are the Message Transfer Part (MTP) and the Signaling Connection Control Part (SCCP). SCCP works out-of-band, thereby providing a lower incidence of errors and fraud, and faster call setup and take-down.

SS7 provides two major capabilities:

- Fast call setup via high-speed circuit-switched connections.
- Transaction capabilities that deal with remote data-base interactions. SS7 information can tell the called party who’s calling and, more important, tell the called party’s computer.

SS7 is an integral part of ISDN. It enables companies to extend full PBX and Centrex-based services—such as call forwarding, call waiting, call screening, call transfer, and so on—outside the switch to the full international network.

Signaling in a Computer Telephony System

Foreign Exchange (FX) trunk signaling can be provided over analog or T1/E1 links. Connecting basic telephone service telephones to a computer telephony system via T1 links requires a channel band configured with FX type connections.

To generate a call from the basic telephone service set to a computer telephony system, a foreign exchange office (FXO) connection must be configured. To generate a call from the computer telephony system to the basic telephone service set, a foreign exchange station (FXS) connection must be configured.

When two PBXs communicate over a tie trunk, they use E&M signaling (stands for Earth and Magneto or Ear and Mouth). E&M is generally used for two-way (either side may initiate actions) switch-to-switch or switch-to-network connections. It is also frequently used for the computer telephony system to switch connections.

Dialing Within a Phone System

Calls within a phone system are considered on-net or off-net, as follows:

- On-net calling refers to calls that stay on a customer’s private network, traveling by private line from beginning to end.

- A call to an off-premise extension connected by a tie trunk is considered an on-net call. The off- premise telephone is located in a different office or building from the main phone system, but acts as if it is in the same location as the main phone system and can use its full capabilities.

- Off-net calling refers to phone calls that are carried in part on a network but are destined for a phone that is not on the network. That is, some part of the conversation’s journey will be over the PSTN or someone else’s network.

Voice Network Addressing

Voice addressing is determined by a combination of international and national standards, local telephone company practices and internal customer-specific codes. Voice addressing historically has had a geographical connotation, but the introduction of wireless and portable services will render that impossible to maintain.

International and national numbering plans are described by the ITU’s E.164 recommendation. It is expected that the local telephone company adheres to this recommendation.

E.164 is only the public network addressing system. There are also private dialing plans, which are nonstandardized and can be considered highly effective by their users.

This slide depicts a trunk group that bypasses the PSTN. Selection of this trunk has been predefined and mapped to the number 8. The access number could be part of the E.164 addressing scheme or part of a private dialing plan.

Alternate numbering schemes are employed by users and providers of PSTN service for specific reasons. An example of a of non-E.164 plan is carrier identification code (CIC), used for selecting different long-distance carriers, tie lines, trunk groups, WATS lines, and private numbering plans, such as seven-digit dialing.

For integrating voice and data networks, each of these areas must be considered.

Voice Routing

Routing is closely related to the numbering plan and signaling that we just described.

At its most basic level, routing enables the establishment of a call from the source telephone to the destination telephone. However, most routing is much more sophisticated and allows subscribers to select specific services.
In terms of implementation, routing is a result of establishing a set of tables or rules within each switch. As a call comes in, the path to the desired destination and the type of features available will be derived from these tables or rules.
It is important to know how routing is done in the telephone network, because this function will be required in an integrated data/voice network.

Voice over Data Networks

Now that you understand how today’s voice networks work, let’s take a look at how real-time voice over a data network works.

Voice over Packet Networks Allow Real-Time Voice on Data Networks

Voice over packet networks provide techniques for sending real-time voice over data networks, including IP, Frame Relay, and Asynchronous Transfer Mode (ATM) networks.

Analog voice is converted into digital voice packets, sent over the data network as data packets, and converted to analog voice on the other end.

Converting from Voice to Data

Analog voice packets are converted to digital data packets with the following steps:

1. A person speaking into the telephone is an analog voice signal.
2. Coder-decoder (CODEC) software converts the signal from analog to digital data packets suitable for transmission over a TCP/IP network.
3. A digital signal processor (DSP) chip compresses the packets for transmission over the data network.

The data network can be an IP LAN, or a leased-line, ATM, or Frame Relay network.

Converting from Data Back to Voice

Digital data packets are converted to Analog voice packets with the following steps:

4. DSP chip uncompresses the packets
5. CODEC software converts the signal from digital data packets back to analog voice
6. Recipient listens to the voice on their telephone

The “Enabling” Technologies

What’s made this all possible is that in the last ten years, a lot of things have happened in voice technology:

Access price/performance: Access products and services have increased in price performance.

Processing: Digital signal processors (DSPs) specialize in processing analog wave forms, which voice or video inherently are. Today, DSPs are cheaper and higher powered, enabling more advanced algorithms to compress, synthesize, and process voice and video signals. CPUs within the devices have increased in power as well.

Voice compression: Voice compression is used to save bandwidth. A variety of voice compression schemes provide a variety of levels of bandwidth usage and voice quality. These compression methods often do not interoperate. Modem, fax, and dual tone multifrequency (DTMF) functionality are all impacted by voice-compression methods.

Standards: Advances have been made over the past few years that enable the transmission of voice traffic over traditional public networks, such as Frame Relay (Voice over Frame Relay).
Standards, such as G.729 for voice compression, FRF.11 and FRF.12 for voice over Frame Relay, and the long list of ATM standards enable different types of traffic to come together in a nonproprietary network.
Additionally, the support of Asynchronous Transfer Mode (ATM) for different traffic types, and the ATM Forum’s recent completion of the Voice and Telephony over ATM specification, will speed up the availability of industry-standard solutions for voice over ATM.

Higher-speed infrastructure: In general, the infrastructures to support voice in corporate environments and in the public network environments are much higher-speed now, so they can carry more voice traffic and effectively take on the voice tasks for the corporation.

Voice Technologies Compression

What makes voice compression possible is the power of Digital Signal Processors. DSPs have continued to increase in performance and decrease in price over time, and as they have, it has made it possible to use new compression schemes that offer better quality and use less bandwidth. The power of the DSP makes it possible to combine this traffic onto a line that formerly supported perhaps only a LAN connection, but now can support voice, data, and LAN integration.

Looking at this chart, quality and bandwidth tend to trade off. PCM is the standard 64Kbps scheme for coding voice; it is the standard for toll quality. The other compression schemes - ADPCM at 32Kbps, 24Kbps and 16Kbps - offer less quality but more bandwidth efficiency. The newer compression schemes -LDCELP at 16Kbps and CS-ACELP at 8Kbps - offer even higher efficiency but with very high quality very acceptable in a business environment.

ADPCM—Adaptive Differential Pulse Code Modulation: consumes only 32 Kbps compared to the 64 Kbps of a traditional voice call; often used on long-distance connections.

LPC—Linear predictive code: a second group of standards that provide better voice compression and, at the same time, better quality. In these standards, the voice coding uses a special algorithm, called linear predictive code (LPC), that models the way human speech actually works. Because LPC can take advantage of an understanding of the speech process, it can be much more efficient without sacrificing voice quality.

CELP—Code-Excited Linear Predictive voice compression: uses additional knowledge of speech to improve quality.

CS ACELP: Further improvements known as conjugate structure algebraic compression enable voice to be coded into 8-kbps streams. There are two forms of this standard, both providing speech quality as good as that of 32-kbps ADPCM.

Voice Quality Guidelines

Silence Suppression by Voice Activity Detection

Voice activity detection (VAD) provides for additional savings beyond that achieved by voice compression.

Telephone conversations are half duplex by nature , because we listen and pause between sentences. Sixty percent of a 64-kbps voice channel typically contains silence. VAD enables traffic from other voice channels or data circuits to make use of this silence.
The benefits of VAD increase with the addition of more channels, because the statistical probability of silence increases with the number of voice conversations being combined.

QoS Also Plays a Role in Voice Quality

The advantages of reduced cost and bandwidth savings of carrying voice over packet networks are associated with some quality of service issues that are unique to packet networks.
In a circuit-switched or TDM environment, bandwidth is dedicated, making QoS—quality of service—implicit, whereas, in a packet-switched environment, all kinds of traffic are mixed in a store-and-forward manner.
So, in a packet-switched environment, there is the need to devise schemes to prioritize real-time traffic.
So… in an integrated voice data network, QoS is essential to ensure the same high quality as voice transmissions in the traditional circuit-switched environment.

QoS and Voice Quality

Some of the quality of service issues customers face include the following:

Delay—Delay causes two problems: echo and talker overlap. Echo is cased by the signal reflections of the speaker’s voice from the far-end telephone equipment back into the speaker’s ear. Echo becomes a significant problem when the round-trip delay becomes greater than 50 milliseconds (ms). Talker overlap becomes significant if the one-way delay becomes greater than 250 ms.

Jitter—Jitter relates to variable inter-packet timing caused by the network that a packet traverses. Removing jitter requires collecting packets and holding them long enough to allow the slowest packets to arrive in time to be played in the correct sequence, which causes additional delay.

Lost packets—Depending on the type of packet network, lost packets can be a severe problem. Because IP networks do not guarantee service, they will usually exhibit a much higher incidence of lost voice packets than ATM networks.

Echo—Echo is present even in a conventional circuit-switched telephone network, but is acceptable because the round-trip delays through the network are smaller than 50 ms and the echo is masked by the normal side tone that every telephone generates. Echo is a problem in voice over packet networks because the round-trip delay through the network is almost always greater than 50 ms. For this reason, echo cancellation techniques must be used.

Solutions to Voice Quality Issues

Quality of service issues for voice may be handled by the H.323, VoIP, VoATM, or VoFR standards, or by an internetworking device. Following are some solutions to quality of service issues:

Delay—Minimize the end-to-end delay budget, including the accumulation delay, processing delay, and network delay.

Jitter—Adjust the jitter buffer size to minimize jitter. On an ATM network, the approach is to measure the variation of packet levels over a period of time and incrementally adapt the buffer size to match the calculated jitter. On an IP network, the approach is to count the number of packets successfully processed and adjust the jitter buffer to target a predetermined allowable late packet ratio.

Lost packets—While dropped packets are not a problem for data (due to retransmission), they cause a significant problem for voice applications. To compensate, voice over packet software can interpolate for lost speech packets by replaying the last packet, or can send redundant information at the expense of bandwidth utilization.

Echo—Echo cancellation techniques are used to compare voice data received from the packet network with voice data being transmitted to the packet network. The echo from the telephone network hybrid is removed by a digital filter on the transmit path into the packet network.

Effect of QoS on Voice Quality

With all of the “marketing hype” around QoS today, many customers have become skeptical of the claims some vendors are making.
Here’s one way to look at the actual effect of Cisco QoS technologies on voice quality.

The blue line represents the total network data load. The green line represents voice quality without QoS. As you can see, the quality of a voice call rises and falls in response to varying levels of background traffic.

The red line represents voice quality with QoS enabled, showing that high voice quality remains constant as background traffic fluctuates.

Voice over Data Transports

We’ve covered the building blocks for voice/data integration. Now, let’s take a look at the different transports customers can consider.
The most widely used is Voice over IP. Voice over Frame Relay and Voice over ATM are also important so we’ll cover these as well.

Standards—VoIP, VoFR, and VoATM

VoIP:

- International Telecommunications Union (ITU) —International standards body for telephony
- ITU-T H.323—International Telecommunications Union recommendation for multimedia (including voice) networking over IP
- International Multimedia Teleconferencing Consortium (IMTC) —International standards body providing recommendations for multimedia networking over IP, including VoIP
- Internet Engineering Task Force (IETF) —Internet standards body

VoFR:

- FRF.11—Implementation agreement, ratified in May 1997 by the Frame Relay Forum, that defines the transport of voice over Frame Relay
- FRF.12—Provides an industry-standard approach to implement small frame sizes (Frame Relay fragmentation) to help reduce delay and delay variation
- Other related FRF standards —FRF.6 - Customer Network Management, FRF.7 - Multicast, FRF.8 - FR/ATM Service Interworking, FRF.9 - Data Compression, FRF.10 - Frame Relay Network to Network

VoATM:

- ATM Forum:
- Traffic Management Specification Version 4.0—af-tm-0056.000
- Circuit Emulation Service 2.0—af-vtoa-0078.000
- ATM UNI Signaling, Version 4.0—af-sig-0061.0000
- PNNI V1.0—af-pnni-0055.000

Voice over Data Transports

All types of packetized voice implementations lend themselves well to both corporate and service provider use.
The Voice over IP (VoIP) approach provide Internet service providers (ISPs) with a competitive weapon against telecommunications companies, while telecommunications companies prefer a virtual circuit environment using Voice over Frame Relay (VoFR) or Voice over ATM (VoATM).

VoIP, VoFR, and VoATM Quality

In terms of quality, voice over Frame Relay (VoFR), voice over ATM (VoATM), and voice over IP (VoIP differ). However, they also differ in terms of cost and in terms of general usability.

Frame Relay’s variance does have an impact on voice quality, but Frame Relay can maintain a business-quality level of communication at lower cost. Therefore, VoFR is slightly lower cost than VoATM, but VoFR provides some usually undetectable variations in quality.

VoIP can go anywhere from utility quality, if used over the Internet to toll quality, if used over an intranet with QoS mechanisms enabled. Yet it will generally provide the lowest cost for connectivity. Thus, VoIP in intranets is highly viable for the business user today and provides the most attractive cost option of the three.

VoATM, meaning voice over real-time variable bit rate (RT-VBR) or constant bit rate (RT-CBR), is fully deterministic in terms of QoS. Voice quality never varies. However, VoATM is generally more costly to implement than is, say, VoFR.

All three options offer significantly lower costs than the costs of building a private or using a PSTN, and usually require a fraction of the bandwidth.

Voice over IP Components

The Voice over IP standard incorporates other components, including:

- G. standards, which specify analog-to-digital conversion and compression (as described earlier in this chapter).
- H.323 standard, which specifies call setup and interoperability between devices and applications.
- Realtime Transport Protocol (RTP), which manages end-to-end connections to minimize the effect of packets lost or delayed in transit on the network.
- Internet Protocol or IP, which is responsible for routing packets on the network.

ITU-T H.323 Standard

ITU-T H.323 is a standard approved by the ITU-T that defines how audiovisual conferencing data is transmitted across networks.
H.323 provides a foundation for audio, video, and data communications across IP networks, including the Internet.
H.323-compliant multimedia products and applications can interoperate, allowing users to communicate without concern for compatibility.
H.323 provides important building blocks for a broad new range of collaborative, LAN-based applications for multimedia communications.

H.323 sets multimedia standards for the existing infrastructure (for example, IP-based networks). Designed to compensate for the effect of highly variable LAN latency, H.323 allows customers to use multimedia applications without changing their network infrastructure.

By providing device-to-device, application-to-application, and vendor-to-vendor interoperability, H.323 allows customer’s products to interoperate with other H.323-compliant products. PCs are becoming more powerful multimedia platforms due to faster processors, enhanced instruction sets, and powerful multimedia accelerator chips.

Applications enabled by the H.323 standard include the following:

- Internet phones
- Desktop conferencing
- Multimedia Web sites
- Internet commerce
- And many others

H.323 Infrastructure

The H.323 standard specifies four kinds of components, which when networked together, provide the point-to-point and point-to-multipoint multimedia communication services: terminals, gateways, gatekeepers, multipoint control units (MCUs).

H.323 terminals are used for real-time bidirectional multimedia communications. An H.323 terminal can either be a PC or a standalone device running an H.323 and the multimedia applications. It supports audio communications and can optionally support video or data communications.

An H.323 gateway provides connectivity between an H.323 network and a non-H.323 network. For example, a gateway can connect and provide communication between an H.323 terminal and the Public Switched Telephone Network (PSTN). This connectivity of dissimilar networks is achieved by translating protocols for call setup and release, converting media formats between different networks, and transferring information between the networks connected by the gateway. A gateway is not required, however, for communication between two terminals on an H.323 network.

A gatekeeper can be considered the “brain” of the H.323 network. Although they are not required, gatekeepers provide important services such as addressing, authorization, and authentication of terminals and gateways, bandwidth management, accounting, billing, and charging. Gatekeepers may also provide call-routing services.

MCUs provide support for conferences of three or more H.323 terminals. All terminals participating in the conference establish a connection with the MCU. The MCU manages conference resources, negotiates between terminals for the purpose of determining the audio or video CODEC to use, and may handle the media stream. The gatekeepers, gateways, and MCUs are logically separate components of the H.323 standard, but can be implemented as a single physical device.

H.323 Gatekeeper Functionality

Gatekeepers provide call control services to network endpoints. A gatekeeper can provide the following services:

Address translation—Performs alias address to transport address translation. Gatekeepers typically use a translation table to perform the address translation.

Admissions control—Authorizes LAN access based on call authorization, bandwidth, or other criteria.

Call control signaling—The gatekeeper chooses to complete call signaling with endpoints or may process the call signaling itself. Alternatively, the gatekeeper may instruct endpoints to connect call signaling channel directly to another to bypass handling a signal channel.

Call authorization—A gatekeeper may reject calls from a terminal upon authorization failure.

Bandwidth management—Controls the number of terminals that are permitted simultaneous access to a LAN.
Call management—Maintains a list of active calls.

H.323 Interoperability

VoIP works with a company’s existing telephony architecture, including its private branch exchanges (PBXs) and analog phones.

VoIP and H.323 enables companies to complete office-to-office telephone and fax calls across data networks, significantly reducing tolls. New applications are available, including unified messaging that integrates e-mail with voice mail and fax.

Choosing VoIP

Customers may choose VoIP as their voice transport medium when they need a solution that is simple to implement, offers voice and fax capabilities, and handles phone-to-computer voice communications. IP networks are proliferating throughout the marketplace. Thus, many customers can use VoIP today.

Integrating Voice and Data on the WAN

The Voice over IP and H.323 standards define how analog voice is converted to data packets and back again. The next step is to use a company’s existing wide-area network (WAN) to transport voice traffic with data traffic.

Serial (Leased Line) Services

T1 is a private-line digital service, operating at 1.544 Mbps in a full-duplex, TDM mode. The 1.544-Mbps transmission rate provides the equivalent capacity of 24 channels running at 64 Kbps each.

The full-duplex feature of T1 allows the simultaneous operation of independent transmit and receive paths. Each data path operates at a transmission rate of 1.544 Mbps. Companies that need less bandwidth can deploy fractional T1 trunks, using any number of channels needed. A fractional service is tariffed on a linear pricing schedule, depending on the number of T1 channels and the distance covered.
The TDM feature allows logical channels to be defined within the T1 serial bit stream. The T1 bit stream may be channelized in many different ways, as follows:

- A single 1.544-Mbps digital channel (non-channelized) between the user’s premises and the central office (CO)
- 24 independent channels, each providing 64 Kbps of bandwidth
- Any variation of 64-Kbps channel combinations

Each logical channel may be independently transmitted and switched. A combination of voice, video, and data may be transmitted over a single T1 line.
Ideal Applications for T1 Services T1 service is ideal for applications that require continuous high-speed transmission capabilities.

Some common T1 applications include the following:

- High-volume LAN interconnection
- Integrated voice, data, video, and imaging transmission
- Compressed video transmission
- Bulk data transfer

Frame Relay Services

Frame Relay is a packet-switching WAN technology that has achieved widespread support among vendors, users, and communications carriers. Its development has been spurred by the need to internetwork LANs at high speeds while maintaining the lower costs associated with packet-switching networks.

Frame Relay offers very high access speeds. In North America, initial Frame Relay access rates start at 56 Kbps and go up to 1.544 Mbps. In Europe, the initial Frame Relay access rates start at 64 Kbps and go up to 2.048 Mbps. Companies can contract with their service provider for a committed information rate (CIR).

The Frame Relay standard today uses permanent virtual circuits (PVCs). All traffic for a PVC uses the same path through the Frame Relay network. The endpoints of the PVC are defined by a data-link connection identifier (DLCI). The CIR, DLCIs, and PVCs are defined when the user initially subscribes to a Frame Relay service.

Frame Relay allows remote host access for applications such as the following:

- Remote host connectivity
- Credit card authorization
- Online information services
- Remote order entry

Frame Relay supports multiple virtual connections over a single physical interface. This means that Frame Relay is often the ideal solution to provide many users with simultaneous access to a remote location. In these cases, the Frame Relay connection helps optimize the return on investment of the host system.

Voice over Frame Relay

Voice over Frame Relay (VoFR) technology consolidates voice and voice-band data (including fax and analog modems) with data services over a Frame Relay network. The VoFR standard is specified in FRF.11 by the Frame Relay Forum.

VoFR allows PBXs to be connected using Frame Relay PVCs. The goal is to replace leased lines and lower costs. With VoFR, customers can easily increase their link speeds to their Frame Relay service or their CIR to support additional voice, fax, and data traffic.

How VoFR Works

A voice-capable router connects both a PBX and a data network to a public Frame Relay network. A voice-capable router includes a Voice Frame Relay Adapter (VFRAD) or a voice/fax module that supports voice traffic on the data network.

Choosing VoFR

Frame Relay provides another popular transport for multiservice networks since Frame Relay networks are common in many areas. Frame Relay is a cost-effective service that supports bursty traffic well.

Frame Relay enables customers to prioritize voice frames over data frames to guarantee quality of service (QoS).

Asynchronous Transfer Mode (ATM) Services

Asynchronous Transfer Mode (ATM) is a technology that can transmit voice, video, data, and graphics across LANs, metropolitan-area networks (MANs), and WANs. ATM is an international standard defined by ANSI and ITU-T that implements a high-speed, connection-oriented, cell-switching, and multiplexing technology that is designed to provide users with virtually unlimited bandwidth. Many in the telecommunications industry believe that ATM will revolutionize the way networks are designed and managed.

Today’s networks are running out of bandwidth. Network users are constantly demanding more bandwidth than their network can provide. In the mid 1980s, researchers in the telecommunications industry began to investigate the technologies that would serve as the basis for the next generation of high-speed voice, video, and data networks. The researchers took an approach that would take advantage of the anticipated advances in technology and enable support for services that might be required in the future. The result of this research was the development of the ATM standard.

How VoATM Works

Using a WAN switch for ATM, customers can connect their PBX network and data network to a public or private ATM network.

One attractive aspect of ATM is its ability to support different QoS, as appropriate for various applications. The QoS spectrum ranges from circuit-style service, where bandwidth, latency, and other parameters are guaranteed for each connection, to packet-style service, where best-effort delivery allocates bandwidth for each active connection.

The ATM Forum developed a set of terms for describing requirements placed on the network by particular types of traffic. These five terms (AAL1 through AAL5) are referred to as adaptation layers, and are used as a common language for discussing what kinds of traffic requirements an application will present to the network.

- AAL1—Connection-oriented, constant bit rate, commonly used for emulating traditional circuit connections.
- AAL2—Connection-oriented, variable bit rate, used for packet video and audio services.v - AAL3/4—Connection-oriented, variable bit rate.
- AAL5—Connectionless, variable bit rate, commonly used for IP traffic as it provides packetization similar to that done with IP.

Choosing VoATM

VoATM is an ideal transport for multiservice networks, particularly for customers who already have an ATM network installed. ATM handles voice, video, and data equally well.

One attractive aspect of ATM is its ability to support different QoS features as appropriate for various applications.

The ATM Forum has defined a number of QoS types, including:

Constant bit rate (CBR)—An ATM service type for nonvarying, continuous streams of bits or cell payloads. Applications, such as voice circuits, generate CBR traffic patterns. The ATM network guarantees to meet the transmitter’s bandwidth and other QoS requirements. Many voice and circuit emulation applications can use CBR.

Variable bit rate (VBR)—An ATM service type for information flows with irregular but fully characterized traffic patterns. VBR is divided into real-time VBR and non-real-time VBR, in which the ATM network guarantees to meet the bandwidth and other QoS requirements. Many applications, particularly compressed video, can use VBR service. It is fairly common in real networks that will never receive the ceiling value.

Unspecified bit rate (UBR)—An ATM service type that provides “best effort” delivery of transmitted data. It is similar to the datagram service available from today’s internetworks. Many data applications can use UBR service.

Available bit rate (ABR)—An ATM service type that provides “best effort” delivery of transmitted data. ABR differs from other “best effort” service types, such as UBR, because it employs feedback to notify users to reduce their transmission rate to alleviate congestion. Hence, ABR offers a qualitative guarantee to minimize undesirable cell loss. Many data applications can use ABR service.

How Packet Technologies Stack Up for Voice

Because Frame Relay technology was originally designed and optimized as a data solution, you could dedicate a public or private Frame Relay network to data and pay separate dialup or Virtual Private Network (VPN) rates for intracompany phone calls. Provided you can afford the different types of equipment, services, and staff resources required to manage both networks, this choice assures you of the highest quality for each type of traffic today. This option is most likely desirable for sites that are very data-heavy.

Another option is to achieve some level of integration by using one piece of circuit- switching equipment, such as a time-division multiplexer (TDM), to connect both the PBX and LAN server to a wide-area network. Customers gain economies by running all WAN traffic over a single service (rather than receiving multiple WAN bills) and avoiding paying phone company rates for intra-enterprise phone calls.

The costly downside is that within the network, bandwidth is likely to be wasted, because you are still reserving circuits for certain types of traffic, and those circuits sit idle when nothing travels across them.

Applications

Now let’s put it all together. How does it actually work? Let’s look at the voice applications on an integrated voice/data network that replace traditional telephony.

Applications for Integrated Voice and Data Networks

Integrated voice and data networks support a variety of applications, all of which are designed to replace leased lines and lower costs. Each of the applications listed above are discussed on the following pages.

- Inter-office calling
- Toll bypass
- On-net to off-net call rerouting
- PLAR replacement
- Tie trunk replacement

On-Net Call, Intra-Office

A voice-capable router can function as a local phone system for intra-office calls. In the example, a user dials a phone extension, which is located in the same office. The voice-capable router routes the call to the appropriate destination.

Toll Bypass—On-Net Call, Inter-Office

A voice-capable router can function as a phone system for inter-office calls to route calls within an enterprise network.

In the example, a user dials a phone extension, which is located in another office location. Notice that the extension number begins with a different leading number than the on-net, intra-office call. The voice-capable router routes the call to another voice-capable router over an ATM, Frame Relay, or HDLC network. The receiving router then routes the call to the PBX, which routes the call to the appropriate phone extension.

This solution eliminates the need for tie trunks between office locations, or eliminates long-distance toll charges between locations.

Toll Bypass—On-Net to Off-Net Dialing

A voice-capable router can provide off-net dialing to a location outside the local office, through the PSTN.
In the example, a user dials 9 to indicate an outbound call, then dials the remaining 7-digit number (this is a local phone call). The voice-capable router routes the call to another voice-capable router over a Frame Relay or HDLC network. The receiving router recognizes that this is an outbound call and routes it to the company’s PBX in New York. Finally, the PBX routes the call to the PSTN and the call is routed to the appropriate destination.
This solution places the call on-net as far as possible, allowing a local PBX to place a local call. This saves significantly on toll charges.

On-Net to Off-Net Call Rerouting

1. Call attempted on-net
2. Remote system rejects call
3. Call rerouted off-net

At times, on-net resources within an enterprise may be busy. However, telephone calls must still be routed. Using a voice-capable router that deploys Ear and Mouth (E&M) signaling, a router can route calls to a PBX, and ultimately to the PSTN over a Frame Relay or HDLC network.

Keep in mind that a PBX cannot reroute a call after a line is “seized.” Therefore, a voice-capable router can seize an off-net trunk and route a call. This solution guarantees that a phone call is placed, regardless of the load on the network.

PLAR—Automatically Dials Extension

A voice-capable router can replace a Private Line, Automatic Ringdown (PLAR) service from a telephone service provider.

In the example, a user takes the phone off-hook, causing another telephone extension to ring. The voice-capable router recognizes that the phone is off-hook, and routes the call over an ATM, Frame Relay, or HDLC network to the remote router. The remote router then routes the call to the PBX, which rings the appropriate extension. This solution eliminates the need for dedicated PLAR lines.

Tie Trunk Replacement PBX to PBX

Voice-capable routers on a WAN can replace tie trunks between remote locations, thereby saving the cost of tie trunks. In essence, the voice-capable router on either side of the ATM, Frame Relay, or HDLC WAN connection is configured as a tie trunk. The router then routes incoming and outgoing calls through the PBX.

Sample Migration

PBX Networking to New Voice Networking

The next slides graphically illustrate the migration from traditional circuit-switched voice networking to the new packet-switched integrated data/voice/video networking.
Here you see two offices… one in Vancouver and one in Toronto. Each has a PBX to handle the office but all calls inter-office go through the PSTN.

By adding voice-capable routers to the existing data network, connecting them to the existing PBXs, the company can first do toll bypass. This represents bandwidth no longer needed for voice traffic that is now going through the routers.

The PBX tie line also goes away now that its function has been replaced by a path between the voice-capable routers.

You can see here the end result. A much simplified network and considerable cost savings.

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- Summary -

As we have seen today, companies are interested in data/voice/video integration for very basic business reasons:

Reduce costs: Phone toll charges; cost of multiple management methods and multiple types of expertise required to support multiple types of networks; capital expenditures on multiple networks

Enable the new applications needed for business growth: Multimedia (data/voice/video) applications require technologies based on multimedia standards

Simplify network design: Through strategic convergence of data, voice, and video networks

And decision-makers have come to the conclusion that recent technical advancements have brought the benefits of voice/data integration within reach, such as: H.323 standards; gateways; voice-compression, silence-suppression, and quality-of-service technologies.