Wednesday, April 1, 2009

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.