Cisco Certified Voice Professional (CCVP)

I see there are a lot of materials in my office now to learn about Voice-Over-IP solutions. In addition, my first task now is to help a team that develops Advanced Technology Demo Units Voice and Unified Communications plus Cisco IP Interoperability and Collaboration System (IPICS). Besides, one of my team just finished her Bachelor's Thesis about IP Telephony and Cisco Unified Communications Manager (CallManager), maybe she can explain or teach me something regarding to IP Telephony.
Then, my friend told me that maybe I should take CCVP rather than CCNP since the condition and environment of my office can support me in pursuing CCVP.
The CCVP (Cisco Certified Voice Professional) validates a network professional's ability to implement and operate Cisco Unified Communications solutions in single site and multi-site deployments. Individuals who hold a CCVP certification can help create an IP telephony solution that is transparent, scalable, and manageable. The CCVP curriculum focuses on Cisco Unified CallManager, voice gateways and gatekeepers, switches, Unified IP Phones, and skills to secure the voice communications and ensure voice quality of service. Candidates must pass five secure, proctored certification-quality exams to become a Cisco Certified Voice Professional (CCVP). The prerequisite for the Cisco Certified Voice Professional (CCVP) is CCNA. As voice technologies continue to grow in importance Cisco helps you keep pace with a rapidly changing world.

The CCVP provides network professionals with the knowledge, skills, and credentials necessary to design and implement end-to-end Cisco IP Telephony solutions. The certification content focuses on Cisco Call Manager, QoS, gateways, gatekeepers, IP phones, voice applications and utilities on Cisco routers and Catalyst switches..

Required Exam(s)	Recommended Training
642-642 QoS	Quality of Service (QoS)
642-432 CVOICE	Cisco Voice over IP (CVOICE) AND Cisco Voice over IP Fundamentals (CVF)
642-426 TUC	Troubleshooting Cisco Unified Communications Systems (TUC)
642-444 CIPT 4.1 OR 642-445 CIPT 5.0	Cisco IP Telephony Part 1 (CIPT1 4.1) AND Cisco IP Telephony Part 2 (CIPT2 4.1) OR Cisco IP Telephony Part 1 (CIPT1 5.0) AND Cisco IP Telephony Part 2 (CIPT2 5.0)
642-453 GWGK	Implementing Cisco Voice Gateways and Gatekeepers (GWGK)

So the steps that I should take are Cisco Voice Fundamentals, a 6-hour e-learning course to establish a strong foundation in VoIP and PSTN fundamentals, then CVOICE, followed by CIPT1, GWGK, or QoS in any order. CIPT2 should be completed only after completing CIPT1.

The final course in the series, IP Telephony Troubleshooting is best taken after having completed the other CCVP courses in the curriculum. Those will help me better understand how to install, configure and deploy the various voice products and technologies tested in the CCVP exams.

From the information that I got, there will be 5 exams to pass:

• CVOICE is the foundation for it all (the most useful exam)
• QoS is not much based on anything; but you need to have some background so it is better after CVOICE (the most interesting one)
• GWGK is CVOICE2 (the hardest to master)
• TUC has some parts built on CIPT knowledge (the easiest to do)
• CIPT is a lot better after GWGK (the most boring)

I will think about CCVP and CCNP, which one should I choose for now...

Voice over ATM - signaling, addressing, routing, delay

Voice over ATM

The ATM Forum and the ITU have specified different classes of services to represent different possible traffic types for VoATM.

Designed primarily for voice communications, constant bit rate (CBR) and variable bit rate (VBR) classes have provisions for passing real-time traffic and are suitable for guaranteeing a certain level of service. CBR, in particular, allows the amount of bandwidth, end-to-end delay, and delay variation to be specified during the call setup.

Designed principally for bursty traffic, unspecified bit rate (UBR) and available bit rate (ABR) are more suitable for data applications. UBR, in particular, makes no guarantees about the delivery of the data traffic.

The method of transporting voice channels through an ATM network depends on the nature of the traffic. Different ATM adaptation types have been developed for different traffic types, each with its benefits and detriments. ATM adaptation layer 1 (AAL1) is the most common adaptation layer used with CBR services.

Unstructured AAL1 takes a continuous bit stream and places it within ATM cells. This is a common method of supporting a full E1 byte stream from end to end. The problem with this approach is that a full E1 may be sent, regardless of the actual number of voice channels in use. (An EI is a wide-area digital transmission scheme used predominantly in Europe that carries data at a rate of 2.048 Mbps.)

Structured AAL1 contains a pointer in the payload that allows the digital signal level 0 (DS0) structure to be maintained in subsequent cells. This allows network efficiencies to be gained by not using bandwidth for unused DS0s. (A DS0 is a framing specification used in transmitting digital signals over a single channel at 64 kbps on a T1 facility.)

The remapping option allows the ATM network to terminate structured AAL1 cells and remap DS0s to the proper destinations. This eliminates the need for permanent virtual circuits (PVCs) between every possible source/destination combination. The major difference from the previous approach is that a PVC is not built across the network from edge to edge.

VoATM Signaling

Figure 19-2 describes the transport method, in which voice signaling is carried through the network transparently. PVCs are created for both signaling and voice transport. First, a signaling message is carried transparently over the signaling PVC from end station to end station. Second, coordination between the end systems allows the selection of a PVC to carry the voice communication between end stations.

Figure 19-2 The VoATM Signaling Transport Model Describes the Transport Method, in Which Voice Signaling Is Carried Through the Network Transparently

At no time is the ATM network participating in the interpretation of the signaling that takes place between end stations. However, as a value-added feature, some products are capable of understanding channel associated signaling (CAS) and can prevent the sending of empty voice cells when the end stations are on-hook.

Figure 19-3 shows the translate model. In this model, the ATM network interprets the signaling from both non-ATM and ATM network devices. PVCs are created between the end stations and the ATM network. This contrasts with the previous model, in which the PVCs are carried transparently across the network.

Figure 19-3 In the VoATM Signaling Translate Model, the ATM Network Interprets the Signaling from Both Non-ATM and ATM Network Devices

A signaling request from an end station causes the ATM network to create an SVC with the appropriate QoS to the desired end station. The creation of an SVC versus the prior establishment of PVCs is clearly more advantageous for three reasons:

•SVCs are more efficient users of bandwidth than PVCs.

•QoS for connections do not need to be constant, as with PVCs.

•The capability to switch calls within the network can lead to the elimination of the tandem private branch exchange (PBX) and potentially the edge PBX. (A PBX is a digital or analog telephone switchboard located on the subscriber premises and used to connect private and public telephone networks.)

VoATM Addressing

ATM standards support both private and public addressing schemes. Both schemes involve addresses that are 20 bytes in length (shown in Figure 19-4).

Figure 19-4 ATM Supports a 20-Byte Addressing Format

The Authority and Format Identifier (AFI) identifies the particular addressing format employed. Three identifiers are currently specified: data country code (DCC), international code designator (ICD), and E.164. Each is administered by a standards body. The second part of the address is the initial domain identifier (IDI). This address uniquely identifies the customer's network. The E.164 scheme has a longer IDI that corresponds to the 15-digit ISDN network number. The final portion, the domain-specific part (DSP), identifies logical groupings and ATM end stations.

In a transport model, you don't need to be aware of the underlying addressing used by the voice network. However, in the translate model, the capability to communicate from a non-ATM network device to an ATM network device implies a level of address mapping. Fortunately, ATM supports the E.164 addressing scheme, which is employed by telephone networks throughout the world.

VoATM Routing

ATM uses a private network-to-network interface (PNNI), a hierarchical link-state routing protocol that is scalable for global usage. In addition to determining reachability and routing within an ATM network, it is also capable of call setup.

A virtual circuit (VC) call request causes a connection with certain QoS requirements to be requested through the ATM network. The route through the network is determined by the source ATM switch based on what it determines is the best path through the network, based on the PNNI protocol and the QoS request. Each switch along the path is checked to determine whether it has the appropriate resources for the connection.

When the connection is established, voice traffic flows between end stations as if a leased line existed between the two. This specification spells out routing in private networks. Within carrier networks, the switch-to-switch protocol is B-ICI. Current research and development of integrated non-ATM and ATM routing will yield new capabilities to build translate-level voice and ATM networks.

VoATM and Delay

ATM has several mechanisms for controlling delay and delay variation. The QoS capabilities of ATM allow the specific request of constant bit rate traffic with bandwidth and delay variation guarantees. The use of VC queues allows each traffic stream to be treated uniquely. Priority can be given for the transmission of voice traffic. The use of small, fixed-size cells reduces queuing delay and the delay variation associated with variable-sized packets.

Voice over ATM

Voice over ATM (VoATM) can be supported as standard pulse code modulated (PCM) voice via circuit emulation (AAL1, described later) or as variable bit rate voice in ATM cells as AAL2 (also described later). ATM offers many advantages for transport and switching of voice. First, quality of service (QoS) guarantees can be specified by service provisioning or on a per-call basis. In addition, call setup signaling for ATM switched virtual circuits (SVCs), Q.2931, is based on call setup signaling for voice ISDN, Q.931. Administration is similar to circuit-based voice networks.

However, VoATM suffers from the burden of additional complexity and incomplete support and interoperability among vendors. It also tends to be more expensive because it is oriented toward all optical networks. Most importantly, ATM is typically deployed
as a WAN Layer 2 protocol and therefore does not extend all the way to the desktop. Nevertheless, ATM is quite effective for providing trunking and tandem switching services between existing voice switches and PBXs.

Voice over Frame Relay (VoFR) has become widely deployed across many networks. Like VoATM, it is typically employed as a tie trunk or tandem-switching function between remote PBXs. It benefits from much simpler administration and relatively lower cost than VoATM, especially when deployed over a private WAN network. It also scales more economically than VoATM, supporting links from T1 down to 56 kbps. When deployed over a carefully engineered Frame Relay network, VoFR works very well and provides good quality. However, voice quality over Frame Relay can suffer depending on network latency and jitter. Although minimal bandwidth and burstiness are routinely contracted, latency and jitter are often not included in service level agreements (SLAs) with service providers. As a result, voice performance can vary. Even if quality is good at first, voice quality can degrade over time as a service provider's network becomes saturated with more traffic. For this reason, many large enterprise customers are beginning to specify latency and jitter, as well as overall packet throughput from carriers. In these situations, voice over Frame Relay can provide excellent service.

Voice over IP (VoIP) has begun to be deployed in recent years as well. Unlike voice over Frame Relay and Voice over ATM, Voice over IP is a Layer 3 solution, and it offers much more value and utility because IP goes all the way to the desktop. This means that in addition to providing basic tie trunk and tandem-switching functions to PBXs, VoIP can actually begin to replace those PBXs as an application. As a Layer 3 solution, VoIP is routable and can be carried transparently over any type of network infrastructure, including both Frame Relay and ATM. Of all the packet voice technologies, VoIP has perhaps the most difficult time supporting voice quality because QoS cannot be guaranteed. Normal applications such as TCP running on IP are insensitive to latency but must retransmit lost packets due to collisions or congestion. Voice is much more sensitive to packet delay than packet loss. In addition to normal traffic congestion, QoS for VoIP is often dependent on lower layers that are ignorant of the voice traffic mingled with the data traffic.

Voice Networking

Basic voice technology has been available for more than 100 years. During that time, the technology has matured to the point at which it has become ubiquitous and largely invisible to most users. This legacy of slow evolution continues to affect today's advanced voice networks in many ways, so it is important to understand the fundamentals of traditional voice technology before emulating it on data networks.

Traditional analog telephone instruments used for plain old telephone service (POTS) use a simple two-wire interface to the network. They rely on an internal two-wire/four-wire hybrid circuit to combine both transmit and receive signals. This economical approach has been effective but requires special engineering regarding echo.

Basic Telephony

Three types of signaling are required for traditional telephony: supervision, alerting, and addressing. Supervision monitors the state of the instrument—for example, allowing the central office or PBX to know when the receiver has been picked up to make a call, or when a call is terminated. Alerting concerns the notification of a user that a call is present (ringing) or simple call progress tones during a call (such as busy, ringback, and so on). Finally, addressing enables the user to dial a specific extension.

In addition to signaling, telephony services also provide secure media transport for the voice itself, analog-to-digital conversion, bonding and grounding for safety, power, and a variety of other functions when needed.

Analog voice interfaces have evolved over the years to provide for these basic functions while addressing specific applications. Because basic POTS two-wire analog interfaces operate in a master/slave model, two basic types of analog interfaces are necessary for data equipment to emulate: the user side and the network side. The user side (telephone) expects to receive power from the network as well as supervision.

A foreign exchange service (FXS) interface is used to connect an analog telephone, fax machine, modem, or any other device that would be connected to a phone line. It outputs 48 vdc power, ringing, and so on, and it accepts dialed digits. The opposite of an FXS interface is a foreign exchange office (FXO) interface. It is used to connect to a switching system providing services and supervision, and it expects the switch to provide supervision and other elements. (Why "foreign"? The terms FXS and FXO were originally used within telephone company networks to describe provision of telephone service from a central office other than normally assigned.)

Within FXS and FXO interfaces, it is also necessary to emulate variants in supervision. Typical telephones operate in a loop start mode. The telephone normally presents a high impedance between the two wires. When the receiver goes off-hook, a low-impedance closed circuit is created between the two wires. The switch, sensing current flow, then knows that the receiver is off-hook and applies a dial tone. The switch also checks to be sure that the receiver is on-hook before sending a ringing signal. This system works well for simple telephones, but it can cause problems on trunks between PBXs and COs with high activity. In that situation, the remote end and the CO switch can both try to seize the line at the same time. This situation, called glare, can freeze the trunk until one side releases it. The solution is to short tip or ring to ground as a signal for line seizure rather than looping it. This is called ground start.

After the line is seized, it is necessary to dial the number. Normal human fingers cannot outrun the dial receivers in a modern switch, but digits dialed by a PBX can. In that case, many analog trunks use a delay start or wink start method to notify the calling device when the switch is ready to accept digits.

Another analog interface often used for trunking is E&M. This is a four- or six-wire interface that includes separate wires for supervision in addition to the voice pair. E&M stands for "ear and mouth" or "Earth and magneto" and is derived from the early telephony days. The E&M leads are used to signal on-hook and off-hook states.

Analog voice works well for basic trunk connections between switches or PBXs, but it is uneconomical when the number of connections exceeds six to eight circuits. At that point, it is usually more efficient to use digital trunks. In North America, the T1 (1.544 Mbps) trunk speed is used, consisting of 24 digitized analog voice conversations. In other parts of the world, E1 (2.048 Mbps) is used to carry 30 voice channels. (Engineers refer to the adoption of E1 and T1 internationally as "the baseball rule"—there is a strong correlation of countries that play baseball to the use of T1. Therefore, the United States, Canada, and Japan have the largest T1 networks, while other countries use E1.)

The first step in conversion to digital is sampling. The Nyquist theorem states that the sampling frequency should be twice the rate of the highest desired frequency. Early telephony engineers decided that a range of 4000 hertz would be sufficient to capture human voices (which matches the performance of long analog loops). Therefore, voice channels are sampled at a rate of 8000 times per second, or once every 125 ms. Each of these samples consists of an 8-bit measurement, for a total of 64000 bits per second to be transmitted. As a final step, companding is used to provide greater accuracy of low-amplitude components. In North America, this is u-law (mu-law), while elsewhere it is typically A-law. For international interworking purposes, it is agreed that the North American side will make the conversion.

To construct a T1, 24 channels are assembled for a total of 1.536 Mbps, and an additional 8 bits are added every 125 ms for framing, resulting in a rate of 1.544 Mbps. Often, T1 frames are combined into larger structures called SuperFrames (12 frames) and Extended-SuperFrames (24 frames). Additional signaling can then be transmitted by "robbing bits" from the interior frames.

Basic T1 and E1 interfaces emulate a collection of analog voice trunks and use robbed bit signaling to transfer supervisory information similar to the E&M analog model. As such, each channel carries its own signaling, and the interface is called channel associated signaling (CAS). A more efficient method uses a common signaling channel for all the voice channels. Primary Rate Interface for ISDN is the most common example of this common channel signaling (CCS).

If voice/data integration is to be successful, all of these voice interfaces must be supported to provide the widest possible range of applications. Over the years, users have grown to expect a certain level of performance, reliability, and behavior of a telecommunications system, which must be supported going forward. All these issues have been solved by various packet voice systems today so that users can enjoy the same level of support to which they have become accustomed.

Voice/Data Integration Technologies - Advances in Applications

Advances in Applications

Real cost savings are sufficient for deployment of voice/data integration technologies. However, there are added benefits, which will become more evident in the future. As applications evolve, organizations will gain increased user productivity from the integration of voice and computer applications. Computer telephony integration (CTI) was begun by PBX vendors in the 1980s to integrate computers with PBXs to provide applications such as advanced call center features (for example, "screen pops" for agents).

However, as voice/data integration continues, the line between voice and data applications will continue to blur. For example, Unified Messaging systems are now available that combine voice mail, e-mail, and fax messaging into a single, convenient system. With these advanced systems, users can have e-mail read to them over the phone or can add document attachments to voice mail. At the enterprise level, new applications such as virtual call centers allow call center agents to be distributed anywhere within reach of the data network, while still receiving the full suite of call center functions and features. They can even receive calls over their computers rather than using a traditional telephone instrument, and they can provide "blended contact center" support to answer Web user questions with electronic chat capability and e-mail between voice calls. These capabilities go far beyond simple cost savings and will ultimately make organizations much more effective and profitable.

The strong pressures driving the integration of voice and data networks have resulted in various solutions to the problem, each with its own strengths and weaknesses. Three general approaches exist:

•Voice over ATM

•Voice over Frame Relay

•Voice over IP

There are also mixed solutions, including voice over IP, over Frame Relay, and so on. These are illustrated in Figure 19-1 The figure shows that voice over ATM and voice over Frame Relay are primarily transport mechanisms between PBXs, while voice over IP can connect all the way to the desktop. More details are available later in this chapter.

Figure 19-1 Mixed Solutions Including Voice over IP, Voice over Frame Relay, and so on.

Voice/Data Integration Technologies - Introduction

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Introduction

Voice/data integration is important to network designers of both service providers and enterprise. Service providers are attracted by the lower-cost model—the cost of packet voice is currently estimated to be only 20 to 50 percent of the cost of a traditional circuit-based voice network. Likewise, enterprise network designers are interested in direct cost savings associated with toll-bypass and tandem switching. Both are also interested in so-called "soft savings" associated with reduced maintenance costs and more efficient network control and management. Finally, packet-based voice systems offer access to newly enhanced services such as Unified Messaging and application control. These, in turn, promise to increase the productivity of users and differentiate services.

Integration of voice and data technologies has accelerated rapidly in recent years because of both supply- and demand-side interactions. On the demand side, customers are leveraging investment in network infrastructure to take advantage of integrated applications such as voice applications. On the supply side, vendors have been able to take advantage of breakthroughs in many areas, including standards, technology, and network performance.

Standards

Many standards for interoperability for voice signaling have finally been ratified and matured to the point of reasonable interoperability. This reduces the risk and costs faced by vendors offering components of a voice/data system, and it also reduces the risk to consumers. Standards such as H.323 (approved by the ITU in June 1996), are now evolving through their third and fourth iterations, while products based on initial standards still enjoy strong capabilities and interoperability. The general maturity of standards has in turn generated robust protocol stacks that can be purchased "off the shelf" by vendors, further ensuring interoperability.

Technology

Recent advances in technology have also enabled voice integration with data. For example, new Digital Signal Processor (DSP) technology has allowed analog signals to be processed in the digital domain, which was difficult or impossible only a few years earlier. These powerful new chips offer tremendous processing speeds, allowing voice to be sampled, digitized, and compressed in real time. Further breakthroughs in the technology allow as many as four voice conversations to be managed at the same time on a single chip, with even greater performance in development. These technologies greatly reduce the cost and complexity of developing products and deploying voice over data solutions.

In other areas, the industry has also enjoyed breakthroughs in voice codec (coder/decoder) technology. Previously, it was assumed that voice quality would suffer as bandwidth was decreased in a relatively linear fashion. However, new, sophisticated algorithms employed in new codecs have changed that view. It is now possible to obtain reasonably good-sounding voice at a fraction of the bandwidth once required. More importantly, these new algorithms have been incorporated into the standards to allow interoperability of highly compressed voice.

Network Performance

Finally, data-networking technology has improved to the point that voice can be carried reliably. Over the last few years, growth in voice traffic has been relatively small, while data traffic has grown exponentially. The result is that data traffic is now greater than voice traffic in many networks. In addition, the relative importance of data traffic has grown, as businesses and organizations come to base more business practices and policies on the ubiquity of data networks. This increase in importance of data networks has forced a fundamental change in the way data networks are engineered, built, and managed. Typical "best-effort" data modeling has given way to advanced policy-based networking with managed quality of service to support an even greater range of applications. Voice traffic, as an application on a data network, has benefited greatly from these technologies. For example, support of delay-sensitive SNA traffic over IP networks resulted in breakthroughs in latency management and queuing prioritization, which was then applied to voice traffic.

As stated previously, deployment of new technologies and applications must also be driven by greater demand from users. Breakthroughs in technology don't necessarily result in increased deployment unless they fill a real user need at a reasonable cost. For example, digital audio tape (DAT) technologies never enjoyed widespread use outside the audiophile community because of the high cost and only marginally better perceived performance than analog tapes. Voice/data integration, however, provides users with very real benefits, both now and in the future. Most users of voice/data integration technologies gain in two ways: Packet voice technologies are less expensive, and, in the future, they will offer much greater capabilities compared to today's circuit-based voice systems.

Economic Advantages

It has been estimated that packet voice networking costs only 20 to 30 percent of an equivalent circuit-based voice network. This is true for both carriers (service providers) and enterprise (private) users. Logically, this implies that enterprise users can operate long-distance voice services between facilities at less cost than purchasing long-distance voice services from a carrier, and it's often true. For example, many enterprise users have deployed integrated voice/data technologies to transport voice over data wide-area networks (WANs) between traditional PBXs across different geographical locations. The resulting savings in long-distance toll charges often provide payback in as little as six months (especially if international calls are avoided). Using data systems to carry voice as "virtual tie lines" between switches is also useful to service providers. In fact, many new carriers have started to embrace packet-based voice technologies as their primary network infrastructure strategy going forward.

However, savings associated with packet voice technologies don't stop with simple transport. It is also possible to switch voice calls in the data domain more economically than traditional circuit-based voice switches. For large, multisite enterprises, the savings result from using the data network to act as a "tandem switch" to route voice calls between PBXs on a call-by-call basis. The resulting voice network structure is simpler to administer and uses a robust, nonblocking switching fabric made up of data systems at its core.