Understanding Voice over IP

This section describes Voice over IP (VoIP), and it suggests why you should use VoIP for voice transmission.

Voice can be transmitted across a traditional telephony network in two forms: analog and digital.

A typical residential telephone is an analog device that converts your voice into an electrical current. The receiving telephone recreates the initial variation of amplitude and frequency. What the party at the other end of the line hears isn’t really the original speech; it's an analogy of the original speech. This is why the signal is called an analog signal.

Analog signaling can be represented by a continuously variable waveform (illustrated below), and can represent an infinite number of states. Because analog signals tend to degrade over distance, however, they need to be periodically amplified.

Waveform Analog Signal

In the early telephone network, when analog transmission was passed through amplifiers to boost the voice signal, the ambient line noise was amplified as well. Therefore, the sound quality of the transmission was degraded, often resulting in an unusable connection. Typical residential telephones are still based on analog technology.

In the 1970s, to improve quality of the received signal, add features, and handle increased call volume, the telephone network began migrating to digital-loop carrier technology, represented by T1, T3, E1, and E3. T1, used primarily in North America and Japan, is a 24-channel digital system that carries data at 1.544 Mbps through the telephone switching network. E1, used predominantly in Europe, carries data at a rate of 2.048 Mbps using 30 64-Kbps digital channels.

In comparing the telephone network to a data network, we could say that the telephone network is a very large circuit-switched network. More than any other network, however, the PSTN is ubiquitous—it exists everywhere, it is dependable and easy to use, and everyday life would be inconceivable without it.

Data networks, on the other hand, are privately owned and administered, so architecture, size, and complexity may vary. Data networks were likely originally designed for a different purpose other than to transport voice traffic, which is more sensitive to delay and can demand more bandwidth than data.

VoIP enables Cisco routers and switches to carry telephony-style voice traffic—that is, live, packetized voice traffic such as telephone calls—over IP-based data networks (intranetworks or internetworks) rather than the PSTN.

To configure an IP network to carry voice, you first need to make sure that the network is well-engineered end-to-end. Although good design practices are essential to VoIP, a complete discussion of network design is beyond the scope of this tutorial.

Cisco routers and switches are equipped to handle the origination, transport, and termination of VoIP traffic. When transmitting voice signals across a data network, the router on the network performs some of the same basic functions as a switch, which are as follows:

The analog signal of the originating call is converted into a digital signal in the digital signal processors (DSPs) in the router.
The voice signal is then segmented into frames, which are then coupled in groups of two and stored in voice packets that conform to the standard IP format.
The IP packets are sent over the Internet in compliance with ITU-T specification H.323, explained in the related document.

At the receiving end, this process is reversed, with the packets being converted to digital signaling, then back to analog to be received and heard on the other end of the call.

The above discussion is a very brief summary of issues relating to analog-to-digital conversion of voice over an IP network. This topic is covered in greater detail later in module 3. Compression techniques, CODECs, and mean opinion scores (MOS) and their relationship to voice quality are discussed in these related documents: Quality of Service Overview, Defining Analog Voice, Waveform Coding Techniques, CODECs, and Reducing Bandwidth Consumption, but are beyond the scope of this course. A future product in the CIM for Voice Internetworking series will cover Quality of Service (QoS) technology topics, configuration, and troubleshooting.

VoIP is IP-based. IP is considered a connectionless or best-effort transport when used in conjunction with the User Datagram Protocol (UDP). For this reason, UDP, which is datagram-based (or connectionless), best suits the specific requirements of voice traffic. UDP is preferred as the transport for VoIP, even though TCP, which is connection-oriented, is generally considered as a more ideal transport mechanism because of its built-in reliability.

So the voice packet to be sent over IP is split into two portions: (1) the control signaling and (2) the actual packetized voice traffic. The control signaling then runs on top of TCP, and the real-time voice is sent across UDP.

Consider the underlying reasons for using UDP as the transport for voice traffic:

Retransmission of dropped packets—the behavior in TCP—is far worse for delay-sensitive voice traffic than is packet loss.
Because UDP is stateless, it removes from the CPU the burden of overhead entailed in maintaining state on connection-oriented protocols such as TCP.
From an application perspective, VoIP uses small-sized packets that are sent out at consistent intervals depending on the DSP and CODEC used. The UDP header, which is 8 bytes long, is smaller in size than the TCP 20-byte header and thus costs less in bandwidth and overhead.

The following figure shows the VoIP packet.

VoIP Packet Structure

TCP offers reliability in that it guarantees retransmission of lost frames, so is used to carry the control signaling, but this reliable delivery is useless in the internetwork transportation of packetized voice because a frame that arrives late as a result of retransmission is as useful as no frame at all—that is, it has no effect. In other words, retransmission of packets is not meaningful. By the time the resent packet arrives at the end user endpoint, the required delivery time has long been transgressed.

Among the many reasons to use VoIP for voice traffic are the following:

Because IP is ubiquitous, it provides contiguous connectivity independent of the media transport that carries it.
Use of VoIP transcends the differences among various transport media and mechanisms because the media used is transparent to an IP infrastructure. The contiguous connectivity of IP offers an important benefit to real-time applications that is not available through direct use of other Cisco technologies, such as Voice over ATM (VoATM) or Voice over Frame Relay (VoFR).
VoIP traffic is easily integrated with traffic from modern applications such as unified messaging or virtual call centers.
As a technology for transporting voice calls, VoIP packet-switched traffic offers cost benefit over circuit-switched networks. One reason for this cost benefit is that Cisco IOS® IP-based networks are less expensive to build and maintain than are circuit-switched networks. Some details about the cost benefit are covered next, in the section "Benefits of Using IP to Transmit Voice."

Implementation of an Internet telephony network in a business can have many benefits. It enables toll bypass, remote PBX presence over WANs, unified voice and data trunking, and POTS-Internet telephony gateways. The primary benefits are significant cost savings because of the following:

Placing a Phone Call from a PC...

Sidebar:

Placing a Phone Call from a PC

You are at home in California and would like to call your grandmother in Boston. If your grandmother were a technology junkie, she could use Microsoft NetMeeting and you could chat via the Internet. Your grandmother doesn't even own a computer, however, so you have to use plain-old telephone service.

Or do you? If your ISP provided some flavor of packet telephony such as VoIP, you could access its network in one of two ways:

You can use your existing dialup connection and start an H.323-compatible application (Microsoft NetMeeting, for example) and tell your application that you want to use the gateway at your ISP. The ISP verifies who you are, permits access to its packet voice gateway, and you place a telephony call to grandma. Grandma doesn't know that you are placing a packetized voice call, because she receives the call on her telephone.
Your ISP might offer an 800-number service (similar to "1 800 COLLECT"). You dial the 800 number, enter your access code, and you have access to the packetized voice network.

In both of these scenarios, the ISP is actually a next-generation telephony company that is subject to all the laws and tariffs of standard telephony carriers.

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Elimination of leased lines—Companies have long relied on private leased-line networks to bypass public telephone charges, but rates applied to leased lines are also high.

Merging voice and data networks—You can eliminate the need to maintain two separate networks for voice and data.
Toll bypass—Save long-distance charges by replacing tie lines that currently hook up their PBX-to-PBX networks and route voice calls across their existing data infrastructure. Employ the concept of lowest-cost routing to bypass tolls on calls that originate on-network. A corporation can also use VoIP to replace smaller key systems at remote offices while maintaining more robust equipment at sites having greater need. A company can also use routers to route voice calls across its existing TCP/IP connections across LATA boundaries from one office to another.

The following figure illustrates ways to route internal-to-internal calls (calls that originate and terminate on the network) and internal-to-external calls (calls that originate on the network and terminate off the network).

How Toll Bypass in a Large Corporation Works

Lowest-Cost Path Routing...

Sidebar

Lowest-Cost Path Routing in a Corporate Network

The following illustration demonstrates how lowest-cost routing works to reduce long-distance costs in a voice-over IP network.

A Long-Distance Call Across Area Codes Is Automatically Routed Across the Network
via the Lowest-Cost Path

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Interoffice fax relay—Real-time fax relay can be used on an interoffice basis. A large percentage of long-distance minutes is fax traffic; in fact, up to 60 percent of long-distance minutes to Japan are faxes. Sending faxes over IP could potentially represent a significant savings for a business.

VoIP is primarily a software feature; however, to use this feature on a Cisco 3600 series router, you must install a voice network module (VNM). The VNM holds either one or two voice interface cards (VICs). Each two-port VIC is specific to a particular signaling type associated with a voice port type, such as FXS, FXO, or E&M analog voice ports.

A Cisco 3640 can house up to three modules with up to a total of six VICs that slide into the voice/fax network modules and provide the interface to the telephony equipment.

FXO, FXS, and E&M Interfaces

Foreign eXchange Office (FXO) and Foreign eXchange Station (FXS) are types of RJ-11 interfaces on Cisco routers that utilize either loop-start or ground-start analog signaling techniques:

FXS: An FXS interface is a standard analog telephone interface which, in essence, mimics the PSTN. It supports both loop-start and ground-start modes of operation, depending upon the configuration. An FXS interface can be used to connect basic devices such as phones, modems, and faxes and must provide voltage, ring generation, off-hook detection, and call progress indicators. The FXS ports on the Cisco 3600 series router can provide the battery at 24-volt DC.
FXO: An FXO interface is like the port on a standard telephone that receives calls from and dials into a PBX or other switched network. Because an FXO port communicates directly with the PSTN or a PBX, it requires that a dial tone, ring indication, and call progress indicators be provided to it. The FXO ports on the Cisco 3600 series router can also support both loop-start and ground-start operation. In addition to the software configuration of the router, for certain products a hardware jumper must be set in order to make FXO ports operate in ground-start mode.

E&M interfaces support E&M signaling types I through V, either two-wire or four-wire operation, and support wink-start, delay-start, and immediate-start signaling. E&M connections are frequently used to connect computer telephony connections to a switch, and are also used in trunking arrangements switch-to-switch and switch-to-network.

Two Views of the Cisco 3600 Series Routers

Sidebar:

Rules for Voice over Data Networks

There are some general restrictions to be aware of when designing a data network to handle voice calls. You need to account for the regulations of any involved telephone jurisdictions, such as LATAs. A company can employ a voice-over-data network anywhere a traditional leased-line, PBX-to-PBX tie line can legally be used. A tie-line network can be used as a model.

Within a telephony jurisdiction, it is generally acceptable for a business to use Internet telephony to support its own voice calling within its own sites, as long as the calls remain within the "user group." A user group consists of employees, contractors, and/or employees of a secondary business with which the original company has close ties.
Business-to-business calling over Internet telephony is usually acceptable, as long as the companies have a close business relationship and the calls remain within the user group.
It is normally acceptable to have calls originate from the PSTN, then traverse the network to a member of the user group or to the business into which the call was placed, as long as the call does not traverse into another country.
When an Internet telephony network is used to connect one public network phone to another public network phone, the Internet voice provider is generally seen as a telephony carrier, subject to restrictions and regulation of that telephony jurisdiction.
If the call originates from a PC-based application (such as Microsoft NetMeeting), some telephony jurisdictions currently regard this type of call as nontelephony, and not subject to regulation, even if the call somehow crosses over to the public network through a gateway of some sort.

These rules are merely guidelines of which you should be aware and they may be subject to change at any time. Always research any pertinent regulations before deploying an Internet telephony network at your company.

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Rules for Voice over Data Networks...

It is important to match the correct interface type with the signaling type required for the application in the VoIP network.

The examples in the first part of this tutorial use a Cisco 3640 router with FXS/FXO and a version of Cisco IOS software that supports VoIP. The next module covers E&M technology and configuration.

The VoIP gateway implementation on the Cisco 3600 series is a subset of the ITU-T standards. Contact the ITU-T for more details or to obtain copies of the standards documents.

International Telecommunications Union web site at http://www.itu.int/

Go on to Configuration of Basic Analog Voice over IP.