Defining Voice Fundamentals

This section begins by defining voice over IP and considering why it is needed in today's corporate environment. Because voice packets are flowing across a data infrastructure, various protocols are required to set up, maintain, and tear down a call. This section defines several popular voice protocols, in addition to hardware components that make up a voice over IP network.

Defining VoIP

VoIP sends packetized voice over an IP network. Typically, the IP network serves as a data network as well, resulting in potential quality and security issues. Fortunately, Cisco offers a collection of quality of service (QoS) and security features to ensure the quality and security of voice transmissions.

The ability to transmit voice over an IP network (for example, the Internet) allows many corporate networks to readily interconnect their sites without purchasing dedicated leased lines between their sites or relying on the public switched telephone network (PSTN), which imposes charges for certain call types (for example, long distance and international calls).

With the advent of VoIP technology, some confusion has arisen around its associated nomenclature. For example, consider the terms VoIP and IP telephony. Both refer to sending voice across an IP network. However, the primary distinction revolves around the endpoints in use. For example, in a VoIP network, traditional analog or digital circuits connect into an IP network, typically through some sort of gateway. However, an IP telephony environment contains endpoints that natively communicate using IP.

To further illustrate the distinction between VoIP and IP telephony, consider Figure 9-1. In the top portion of the figure, the endpoints in the VoIP network are an analog phone (connected to an analog port on a gateway) and a private branch exchange (PBX) (connected to a digital port on a different gateway). Because neither of these endpoints natively speaks IP, the topology is considered a VoIP network. The bottom portion of Figure 9-1 shows a Cisco IP phone, which does natively communicate using IP. The Cisco IP phone registers with a Cisco Unified Communications Manager server, which makes call routing decisions on behalf of the Cisco IP phone. Therefore, the bottom topology in the figure is considered an IP telephony network. Realize, however, that some literature might use the terms VoIP and IP telephony interchangeably.

Key Topic

Figure 9-1 VoIP Versus IP Telephony


Analog Phone

Analog Phone




Analog Phone

IP Telephony

IP Phone


Switch w

IP Phone



Cisco Unified Communications Manager

Cisco Unified Communications Manager

The Need for VoIP

Originally, one of the primary business drivers for the adoption of VoIP was saving money on long distance calls. However, increased competition in the industry drove down the cost of long distance calls to the point that cost savings alone was insufficient motivation for migrating a PBX-centric telephony solution to a VoIP network. However, several other justifications exist for purchasing VoIP technology:

•— ■ Reduced recurring expenses: In many traditional PBX-centric networks, a digital T1

1 Topic circuit typically could carry either 23 or 24 simultaneous voice calls (based on the type of signaling being used). Specifically, a T1 usually had 23 or 24 channels available. Each channel had a bandwidth of 64 kbps and could handle one, and only one, phone call. However, VoIP networks often leverage coder/decoders (codecs) to compress voice. Each voice call consumes less than 64 kbps of bandwidth per call, thereby allowing additional simultaneous calls, as compared to traditional technology.

■ Adaptability: Because VoIP networks send voice traffic over an IP network, administrators have a high level of control over the voice traffic. Different customers could be granted access to different voice applications (for example, a messaging application or an interactive voice response [IVR] application).

■ Advanced functionality: VoIP and IP telephony networks can also offer advanced features, such as the following:

— Call routing: Existing routing protocols (for example, EIGRP and OSPF) could be used to provide rapid failover to a backup link if a primary network link failed. Additionally, calls could be routed over different network links based on link quality or the link's current traffic load.

— Messaging: A solution such as Cisco Unity could be used to provide a single repository for a variety of messaging types. For example, a Microsoft Exchange message store could be used to consolidate the storage of fax transmissions, e-mail messages, and voice mail. Then a user could, for example, call into a Cisco Unity system and have her email read to her via text-to-speech conversion.

— Call center solutions: Cisco offers a variety of solutions for call centers. For example, Cisco's Contact Center and Contact Center Express solutions can intelligently route incoming calls to appropriate call center agents. Also, because the call center would be using Cisco IP Phones, the phones can be geographically separated (for example, call center agents working from home).

— Security: If an attacker were to intercept and capture VoIP packets, he could potentially play them back to eavesdrop on a conversation. As another example, a user might enter her personal identification number (PIN) into a bank's IVR system, and the attacker might capture those packets. Attackers might also introduce rogue devices (for example, IP phones or call agent servers) into the network. Fortunately, Cisco offers a variety of technologies and best practices for hardening the security of a VoIP network.

— Customer-facing solutions: Some customers might prefer to interact via a chat interface or e-mail, as opposed to talking with a company's customer service representative. Because a VoIP network works over a data network, data network features such as chat and e-mail can be integrated into a customer's selection of contact options, thereby increasing the customer's level of satisfaction.

VoIP Network Components

Figure 9-2 illustrates components commonly found in a VoIP network. They are described in Table 9-2.

Figure 9-2 VoIP Network Components

Figure 9-2 VoIP Network Components




IP phone

IP phones use an Ethernet network connection to send and receive voice calls. Figure 9-3 shows an example of a Cisco IP Phone, the Cisco 7970G.

Call agent

Call agents replace many of the features previously provided by PBXs. For example, a call agent can be configured with rules that determine how calls are routed. Cisco Unified Communications Manager (UCM) is an example of a call agent.


Gateways can forward calls between different types of networks. For example, you could place a call from an IP phone in your office, through a gateway, and to the PSTN to call your home.


Gatekeepers can be thought of as the traffic cops of the WAN. For example, because bandwidth on a WAN typically is somewhat limited, a gatekeeper can monitor the available bandwidth. Then, when there is not enough bandwidth to support another voice call, the gatekeeper can deny future call attempts.

Multipoint Control Unit (MCU)

MCUs are useful for conference calling. In a conference call, you might have multiple people talking at the same time, and everyone on that conference call can hear them. It takes processing power to mix together these audio streams. MCUs provide that processing power. MCUs might contain digital signal processors (DSP), which are dedicated pieces of computer circuitry that can mix together those audio streams.

Application server

Application servers offer additional services, such as voice mail, to a VoIP environment.

Table 9-2 VoIP Components (Continued)

Table 9-2 VoIP Components (Continued)



Videoconference station

Videoconference stations are devices and/or software (such as Cisco Unified Video Advantage) that allow a calling or called party to view and/or transmit video as part of their telephone conversation.

Voice-enabled switch

A voice-enabled Cisco Catalyst switch can provide inline power to an attached Cisco IP Phone, eliminating the need for an external power supply connected to the IP Phone. Also, a voice-enabled switch can recognize voice frames arriving from the attached IP Phone and give those frames higher priority than other frames.

Figure 9-3 Cisco 7970G IP Phone

Figure 9-3 Cisco 7970G IP Phone

VoIP Protocols

To support communication among Cisco IP Phones, analog telephones, traditional PBXs, and the PSTN (as just a few examples), VoIP networks require a collection of protocols. Some protocols are signaling protocols (for example, H.323, MGCP, H.248, SIP, and SCCP) used to set up, maintain, and tear down a call. Other protocols are targeted at the actual voice packets (for example, RTP, SRTP, and RTCP) rather than signaling information. Table 9-3 describes some of the more common VoIP protocols.

Table 9-3 VoIP Protocols




H.323 is an ITU standard. Rather than being a single protocol, it is a suite of protocols. Beyond protocols, the H.323 standard also defines certain devices, such as VoIP gateways and gatekeepers. H.323 is considered a peer-to-peer protocol, because some H.323 devices can make their own call-routing decisions, as opposed to relying on an external database.


Originally developed by Cisco, Media Gateway Control Protocol (MGCP) is considered a client/server protocol. The client (for example, an analog port in a voice-enabled router) can communicate with a server (for example, a Cisco Unified Communications server) via a series of events and signals. The server could tell the client that in the event of an attached phone going off-hook, play the signal of dial tone to that phone.


Based on MGCP, the H.248 standard is also known as Megaco. Specifically, H.248 is a joint IETF and ITU standard. Although H.248 is similar to MGCP, it is more flexible in its support for gateways and applications.


Session Initiation Protocol (SIP), like H.323, is considered a peer-to-peer protocol. SIP is a very popular protocol to use in mixed-vendor environments, perhaps because of SIP's use of existing protocols, such as HTTP and SMTP.


Skinny Client Control Protocol (SCCP), which is often called skinny protocol, is a Cisco-proprietary signaling protocol. SCCP is often used for signaling between Cisco IP Phones and Cisco Unified Communications Manager servers. However, some Cisco gateways also support SCCP. SCCP is considered a client/server protocol, like MGCP and H.248.


Real-time Transport Protocol (RTP) carries the voice payload. Interestingly, although RTP is a Layer 4 protocol, it is encapsulated inside UDP (also a Layer 4 protocol). Although the UDP port numbers used can vary by vendor, in Cisco environments, RTP typically uses even UDP ports in the range 16,384 to 32,767.


RTP Control Protocol (RTCP) provides information about an RTP flow (for example, information about the quality of the call). In a Cisco environment, RTCP typically uses odd UDP ports in the range 16,384 to 32,767.


Secure RTP (SRTP) secures the transmission of voice via RTP. Specifically, SRTP adds encryption, authentication, integrity, and anti-replay mechanisms to voice traffic.

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