Fundamentals of Packet Telephony Networks

The increased efficiency of packet networks (for example, VoIP networks) and the ability to statistically multiplex voice traffic with data packets allows companies to maximize their return on investment (ROI) in data network infrastructures. Multiplexing voice traffic with data traffic reduces the number of costly circuits dedicated to servicing voice applications.

As demand for voice services expands, it is important to understand the different requirements of voice and data traffic. Previously, voice and data networks were separate and could not impact each other. Today, it is necessary to determine the protocols available to control voice calls and ensure that data flows are not negatively impacted.

This section delves into the benefits of packet telephony networks and provides an overview of basic packet telephony operations. Additionally, the fundamental components of packet networks are introduced. Finally, as a design consideration, this section considers the fragile nature of voice packets.


Packet Telephony Components

The basic components of a packet voice network, as shown in Figure 1-12, include the following:

• IP phones Provide IP voice to the desktop.

• Gatekeeper Provides Call Admission Control (CAC), bandwidth control and management, address translation, and call routing.

• Gateway Provides translation between VoIP and non-VoIP networks, such as the PSTN. It also provides physical access for local analog and digital voice devices, such as telephones, fax machines, key sets, and PBXs.

• Multipoint control unit (MCU) Provides real-time connectivity for participants in multiple locations to attend the same videoconference or meeting.

• Call agent Provides call control for IP phones, CAC, bandwidth control and management, and address translation. The call agent also serves as a repository for call routing information.

• Application servers Provide services such as voice mail, unified messaging, or call center support.

• Videoconference station Provides access for end-user participation in videoconferencing. The videoconference station contains a video capture device for video input and a microphone for audio input. The user can view video streams and hear the audio that originates at a remote user station. Cisco targets its VT Advantage product at desktop videoconferencing applications.

Other components, such as software voice applications, interactive voice response (IVR) systems, and softphones, provide additional services to meet the needs of enterprise sites.


Call Control

Call control allows users to establish, maintain, and disconnect a voice flow across a network, as shown in Figure 1-13.


Although different protocols address call control in different ways, they all provide a common set of services. The following are the basic components of call control:

• Call setup Checks call-routing configuration to determine the destination of a call. The configuration specifies the bandwidth requirements for the call. When the bandwidth requirements are known, CAC determines if sufficient bandwidth is available to support the call. If bandwidth is available, call setup generates a setup message and sends it to the destination. If bandwidth is not available, call setup notifies the initiator by presenting a busy signal. Different call control protocols, such as H.323, Media Gateway Control Protocol (MGCP), and Session Initiation Protocol (SIP), define different sets of messages to be exchanged during setup.

• Call maintenance Tracks packet count, packet loss, and interarrival jitter, or delay, when the call is set up. Information passes to the voice-enabled devices to determine if connection quality is good or if it has deteriorated to the point where the call should be dropped.

• Call teardown Notifies voice-enabled devices to free resources and make them available for the next call when either side terminates a call.

From a design perspective, you can set up call control in either a distributed or centralized architecture. The following sections describe both types.


Distributed Call Control

Distributed call control, an example of which is shown in Figure 1-13, offers an environment where call control is handled by multiple components in the network. This approach to call control is possible where the voice-capable device is configured to support call control directly. This is the case with a voice gateway when protocols, such as H.323 or SIP, are enabled on the device. In Figure 1-14, each location contains a Cisco Unified CallManager cluster. Each cluster is capable of handling call processing. Therefore, the topology shown demonstrates one example of distributed call control.


Distributed call control enables the gateway to perform the following procedure:

1. Recognize the request for service
2. Process dialed digits
3. Route the call
4. Supervise the call
5. Terminate the call


Centralized Call Control

Centralized call control, an example of which is illustrated in Figure 1-15, allows an external device (call agent) to handle the signaling and call processing, leaving the gateway to translate audio signals into voice packets after call setup. The call agent is responsible for all aspects of signaling, thus instructing the gateways to send specific signals at specific times. Also, the centralized call control model can leverage Cisco's Survivable Remote Site Telephony (SRST) feature to provide redundancy in the event of a WAN outage by having the voice-enabled router at the remote site perform basic call processing functions. In the figure, a Cisco Unified CallManager cluster located at the Headquarters location is in charge of call control. Therefore, the topology shown demonstrates an example of centralized call control.


When the call is set up, the following occur:

• The voice path runs directly between the two gateways and does not involve the call agent.

• When either side terminates the call, the call agent signals the gateways to release resources and wait for another call.

The use of centralized call control devices is beneficial in several ways:

• It centralizes the configuration for call routing and CAC. In a large voice environment, centralization can be extremely beneficial.

• The call agent is the only device that needs the intelligence to understand and participate in call control functions. These call control functions enable the customer to purchase less expensive voice-gateway devices and point to a single device to handle call control.

MGCP is one example of a centralized call control model.


Real-Time Versus Best-Effort Traffic

Voice and data can share the same medium. However, their traffic characteristics differ widely: Voice is real-time traffic and data is typically sent as best-effort traffic.

Traditional telephony networks were designed for real-time voice transmission, and therefore they cater to the need for a constant voice flow over the connection. Resources are reserved end to end on a per-call basis and are not released until the call is terminated. These resources guarantee that voice flows in an orderly manner. Good voice quality depends on the capacity of the network to deliver voice with guaranteed delay and timing.

Traditional data networks were designed for best-effort packet transmission. Packet telephony networks transmit with no guarantee of delivery, delay, or timing. Data handling is effective in this scenario because upper-layer protocols, such as TCP, provide for reliable, although untimely, packet transmission. TCP trades delay for reliability. Data can typically tolerate a certain amount of delay and is not affected by interpacket jitter.

A well-engineered, end-to-end network is required when converging delay-sensitive traffic, such as VoIP, with best-effort data traffic. Fine-tuning the network to adequately support VoIP involves a series of protocols and features to improve quality of service (QoS). Because the IP network is, by default, best effort, steps must be taken to ensure proper behavior of both the real-time and best-effort traffic. Packet telephony networks succeed, in large part, based on the QoS parameters that are implemented network-wide.

Multiplexing

A two-wire analog local loop typically carries one call at a time. To make better use of wiring facilities, different multiplexing techniques have been implemented to enable two-wire or four-wire connections to carry multiple conversations at the same time.

Time-division multiplexing (TDM) is used extensively in telephony networks to carry multiple conversations concurrently across a four-wire path, as shown in Figure 1-10. TDM involves simultaneously transmitting multiple separate voice signals over one communications medium by quickly interleaving pieces of each signal, one after another. Information from each data channel is allocated bandwidth based on preassigned timeslots, regardless of whether there is data to transmit.

Frequency-division multiplexing (FDM), as illustrated in Figure 1-11, involves carrying multiple voice signals by allocating an individual frequency range to each call. FDM is typically used in analog connections, although its functionality is similar to that of TDM in digital connections. FDM is used in cable or digital subscriber line (DSL) connections to allow the simultaneous use of multiple channels over the same wire.


If you have cable television service at your home, the television channels are all carried (and multiplexed) over a single pair of wires. This includes both the audio signals and the video signals. All the channels are present on the cable wires all the time. When you select the channel you want to watch, your set-top cable tuner determines which channel is sent to your television.