Content Overview When migrating to a Voice over IP (VoIP) network, all network requirements, including power and capacity planning, must be examined. In addition, congestion avoidance techniques should be implemented. This module highlights the basic issues and defines the initial steps to take to ensure a functional VoIP implementation.

Content 7.1 Planning for Implementation of Voice in a Campus 7.1.1 Converged Network Benefits The benefits of packet telephony versus circuit-switched telephony are as follows:

More efficient use of bandwidth and equipment: Traditional telephony networks use a 64-kbps channel for every voice call. Packet telephony shares bandwidth among multiple logical connections and offloads traffic volume from existing voice switches.
Lower costs for telephony network transmission: A substantial amount of equipment is needed to combine 64-kbps channels into high-speed links for transport across the network. Packet telephony statistically multiplexes voice traffic alongside data traffic. This consolidation represents substantial savings on capital equipment and operations costs.
Consolidated voice and data network expenses: Data networks that function as separate networks to voice networks become major traffic carriers. The underlying voice networks are converted to utilize the packet-switched architecture to create a single integrated communications network with a common switching and transmission system. The benefit is significant cost savings on network equipment and operations.
Increased revenues from new services: Packet telephony enables new integrated services, such as broadcast-quality audio, unified messaging, and real-time voice and data collaboration. These services increase employee productivity and profit margins well above those of basic voice services. In addition, these services enable companies and service providers to differentiate themselves and improve their market position.
Greater innovation in services: Unified communications use the IP infrastructure to consolidate communication methods that were previously independent; for example, fax, voice mail, e-mail, wireline telephones, cellular telephones, and the Web. The IP infrastructure provides users with a common method to access messages and initiate real-time communications—independent of time, location, or device.
Access to new communications devices: Packet technology can reach devices that are largely inaccessible to the time-division multiplexing (TDM) infrastructures of today. Examples of such devices are computers, wireless devices, household appliances, personal digital assistants, and cable set-top boxes. Intelligent access to such devices enables companies and service providers to increase the volume of communications they deliver, the breadth of services they offer, and the number of subscribers they serve. Packet technology, therefore, enables companies to market new devices, including videophones, multimedia terminals, and advanced IP phones.
Flexible new pricing structures: Companies and service providers with packet-switched networks can transform their service and pricing models. Because network bandwidth can be dynamically allocated, network usage no longer needs to be measured in minutes or distance. Dynamic allocation gives service providers the flexibility to meet the needs of their customers in ways that bring them the greatest benefits.

Content 7.1 Planning for Implementation of Voice in a Campus 7.1.2 VoIP Network Components The basic components of a VoIP network are:

IP phones: Provide IP voice to the desktop.
Gatekeeper: Provides connection admission control (CAC), bandwidth control and management, and address translation.
Gateway: Provides translation between VoIP and non-VoIP networks, such as the public switched telephone network (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.
Application servers: Provide services such as voice mail, unified messaging, and Cisco CallManager Attendant Console.
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.

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

Content 7.1 Planning for Implementation of Voice in a Campus 7.1.3 Traffic Characteristics of Voice and Data Voice traffic has extremely stringent quality of service (QoS) requirements. Voice traffic usually generates a smooth demand on bandwidth and has minimal impact on other traffic as long as voice traffic is managed.Although voice packets are typically small (60 to 120 bytes), they cannot tolerate delay or drops. The result of delays and drops is often unacceptable voice quality. Because drops cannot be tolerated, User Datagram Protocol (UDP) is used to package voice packets. TCP retransmit capabilities have no value. For voice quality, the delay should be no more than 150 ms (one-way requirement) and less than 1 percent packet loss. A typical voice call requires 17 to 106 kbps of guaranteed priority bandwidth, plus an additional 150 bps per call for voice-control traffic. Multiplying these bandwidth requirements by the maximum number of calls expected during the busiest time period indicates the overall bandwidth required for voice traffic. The QoS requirements for data traffic vary greatly. Different applications (for example, a human resources application versus an automated teller machine [ATM] application) may make greatly different demands on the network. Even different versions of the same application may have varying network traffic characteristics. Data traffic can demonstrate either smooth or bursty characteristics, and it differs from voice and video in terms of delay and drop sensitivity. Almost all data applications can tolerate some delay and generally can tolerate high drop rates. Because data traffic can tolerate drops, the retransmit capabilities of TCP become important and, as a result, many data applications use TCP. It is important to be able to identify different types of traffic that move over networks. With TCP/IP, most applications can be identified by their use of TCP or UDP port numbers, and with TCP, a stream of traffic usually occurs. However, some applications use dynamic port numbers that make classifications more difficult. Cisco IOS software supports network-based application recognition (NBAR), which can be used to recognize dynamic port applications.

Content 7.1 Planning for Implementation of Voice in a Campus 7.1.4 VoIP Call Flow VoIP calls can contend with normal client data for bandwidth. If both the client PC and the VoIP phone are on the same VLAN, each will try to use the available bandwidth without consideration of the other device. To avoid this issue, use two VLANs to allow separation of VoIP and client data. After data is separated, QoS can be applied to prioritize the VoIP traffic as it traverses the network.A major component of designing a successful IP telephony network is properly provisioning the network bandwidth. You can calculate the required bandwidth by adding the bandwidth requirements for each major application, including voice, video, and data. This sum represents the minimum bandwidth requirement for any given