What You Will Learn
After reading this chapter, you should be able to:
Explain why QoS is critical on VoIP networks.
Distinguish between congestion management and congestion avoidance.
Describe how to limit the bandwidth used by specific traffic types.
Identify strategies for maximizing available WAN bandwidth for VoIP traffic.
Traditionally, networks physically separated voice, video, and data traffic. These traffic types literally flowed over separate media (for example, leased lines or fiber optic cables). Today, however, network designers leverage the power of existing data networks to transmit voice and video, thus achieving significant cost savings by reducing equipment, maintenance, and even staffing costs.
Today's converged networks present a challenge, however. Specifically, multiple applications contend for bandwidth, and some applications (for example, voice) are more intolerant of delay, sometimes called latency, than other applications, such as FTP file transfers. A lack of bandwidth overshadows most quality problems.
A lack of bandwidth might cause packets to suffer from one or more of the following symptoms:
Delay Delay is the time required for a packet to travel from its source to its destination. You might have witnessed delay on the evening news, when the news anchor is talking via satellite to a foreign news correspondent. Due to the satellite delay, the conversation begins to feel unnatural.
Jitter Jitter results from the uneven arrival of packets. Imagine a Voice over IP (VoIP) conversation where packet 1 arrives followed by packet 2 20 milliseconds (ms) later. After another 70 ms packet 3 arrives, and then packet 4 arrives 20 ms behind packet 3. This variation in arrival times, called jitter, does not result from dropped packets. However, the jitter might sound like dropped packets (that is, gaps in the speech) from the listener's perspective.
Drops Routers and switches contain buffers to store packets when the network link (that is, the physical network connection) lacks sufficient bandwidth to transmit the packets at the moment. Packet drops occur when a link is congested and a buffer overflows. Some traffic types, such as web traffic, retransmit dropped packets. However, dropped voice and video packets are gone forever. These traffic types, which use User Datagram Protocol (UDP) for transmission, lack the ability to retransmit dropped packets. UDP is still preferred over Transmission Control Protocol (TCP) for voice, however, due to UDP's reduced overhead. And if you think about it, you wouldn't want voice packets retransmitted anyway, because voice packets arriving out of order would sound like gibberish.
Fortunately, quality of service (QoS) features available on Cisco routers and switches can recognize important traffic and then treat that traffic in a special way. For example, you might wish to allocate 128 kbps of bandwidth for VoIP traffic, and optionally give that traffic priority treatment. At the same time, you may want your web traffic to receive 64 kbps of non-priority bandwidth.