Introducing QoS 3.1.2 Quality Issues in Converged Networks With inadequate network configuration, voice transmission is irregular or unintelligible. Gaps in speech where pieces of speech are interspersed with silence are particularly troublesome. Delay causes poor caller interactivity, which can cause echo and talker overlap. Echo is the effect of the signal reflecting the voice of the speaker from the far-end telephone equipment back into the ear of the speaker. Talker overlap is caused when one-way delay becomes greater than 250 ms. When this long delay occurs; one talker steps in on the speech of the other talker. The worst-case result of delay is a disconnected call. If there are long gaps in speech, the parties will hang up. If there are signaling problems, calls are disconnected. Such events are unacceptable in voice communications, yet are quite common with an inadequately prepared data network that is attempting to carry voice. Converged enterprise networks face four major issues :

Bandwidth capacity: Large graphics files, multimedia uses, and increasing use of voice and video cause bandwidth capacity problems over data networks.
End-to-end delay (both fixed and variable): Delay is the time it takes for a packet to reach the receiving endpoint after being transmitted from the sending endpoint. This period of time is called the “end-to-end delay” and consists of two components:
- Fixed network delay: Two types of fixed network delay are serialization and propagation delays. Serialization is the process of placing bits on the circuit. The higher the circuit speed, the less time it takes to place the bits on the circuit. Therefore, the higher the speed of the link, the less serialization delay is incurred. Propagation delay is the time it takes frames to transit the physical media.
- Variable network delay: Processing delay is a type of variable delay and is the time required by a networking device to look up the route, change the header, and complete other switching tasks. In some cases, the packet must also be manipulated, as, for example, when the encapsulation type or the hop count must be changed. Each of these steps can contribute to processing delay.
Variation of delay (also called jitter): Jitter is the delta, or difference, in the total end-to-end delay values of two voice packets in the voice flow.
Packet loss: WAN congestion is the usual cause of packet loss. In the case of voice traffic, packet loss causes dropped speech. Conversations are difficult to follow and communications are confused. If one talker tries to accommodate for the lost speech by repeating the speech, the result on the receiving end can sound like stuttering.

Content 3.1 Introducing QoS 3.1.3 Measuring Available Bandwidth Figure shows an empty network with four hops between a server and a client. Each hop uses different media with different bandwidths. The maximum available bandwidth is equal to the bandwidth of the slowest link as follows: Bandwidthmax = min (10 Mbps, 256 kbps, 512 kbps, 100 Mbps) = 256 kbps The calculation of the available bandwidth, however, is much more complex in cases where multiple flows are traversing the network. An IP flow is a unidirectional series of IP packets of a given protocol traveling between a source and a destination within a certain period. When there are multiple flows, use this formula to calculate average bandwidth available per flow as follows: Bandwidthavail = Bandwidthmax / flows Inadequate bandwidth can have performance impacts on network applications, especially those that are time-sensitive (such as voice) or consume a lot of bandwidth (such as videoconferencing). These performance impacts result in poor voice and video quality. In addition, interactive network services, such as terminal services and remote desktops, may suffer from lower bandwidth, which results in slow application response.

Content 3.1 Introducing QoS 3.1.4 Increasing Available Bandwidth Bandwidth is one of the key factors that affect QoS in a network; the more bandwidth there is, the better the QoS will be. However, simply increasing bandwidth will not necessarily solve all congestion and flow problems. Intuitively, the easiest way to increase bandwidth would seem to be to increase the link capacity of the network to accommodate all applications and users, allowing extra, spare bandwidth. Although this solution sounds simple, increasing bandwidth is expensive and takes time to implement. There are often technological limitations in upgrading to a higher bandwidth. In any event, ignoring QoS in favor of increasing bandwidth is at best a temporary fix. The faster the network, the faster traffic will increase and the problems return. Figure illustrates a more rational approach of using advanced queuing and compression techniques. Queuing means to classify traffic into QoS classes and then prioritize each class according to its relative importance. The basic queuing mechanism is first-in, first-out (FIFO). Other queuing mechanisms provide additional granularity to serve voice and business-critical traffic. Such traffic types should receive sufficient bandwidth to support their application requirements. Voice traffic should receive prioritized forwarding, and the least important traffic should receive the unallocated bandwidth that remains after prioritized traffic is accommodated. Cisco IOS QoS software provides a variety of mechanisms that can be used to assign bandwidth priority to specific classes of traffic:

FIFO
Priority queuing (PQ) or custom queuing (CQ)
Modified deficit round robin (MDRR)
Distributed type of service (ToS)-based and QoS group-based weighted fair queuing (WFQ)
Class-based weighted fair queuing (CBWFQ)
Low-latency queuing (LLQ)

A way to increase the available link bandwidth is to optimize link usage by compressing the payload of frames (virtually). Compression, however, also increases delay because of the complexity of compression algorithms. Using hardware compression can accelerate packet payload compressions. Stacker and Predictor are two compression algorithms that are available in Cisco IOS software. Another mechanism that is used for link bandwidth efficiency is header compression. Header compression is especially effective in networks where most packets carry small amounts of data (that is, where the payload-to-header ratio is small). Typical examples of header compression are TCP header compression and Real-Time Transport Protocol (RTP) header compression. Note
Payload compression is always end-to-end compression, and header compression is hop-by-hop compression. Example: Using Available Bandwidth More Efficiently
In a network with remote sites that use interactive traffic and voice for daily business, bandwidth availability is an issue. In some regions, broadband bandwidth services are difficult to obtain or, in the worst case, are not available. This situation means that available bandwidth resources must be used efficiently. Advanced queuing techniques, such as CBWFQ or LLQ, and header compression mechanisms, such as TCP and RTP header compression, are needed to use the bandwidth much more efficiently. Figure shows an example of how to use bandwidth efficiently using advanced queuing and header compression mechanisms. In this scenario, a low-speed WAN link connects two office sites. Both sites are equipped with IP phones, PCs, and servers that run interactive applications, such as terminal services. Because the available bandwidth is limited, an appropriate strategy for efficient bandwidth use must be determined and implemented. Administrators must chose suitable queuing and compression mechanisms for the network based on the kind of traffic that is traversing the network. The example in Figure uses LLQ and RTP header compression to provide the optimal quality for voice traffic. CBWFQ and TCP header compression are effective for