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