managing interactive data traffic.
Content
3.1 Introducing QoS 3.1.5 Effects of
End-to-end Delay and Jitter End-to-end delay and jitter
have a severe quality impact on the network as follows:
- End-to-end delay is the sum of all types of delays.
- Each hop in the network has its own set of variable
processing and queuing delays, which can result in jitter.
Figure shows four types of delay: -
Processing delay: Processing delay is the time that it
takes for a router (or Layer 3 switch) to take the packet from
an input interface and put the packet into the output queue of
the output interface. The processing delay depends on the
following factors:
- CPU speed
- CPU use
- IP switching mode
- Router architecture
-
Configured features on both the input and output
interfaces
Note
Many high-end
routers or Layer 3 switches use advanced hardware architectures
that speed up packet processing and do not require the main CPU
to process the packets. - Queuing delay: Queuing
delay is the time that a packet resides in the output queue of
a router. Queuing delay depends on the number of packets that
are already in the queue and packet sizes. Queuing delay also
depends on the bandwidth of the interface and the queuing
mechanism.
- Serialization delay: Serialization
delay is the time that it takes to place a frame on the
physical medium for transport. This delay is typically
inversely proportional to the link bandwidth.
-
Propagation delay: Propagation delay is the time that it
takes for the packet to cross the link from one end to the
other. The time it takes for the packet to cross usually
depends on whether it is data, voice, or video that is being
transmitted. For example, satellite links produce the longest
propagation delay because of the distance to communications
satellites.
Figure summarizes the impact of delay
and jitter on the quality of networks. The International
Telecommunication Union (ITU) considers network delay for voice
applications in Recommendation G.114. This recommendation
defines three bands of one-way delay as shown in Figure . These
recommendations are intended for national telecom
administrations. Therefore, they are more stringent than would
normally be applied in private voice networks. When the
location and business needs of end users are well known to the
network designer, more delay can prove acceptable. For private
networks, 200 ms of delay is a reasonable goal and 250 ms a
limit.
Content 3.1 Introducing QoS
3.1.6 Reducing the Impact of Delay on Quality When
considering solutions to the delay problem, there are two
things to note: - Processing and queuing delays are
related to devices and are bound to the behavior of the
operating system.
- Propagation and serialization delays
are related to the media.
There are many ways to
reduce the delay at a router. Assuming that the router has
enough power to make forwarding decisions rapidly, the
following factors have the most influence on queuing and
serialization delays: - Average length of the
queue
- Average length of packets in the queue
- Link bandwidth
Figure illustrates how
administrators can accelerate packet dispatching for
delay-sensitive flows in the following ways:
- Increase link capacity: Sufficient bandwidth causes
queues to shrink so that packets do not wait long before
transmittal. Increasing bandwidth reduces serialization time.
This approach can be unrealistic because of the costs that are
associated with the upgrade.
- Prioritize
delay-sensitive packets: This approach can be more
cost-effective than increasing link capacity. WFQ, CBWFQ, and
LLQ can each serve certain queues first (this is a preemptive
way of servicing queues).
- Reprioritize
packets: In some cases, important packets need to be
reprioritized when they are entering or exiting a device. For
example, when packets leave a private network to transit an
Internet service provider (ISP) network, the ISP may require
that the packets be reprioritized.
- Compress
payload: Payload compression reduces the size of packets,
which virtually increases link bandwidth. Compressed packets
are smaller and take less time to transmit. Compression uses
complex algorithms that add delay. If you are using payload
compression to reduce delay, make sure that the time that is
needed to compress the payload does not negate the benefits of
having less data to transfer over the link.
- Use
header compression: Header compression is not as
CPU-intensive as payload compression. Header compression
reduces delay when used with other mechanisms. Header
compression is especially useful for voice packets that have a
bad payload-to-header ratio (relative large header in
comparison to the payload), which is improved by reducing the
header of the packet (RTP header compression).
By
minimizing delay, network administrators can also reduce jitter
(delay is more predictable than jitter and easier to reduce).
Figure shows examples of ways to increase bandwidth efficiency
in a network. In this scenario, an ISP providing QoS connects
the offices of the customer to each other. A low-speed link
(512 kbps) connects the branch office while a higher-speed link
(1024 kbps) connects the main office. The customer uses both IP
phones and TCP/IP-based applications to conduct daily business.
Because the branch office only has a bandwidth of 512 kbps, the
customer needs an appropriate QoS strategy to provide the
highest possible quality for voice and data traffic. In this
example, the customer needs to communicate with HTTP, FTP,
e-mail, and voice services in the main office. Because the
available bandwidth at the customer site is only 512 kbps, most
traffic, but especially voice traffic, would suffer from
end-to-end delays. In this example, the customer performs TCP
and RTP header compression, LLQ, and prioritization of the
various types of traffic. These mechanisms give voice traffic a
higher priority than HTTP or e-mail traffic. In addition to
these measures, the customer has chosen an ISP that supports
QoS in the backbone. The ISP performs reprioritization on the
customer's traffic, according to the QoS policy, so the traffic
streams arrive on time at the customer's main office. This
design guarantees that voice traffic has high priority and a
guaranteed bandwidth of 128 kbps, FTP and e-mail traffic
receive medium priority and a bandwidth of 256 kbps, and HTTP
traffic receives low priority and a bandwidth of 64 kbps.
Signaling and other management traffic uses the remaining 64
kbps.
Content 3.1 Introducing QoS
3.1.7 Packet Loss After delay, the next most
serious concern for networks is packet loss. Usually, packet
loss occurs when routers run out of buffer space for a
particular interface (output queue). Figure gives some examples
of the results of packet loss in a converged network. Figure
illustrates a full interface output queue, which causes newly
arriving packets to be dropped. The term that is used for such
drops is “output drop” or “tail drop” (packets are dropped at
the tail of the queue). Routers might also drop packets for
these other less common reasons: - Input queue
drop: The main CPU is busy and cannot process packets (the
input queue is full).
- Ignore: The router runs
out of buffer space.
- Overrun: The CPU is busy
and cannot assign a free buffer to the new packet.
- Frame errors: The hardware detects an error in a
frame; for example, cyclic redundancy checks (CRCs), runt, and
giant.
Content 3.1 Introducing
QoS 3.1.8 Congestion Management: Ways to
Prevent Packet Loss Packet loss is usually the result of
congestion on an interface. Most applications that use TCP
experience slowdown because TCP automatically adjusts to