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CHAPTER 1
INTRODUCTION
1.1 PREAMBLE
The Internet carries high quality multimedia traffic along with other
applications and services. Distributed multimedia consists of voice, video and
data. Video conferencing, video on demand (VoD), distant learning and
distributed games are some of its applications. Providing quality of service
(QoS) for multimedia streaming has been a difficult and challenging problem.
When networks carry all types of traffic it is important that the QoS
constraints are met wherever necessary, or when the users are prepared to pay
for premium services.
As more and more users and organisations use the Internet for their
multimedia applications, there is a significant rise in the research interest on
QoS. Xiao and Ni (1999) provide an overview of QoS in the Internet. The
various QoS parameters acceptable for applications are throughput, delay,
delay variation (jitter), loss and error rates. The most prevailing factor in the
degradation of service is the packet loss at the routers during congestion
(Shyu et al 2003).
There has been considerable amount of research work carried out to
maximize throughput and minimize loss rates. One such approach is the use
of active queue management at the IP routers to minimize loss. Another
approach is the design of the IP routers to include a special scheduler to
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reduce delay or jitter. This thesis proposes an integrated approach of a novel
active queue management (AQM) scheme and a new scheduler at the high
performance routers to support QoS. The AQM aims at increasing throughput
and reducing loss rate. The scheduler is designed to reduce jitter.
The Internet carries packets of various applications, with the basic
transport layer protocol of the Internet being the Transmission Control
Protocol (TCP). But due to the overhead such as retransmission and
acknowledgement techniques, TCP is not suitable for real time multimedia
flows. User Datagram Protocol (UDP) is the preferred protocol for these
applications (Chung and Claypool 2000). Hence alternative solutions are
needed to support distributed multimedia flows in the Internet. The QoS
constraints for these flows normally require throughput, delay and jitter as
important parameters. As the combination of throughput and one of the others
(packet loss, cost, delay, delay jitter) is NP complete (Wang and Crowcroft
1996) and jitter reduction is needed for the inelastic applications taken for this
work, jitter is taken as the second QoS parameter in this work.
This chapter provides a high-level overview of the IP networks and
the functioning of the IP routers. It describes the attempts made at providing
QoS and resource management to support distributed multimedia
applications. The chapter introduces traditional buffer management and
scheduling in these routers. It briefly presents the inspiration for AQM and
the scheduler. The various Internet models are described. The objectives of
this work and the motivation for a novel Internet QoS architecture are
explained. The primary contributions of this work also presented. The chapter
ends with a discussion on the organization of the rest of the thesis.
3
1.2 OVERVIEW OF IP ROUTERS
This section discusses the arrangement and the functioning of
today’s Internet and the way the IP routers function. It introduces the
organization of an IP router along with its components and describes the
operation of an IP router.
1.2.1 IP Routers
The Internet is bound by the basic concepts of Internet Protocol
(IP), addressing and routing (Keshav 1997). Routers are network layer
devices used to interconnect different networks (Peterson and Davie 2001).
Their primary role is to switch packets from input links to output links. In
order to do so a router must be able to determine the path that every incoming
packet needs to follow and decide which outgoing link to select (Kurose and
Ross 2003). They must also deal with heterogeneous link technologies,
provide scheduling support for differential service and anticipate in complex
distributed algorithms to generate globally coherent routing tables (Keshav
and Sharma 1998). These demands, along with a voracious need for
bandwidth requirement by applications, challenge their design.
Routers are found at every level in the Internet. Primarily there are
three types of routers:
a. backbone routers
b. enterprise routers
c. access routers
Routers in access networks allow homes and small businesses to
connect to an Internet Service Provider (ISP). Routers in enterprise networks
4
link thousands of computers within a campus or enterprise. Usually routers in
the backbone are not directly accessible to end-systems. Instead, they link
together ISPs and enterprise networks with long distance trunks.
1.2.1.1 Components of a router
Figure 1.1a abstracts the architecture of a generic router
(Tantawy 1994). A generic router has the basic functionalities that include
route processing, packet forwarding, traffic prioritization etc. A decentralized
router architecture has network interface that offers the processing power and
the buffer space needed for packet processing tasks related to the packets
flowing through it (Figure 1.1.b). Functional components process the
inbound, outbound traffic and time-critical port processing tasks such as
protocol functions that lie in the critical path of data flow and the QoS
processing functions. QoS guarantees are provided by classifying packets into
predefined service classes (Stallings 2002).
1.2.1.2 The IP packet processing steps
Figure 1.2 helps in understanding the packet processing done inside
a router. The IP packet processing steps are as follows (Keshav and
Sharma 1998) :
1. IP Header Validation: As a packet enters an input port (ingress
port), the forwarding logic verifies all Layer 3 information
(header length, packet length, protocol version, checksum, etc.).
2. Route Lookup and Header Processing: The router then performs
an IP address lookup using the packet’s destination address to
determine the output port (egress port) and performs all IP
forwarding operations (TTL decrement, header checksum, etc.).
5
Figure 1.1 A generic switch-based distributed router architecture
Switch Fabric
Network Interface
Network Interface
Route Processor
(CPU)
Switch Fabric Interface
Inbound Processing
Local Processing Subsystem
s
Outbound Processing
Media-Specific Interface
a) Functional diagram
Post Processing
(Route Resolution
Logic)
Queue Manager
Memory
MAC PHY
Network Interface
b) Generic Architecture
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Shared Memory
System Controller
Ingress Port
Ingress Port
Egress Port
Egress Port
5
9
1 2 3 4 8
Packets
Packets
Figure 1.2 IP packet processing in shared memory router architecture
6
7
Packets
Packets
7
3. Packet Classification: The forwarding engine examines
Layer 4 and higher layer packet attributes relative to any QoS
and access control policies.
4. With these attributes in hand, the router performs one or more
of the following parallel functions: associates the packet with
the suitable priority and the right egress port(s) , redirects the
packet to a different (overridden) destination (ICMP redirect),
drops the packet according to a congestion control policy
(e.g., RED), or a security policy and performs the appropriate
accounting functions (statistics collection, etc.).
5. The forwarding engine notifies the system controller about the
packet arrival.
6. The system controller allocates a memory location for the
arriving packet.
7. Once the packet has been passed to the shared memory, the
system controller signals the proper output port(s).
8. The output port(s) gets the packet from the known shared
memory location using any of the following algorithms:
eighted Fair Queueing (WFQ), Weighted Round-Robin
(WRR), Strict Priority (SP), etc.
9. When the packet is retrieved by the appropriate destination
outbound link(s) has retrieved the packet, it informs the
system controller and relinquishes the memory location for
new traffic.
8
1.2.1.3 Input vs. output queued routers
The routers need buffers to store the packets before an output line
becomes available. These buffers may be placed at the input ports or at the
output ports. This section looks at the pros and cons of each choice.
In a purely input-queued router, packets are queued at the input
buffer. An arbiter guarantees access to the output links by scheduling these
packets. Hence there is no need for an output queue. The advantage of the
input-queued approach is that the speedup of the switch fabric can improve
the performance of the router. The disadvantage is that if First-In-First-Out
(FIFO) order is used to serve the queue, the packet at the head of the queue
may block other packets though their output lines are free. This is known as
head-of-line blocking. Many researchers have suggested different algorithms
for overcoming this problem.
A pure output-queued router buffers packets only at the outputs. It
uses any of the scheduling policies to send the packets through the out links.
The major advantage is that this approach does not suffer from the head-of-
blocking problem. But, if all the incoming packets are destined for the
same output link, then there is a need for input buffers, to avoid packet loss
(Keshav 1997). This gives rise to the hybrid approach of having input and
output buffers.
Thus, a combination of input buffered and output buffered switch is
required, i.e., CIOB (Combined Input and Output Buffered). The goal of most
designs, then, is to find the minimum speedup required to match the
performance of an output buffered switch using a CIOB and Virtual Output
Queues (VOQs).
9
An input port provides several functions. It carries out datalink
layer encapsulation and decapsulation. It may also have the intelligence to
look up an incoming packet’s destination address in its forwarding table to
determine its destination port (this is also called route lookup). The algorithm
for route lookup can be implemented using custom hardware, or each line
card may be equipped with a general-purpose processor. In order to provide
QoS guarantees, a port may need to classify packets into predefined service
classes. Output ports store packets before they are transmitted on the output
link. They can implement sophisticated scheduling algorithms to support
priorities and guarantees. Like input ports, output ports also need to support
datalink layer encapsulation and decapsulation and a variety of higher-level
protocols (Keshav and Sharma 1998).
Input-queued and output-queued routers share the route lookup
bottleneck, but each of them has an additional performance bottleneck that the
other does not have. The output-queued switches must run the switch fabric at
a speed greater than the sum of the speeds of the incoming links. This also
requires storing packets rapidly in output buffers. One way to get around this
problem is to place all queueing at the input. Input-queuing is often criticized
because of the head-of-line (HoL) blocking problem: packets blocked at the
head of an input queue prevent schedulable packets deeper within the queue
from accessing the switch fabric (Karol et al 1987).
However, with this approach, an arbiter must resolve contention for
the switching fabric and for the output queue. It is hard to design arbiters that
run at high speeds and can also fairly schedule the switch fabric and the
output line (McKeown et al 1996). Another disadvantage of input queuing is
that packet scheduling algorithms for providing quality of service are usually
specified in terms of output queues. Each input port controller should imitate
the actions of the entire set of output port controllers. With the different link
10
technologies, building a general-purpose input port controller is a challenging
task. Another problem with pure input-queued routers is that AQMs such as
Random Early Discard (Floyd and Jacobson 1993) depend on the length of
the output queue. With an input-queued switch, the output queue length is not
known. Due to these practical problems, it is significant that hybrid
approaches with both input and output queuing is a necessity for the next
generation networks.
1.2.1.4 Prioritization and Resource reservation
In this section, challenges in router design and the solutions in the
design of the next generation of IP routers are discussed. Flow identification
is a significant problem in routers. A flow is constituted by the set of packets
traveling through the Internet between a given source and a given destination.
A flow can result from the set of packets within a long-lasting TCP
connection or from the set of UDP packets in an audio or video session.
Optimization of usage of resources, such as buffers and cache entries is
sought. Therefore, it is necessary to identify flows on-the-fly. Flows that
require real-time QoS guarantees should be identified by matching incoming
packet headers with a set of pre-specified filters. Since classification is to be
done for each incoming packet, fast classification schemes are needed.
Since the Internet was designed for best-effort traffic, it has poor
support for resource reservations, even for the simple priority schemes. The
QoS requirements of the applications may demand support for resource
reservation in the routers. Resource reservation goes hand-in-hand with flow
classification, because resources are reserved on behalf of prespecified flows.
This coupling makes resource reservation an open problem. Even if there are
efficient flow classifiers, resource reservation additionally requires either
policing, so that the demand of an individual flow is limited, or some form of
11
segregation in packet scheduling, so that over-limit flows are automatically
discouraged. Given the complexity of implementing Fair-Queueing type
scheduling algorithms at high speed, there has been much research work done
on efficient policers.
1.2.2 QoS Architecture
The general definition of QoS is “A defined level of performance in
a data communications system”. Network providers need performance metrics
that they can agree with their peers and with service providers buying
resources from them with certain performance guarantees. The following four
system performance metrics are considered the most important for end-to-end
QoS:
Throughput: Throughput is the effective data transfer rate
measured in bits per second. Sharing a network lowers the
throughput that can be realized by any user, due to the
overhead imposed by the extra bits included in every packet
for identification and other purposes. A minimum rate of
throughput may be required by an application.
Packet loss: When a network link is congested, packets queue
up in the buffers of the routers. If the link remains congested
for too long, the buffered queues will overflow and data will
be lost.
Delay: The time taken by data to travel from the source to the
destination is known as delay.
Jitter: The variation of delay is jitter. Jitter results due to
variations in queue length, variations in the processing time
12
needed to reorder packets that arrived out of order because
they traveled over different paths and variations in the
processing time needed to reassemble packets that were
segmented by the source before being transmitted.
The important delays along the path are the nodal processing delay,
queuing delay, transmission delay and propagation delay. The most
complicated part of delay is the queuing delay. Many research papers and
books had been published on queuing delay (Kleinrock 1975; Bersekas and
Gallager 1992). When characterizing queuing delay, the average delay or
variance of delay (jitter) and the probability of the queuing delay exceeding
certain bound are explored (Kurose and Ross 2003).
1.2.2.1 Elastic and Inelastic traffic
Internet traffic may be divided into two broad categories, namely,
elastic and inelastic. Elastic traffic is one that can adjust to changes in delay
and throughput across the Internet. Inelastic traffic does not adapt to changes
in delay and throughput due to the nature of the applications. Many of the real
time multimedia applications fall under this category. Inelastic traffic
introduces two new requirements into the internet architecture. They are
preferential treatment to more demanding applications and support to elastic
applications in the presence of inelastic traffic (Stallings 2002).
1.2.2.2 QoS and router design
The QoS metrics can be controlled by the router design. There are
currently three main mechanisms to achieve a network performance that is
‘better than Best Effort’:
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Overprovision of capacity
Pre-reservation of resources
Prioritisation of certain services/users
In the access network, however, there is typically not much
installed fiber and therefore generally capacity is limited. Under these
circumstances, in order to support higher QoS than “Best Effort”, it is
necessary to be able to treat certain traffic differently than the rest, either by
specifically reserving resources (e.g. Integrated Services (IS)), or by
prioritising it (e.g Differentiated Services (DS)). The Internet Engineering
Task Force (IETF) has come up with DS, IS, and Resource Reservation
Protocol (RSVP) as beyond ‘Best Effort’ activities (Kurose and Ross 2003).
The network providers should achieve the service level agreement
(SLA) guarantees using the most cost-effective mechanisms. The IP router
design concentrates on the buffer management and the scheduling.
1.2.3 Buffer Management and QoS
The prioritization of mission critical applications and the support of
IP telephony and video conferencing create the requirement for supporting
QoS enforcement at the router. These applications are sensitive to both
absolute delay and delay jitter.
Beyond Best-Effort service, routers are beginning to offer a number
of QoS or priority classes. Priorities are used to indicate the preferential
treatment of one traffic class over another. The output buffered switch will
have multiple buffers at each output port and one buffer for each QoS traffic
class. The buffers may be physically separate or a physical buffer may be
divided logically into separate virtual buffers.
14
Buffer management here refers to the discarding policy for the
input of packets into the buffers (e.g., Drop Tail, Drop-From-Front, Random
Early Detection (RED), etc.) and the scheduling policy for the output of
packets from the buffers (e.g., strict priority, weighted round-robin (WRR),
weighted fair queueing (WFQ), etc.). Buffer management in the IP router
involves both dimensions of time (packet scheduling) and buffer space
(packet discarding). The IP traffic classes are distinguished in the time and
space dimensions by their packet delay and packet loss priorities. Therefore
buffer management and QoS support is an integral part of the switch fabric
design (Keshav and Sharma 1998).
1.2.4 Active Queue Management
Scheduling and AQM are the two ways to support QoS in IP
routers. In traditional implementations of router queue management, the
packets are dropped when a buffer becomes full, in which case the
mechanism is called Drop-Tail. Internet routers can improve application
goodput and response times by detecting congestion early and improving
fairness among flows. This is implemented in the routers by dropping packets
before a buffer becomes full, so that the senders can respond to the congestion
before the actual buffers overflow. Such a proactive approach is known as
Active Queue Management (AQM). Many AQMs have been proposed
including the most popular RED (Floyd and Jacobson, 1993). RED had been
recommended by IETF at the IP routers (Zheng and Atiquzzaman 2002). Due
to the characteristics of RED, TCP flows are benefited and UDP flows are
punished.
1.2.5 Scheduler
A scheduler at the IP routers decides which packet to send next
(Keshav 1997). If the packets arriving at all the input ports of a router wish to
15
leave from the same output port and if the output trunk speed is the same as
the input trunk speed, only one of these packets can be transmitted in the time
it takes for all of them to arrive at the output port. In order to prevent packet
loss, the output port provides buffers to store excess arriving packets and
serves packets from the buffer as and when the output trunk is free. The
obvious way to serve packets from the buffer is in the order they arrived at the
buffer, that is, in first-come-first-served (FCFS), or, FIFO order. FCFS
service is trivial to implement, requiring the router or switch to store only a
single head and tail pointer per output trunk. However, this solution has its
problems, because it does not allow the router to give some sources a lower
delay than others, or prevent a malicious source, that sends an unending
stream of packets as fast as it can. This may cause the other well-behaved
streams to loose packets. An alternative service method called Fair Queuing
solves these problems, albeit at a greater implementation cost (Demers et al,
1990). In the Fair Queuing approach, each source sharing a bottleneck link is
allocated an ideal rate of service at that link. Specifically, focusing only on
the sources that are backlogged at the link at a given instant in time, the
available service rate of the trunk is partitioned in accordance with a set of
weights. Fair Queuing and its variants are mechanisms that serve packets
from the output queue to approximately partition the trunk service rate in this
manner.
All versions of Fair Queuing require packets to be served in an
order different from the one in which they arrived. Consequently, Fair
Queuing is more expensive to implement than FCFS, since it must decide the
order in which to serve incoming packets and then manage the queues in
order to carry this out. When the traffic intensity is high, it is expensive to
implement Fair Queuing since Fair Queuing requires some form of per
conversation state to be stored on the routers. Fair Queuing has three
important and useful properties. First, it provides protection, so that a well-
behaved source does not see packet losses due to misbehavior by other
16
sources. Second, by design, it provides fair bandwidth allocation. If the sum
of weights of the sources is bounded, each source is guaranteed a minimum
share of link capacity. Finally, it can be shown that if a source is leaky-bucket
regulated, independent of the behavior of the other sources, it receives a
bound on its worst-case end-to-end delay. For these reasons, almost all
current routers support some variant of Fair Queueing.
A related scheduling problem has to do with the partitioning of link
capacity among different classes of users. It has been shown that extensions
of Fair Queueing are compatible with hierarchical link-sharing requirements
(Bennett and Zhang 1996; Goyal and Vin 1997). Fast implementations of
algorithms that provide both hierarchical link sharing and per-connection QoS
guarantees are an area of active research (Bennett and Zhang 1997). All
future routers are expected to provide some form of Fair Queueing at output
queues.
1.2.6 Internet Models
The Internet, as originally conceived, offers only a very simple
QoS, point-to-point best-effort data delivery. Before real-time applications
such as remote video, multimedia conferencing, visualization and virtual
reality can be broadly used, the Internet infrastructure must be modified to
support real-time QoS, which provides some control over end-to-end packet
delays. This extension must be designed from the beginning for multicasting;
simply generalizing from the unicast case does not work. The fundamental
service model of the Internet, as embodied in the best-effort delivery service
of IP, has been unchanged since the beginning of the Internet research project
three decades ago (Cerf and Kahn 1974).
17
Real-time QoS is not the only issue for a next generation of traffic
management in the Internet. Network operators are requesting the ability to
control the sharing of bandwidth on a particular link among different traffic
classes. They want to be able to divide traffic into a few administrative classes
and assign to each a minimum percentage of the link bandwidth under
conditions of overload, while allowing "unused" bandwidth to be available at
other times. These classes may represent different user groups or different
protocol families, for example. Such a management facility is commonly
called controlled link-sharing. IS (also known as IntServ) is an Internet
service model that includes best-effort service, real-time service and
controlled link sharing. IntServ relies on per-flow admission control, policing
and scheduling and it is not scalable.
RSVP is a resource reservation setup protocol designed for an
integrated services Internet. The RSVP protocol is used by a host to request
specific QoS from the network for particular application data streams or
flows. RSVP is also used by routers to deliver QoS requests to all nodes
along the path(s) of the flows and to establish and maintain state to provide
the requested service. RSVP requests result in resources being reserved in
each node along the data path. Some researchers concluded that there is an
inescapable requirement for routers to be able to reserve resources, in order to
provide special QoS for specific user packet streams, or "flows".
The DS (also known as DiffServ) Internet model differed from the
above approach. The motivation factors for Diffserv were scalability,
aggregation and high resource utilization. But, it heavily relies on network-
wide Service Level Agreement (SLA) monitoring and tactical and strategical
capacity planning. DiffServ and other class-based schemes (Parris et al 1999;
Chung and Claypool, 2000) offer differentiated service to incoming traffic.
18
They require complex mechanisms and need many network components such
as markers, traffic shapers etc.
Alternate Best Effort (ABE) (Hurley et al, 2001) provides an AQM
where throughput is forfeited for delay-sensitive traffic. Since no degrees of
sensitivity provided (delay- or throughput- sensitive), ABE is not flexible.
1.3 OBJECTIVES
The objective of the new Internet Service QoS architecture
combines the per-flow model of the IntServ and the per-hop model of the
DiffServ architectures. The flow of a favored multimedia is differentiated at
the IP router and given preferential treatment. The DS code point (DSCP) is
used to mark packets to select a favored packet. This is essentially a field in
the Type of Service (ToS) byte of the IPv4, or the DS field of IPv6 header.
The packet classification function is part of the AQM.
The QoS architecture is designed for inelastic flows that need QoS
support for delay jitter and packet loss.
1.4 MOTIVATION
The IP router functions with a simple logic. When the packets
arrive at the router, if the output buffer is empty, it may be just forwarded to
the destination through an output link. This link is selected by referring to the
forwarding table at the IP router. When the buffers are full, the packet may
even be dropped due to the Best Effort nature of the Internet. Hence there is a
need for QoS architecture at the IP routers to support distributed multimedia
applications that require good QoS. Some important considerations for the
router design are throughput, packet loss, packet delays, amount of buffering
19
and complexity of implementation. For given input traffic, the router designs
aim to maximize throughput and minimize packet delays and losses. In
addition, the total amount of buffering should be minimal (to sustain the
desired throughput without incurring excessive delays) and implementation
should be simple.
As the Internet is unfriendly to stream traffic generated by voice
and video (Parris et al 1999), it is needed to expand the IP designs for the next
generation networks. The IP routers should be designed in such a way to
encompass necessary intelligence to recognize, control and prioritize different
types of traffic. The arriving traffic should be differentiated and the individual
QoS requirements should be satisfied. Especially, in distributed systems the
performance depends on the structure of the communication network (Fischer
and Merritt 2003). Research has been done to improve performance by
improving the bandwidth requirements and reducing jitter. The AQM
approach, while reducing congestion, also improves the performance of the
applications. The research community has proposed various AQM techniques
out of which RED has become the de-facto standard after the IETF
recommended it for the IP routers.
The traditional Tail-Drop algorithm drops packets only if there are
buffer overflows. It is easy to implement (Keshav 1997) but may result in
longer queuing delay for the packets. RED avoids congestion by dropping
packets randomly before the buffer overflows. Though this scheme reduces
the average delay and helps avoid congestion, the low pass filter algorithm,
which is used to calculate the average queue length in RED, results in poor
response time when RED recovers from congestion (Zheng and Atiquzzaman
2002). RED can reduce the queuing delays at the routers and hence the
end-to-end delay, but increases the jitter of non-bursty streams (Bonald et al
2000).
20
Suggestions have been put forth by many researchers, for AQMs
which do not strictly use average queue lengths (Ott et al 1999; Athuraliya
et al 2001; Hollot et al 2001b; Kunniyur and Srikant 2004). Many of them
require internet-like configurations and especially Explicit Congestion
Notification (ECN) marking (Floyd, 1994) which they recommend. Some are
difficult to configure and some are complex to implement. Many studies have
debated whether dropping/marking should be based on queue length or on
input and output rates (or alternatively the queue length slope over time). The
objective is to keep the average queuing delay under a specified target, thus
reducing web response time, without significantly affecting application
throughput and link utilisation.
Other weaknesses of RED have been reported and several
approaches to overcome them have been proposed. When a mixture of the
various traffic types shares a link, RED allows unfair bandwidths. This AQM
uses per-flow soft states with instantaneous buffer size monitoring.
RED parameters can be tuned to provide either high utilisation or
low queuing delay but not both (Lapsley and Low 1999; Athurliya et al 2001).
Certain researchers use a class of algorithms, which also includes the PI
controller, (Hollot et al 2001b) that is mainly designed for marking (ECN)
and has not been adequately studied without ECN.
Hence there arises a need to achieve a simple AQM algorithm that
can resolve the QoS issues without compromising the utilization factor. In
addition a scheduler policy is needed at the output queues to guarantee jitter.
The novel Gentle- Flow-based Proactive Queuing (GFPQ) algorithm drops
packets from the buffer proactively with preferential treatment to prioritized
multimedia packets. The Jitter Guaranteed Time-stamp Scheduler (JGTS)
scheduler guarantees jitter to an upper bound with the help of time stamps.
21
This also led to the idea of a new, simple service model for the Internet to
offer ‘better than Best Effort’ service. The two common service models IS
and DS have been deployed to a significant extent in the Internet.
New research works have been progressing in other directions as
well to introduce new service models. Hurley et al (2001) discuss the ABE
Model. Scavenger Networks introduce a model of marking voluntarily some
of the packets as low-priority ones. QBone Scavenger Service (QBSS) can
expand to consume unused capacity. Users (or their applications) voluntarily
mark some traffic for scavenger treatment by setting Differentiated Services
Code Point (DSCP) in the IP packet headers to binary 001000. Routers put
this traffic into a special queue with very small allocated capacity using a
queuing discipline such as Weighted Round-Robin (WRR), Modified Deficit
Round-Robin (MDRR), Weighted Fair Queuing (WFQ), or a similar scheme.
This thesis indicates that the integrated design of the novel AQM
and the new scheduler can be used for the new service model. Here, the two
types of traffic, namely favored and the non-favored are treated differently at
the IP routers, but still ensures inter-protocol fairness involving all types of
traffic such as TCP, UDP, and TCP-FTP.
1.5 CONTRIBUTIONS
The specific contributions made in this thesis are in two areas,
namely design and analysis of a novel AQM and design of a new scheduler.
From these, a novel Internet Service model with a QoS architecture is
designed. An overview of the contributions made is given below.
22
1.5.1 AQM Issues
GFPQ framework is built on the ‘queue based’ approach. A
novel AQM algorithm at the input queues of the IP router is
evolved. It is designed in such a way that when a favored
multimedia packet arrives, it is at best accommodated into the
queue. The QoS for such flows is supported by reducing their
blocking probability (drop rate). The other types of traffic are
considered ‘non-favored’. Yet, the throughput is ensured for
all types of packets. This leads to Inter-Protocol fairness and
ensures that favoring a single class does not affect
significantly the throughput of others. This leads to not only
improvement in throughput for favored multimedia UDP
flows, but also for other types of traffic. Thus GFPQ with
packet classification supports QoS, namely loss rate.
GFPQ is an AQM that is based on instantaneous queue
monitoring as against average queue length used in RED and
the first version, FPQ. This helps in reducing the average
queuing delay at the router.
GFPQ is stateless and easy to implement. The design and
implementation methodologies are presented in this work.
Performance analysis of the queuing model for GFPQ is
presented.
Qualitative analysis of GFPQ is presented. The analytical
modeling of the pushout policies of GFPQ supported by
quantitative analysis through simulation is also presented.
This framework is also generic in the sense that any mix of
traffic can be used with cross-over traffic.
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However, it is recognized that the present and next-generation
multimedia applications have a jitter problem. As jitter is an important QoS
parameter, it becomes necessary to address this issue. Therefore, the QoS
architecture concentrates on reducing jitter with the help of a novel scheduler
at the output queues of the routers.
1.5.2 Jitter reduction by scheduling
A novel scheduling technique JGTS is designed. The JGTS
incorporates a jitter manager module whose function is to
reduce jitter for the multimedia flows.
The time complexity analysis of JGTS is presented.
Scalability issues are also addressed at the scheduler with
throughput and jitter characteristics. The effectiveness of this
technique is analyzed in a standalone mode using simulation
and the results are presented.
The unified scheme with GFPQ and JGTS is tested for
scalability and its performance is found to be better than the
de-facto RED technique.
1.5.3 QoS Architecture
A novel QoS architecture with GFPQ and scheduler JGTS is
proposed to support multimedia flows at the network routers.
Enhanced QoS for loss reduction and jitter reduction at the
input and output queues, respectively, is provided.
The unified framework is found to increase the overall
throughput for the favored multimedia applications.
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This QoS architecture leads to a novel Internet service model
to provide QoS for high-performance applications. The
performance enhancement at existing IP routers, without
change in the infrastructure is proposed.
1.6 ORGANIZATION OF THE THESIS
This thesis consists of six chapters. Chapter 1 introduces the
Internet models and describes the IP router organization. It also gives the
conventional queue management and scheduling techniques deployed in the
best effort Internet.
Chapter 2 presents the state of the art of the resource management
techniques and scheduling in the traditional and derived Internet. It brings out
the related research works done in the areas of AQM, scheduling the Internet
service models. The buffer management techniques employed in IP routers
supported by schedulers at the output queues are discussed.
Chapter 3 discusses the two approaches used in this work for
providing quality of service in IP routers. It starts with the block diagram of
the integrated system showing the buffer management and scheduler. The
GFPQ framework used for the active queue management at the input queues
is presented here. The novel scheduler design (JGTS) and deployment
techniques are discussed.
Chapter 4 explains the AQM and scheduler algorithms and their
analysis. This discusses the introduction of the FPQ based AQM to make the
framework help in getting the first objective namely, the loss rate reduction.
The GFPQ algorithm that makes effective use of the available buffer space is
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then introduced. This is the second version of the AQM proposed in this
work. The queuing model and the analysis of the AQM are then discussed.
Design objectives for this proposed AQM are also provided. The random
replacement algorithm in the AQM and the pushout policies are discussed.
The design of the novel scheduler JGTS for deployment at the output queue
and its algorithm are presented.
The results are discussed in chapter 5. The performance evaluation
of GFPQ and JGTS are presented. The effects of the combined approach are
also discussed. Simulation results showing the effectiveness of these
techniques are provided. The comparison of results with the various
traditional techniques are provided and analyzed, with respect to throughput,
delay and delay jitter. The importance of the integrated approach is
emphasized with the results showing jitter and loss rate reduction.
Chapter 6 discusses the most significant contributions of this thesis
and possible future research directions.