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Course Material - HSN

Frame relay and ATM are high-speed networking technologies. Frame relay uses a two-layer architecture of physical and data link layers, with minimal data link control. It does not support hop-by-hop flow control or error control. ATM is based on small, fixed-size cells and uses virtual circuits to transfer data asynchronously. Each ATM cell is 53 bytes with a 5-byte header including a virtual path/channel identifier. The ATM architecture separates user and control planes. It supports quality of service guarantees by dividing user data into fixed-length cells transported over virtual connections.

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60 views99 pages

Course Material - HSN

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MUTHAYAMMAL ENGINEERING COLLEGE

(An Autonomous Institution)

(Approved by AICTE, New Delhi, Accredited by NAAC & Affiliated to Anna University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu

L-1
LECTURE HANDOUTS

CSE III/VI

Course Name with Code :High Speed Networks-16CSE10

Course Faculty : C.Nithya

Unit : I- High Speed Networks Date of Lecture:

Topic of Lecture: Frame relay Networks-Asynchronous Transfer Mode

Introduction :
Designed to eliminate much of the overhead in X.25
_ Call control signaling on separate logical connection from user data
_ Multiplexing/switching of logical connections at layer 2 (not layer 3)
_ No hop-by-hop flow control and error control
_ Throughput an order of magnitude higher than X.25
Prerequisite knowledge for Complete understanding and learning of Topic:
• Open system
• Closed system
• Protocol
• Host name
• IP Address
Detailed content of the Lecture:
Frame Relay Architecture
X.25 has 3 layers: physical, link, network
• Frame Relay has 2 layers: physical and data link (or LAPF)
• LAPF core: minimal data link control
• Preservation of order for frames
• Small probability of frame loss
• LAPF control: additional data link or network layer end-to-end functions
LAPF Core
1. Frame delimiting, alignment and transparency
2. Frame multiplexing/demultiplexing
3. Inspection of frame for length constraints
4. Detection of transmission errors
5. Congestion control
User Data Transfer
• No control field, which is normally used for:
• Identify frame type (data or control)
• Sequence numbers
• Implication: Connection setup/teardown carried on separate channel
• Cannot do flow and error control

Frame Relay Call Control

• Frame Relay Call Control
• Data transfer involves:
• Establish logical connection and DLCI
• Exchange data frames
• Release logical connection
4 message types needed
• SETUP
• CONNECT
• RELEASE
• RELEASE COMPLETE
Asynchronous Transfer Mode (ATM):
• It is an International Telecommunication Union- Telecommunications Standards Section
(ITU-T) efficient for call relay and it transmits all information including multiple service
types such as data, video or voice which is conveyed in small fixed size packets called cells.
• Cells are transmitted asynchronously and the network is connection oriented.
• ATM is a technology which has some event in the development of broadband ISDN in
1970s and 1980s, which can be considered an evolution of packet switching.
• Each cell is 53 bytes long – 5 bytes header and 48 bytes payload. Making an ATM call
requires first sending a message to set up a connection.
• Subsequently all cells follow the same path to the destination. It can handle both constant
rate traffic and variable rate traffic.
• Thus it can carry multiple types of traffic with end-to-end quality of service.
• ATM is independent of transmission medium, they maybe sent on a wire or fiber by
themselves or they may also be packaged inside the payload of other carrier systems.
• ATM networks use “Packet” or “cell” Switching with virtual circuits. It’s design helps in
the implementation of high performance multimedia networking.
ATM Cell Format –
As information is transmitted in ATM in the form of fixed size units called cells. As known already
each cell is 53 bytes long which consists of 5 bytes header and 48 bytes payload.
Asynchronous Transfer Mode can be of two format types which are as follows:
1. UNI Header: which is used within private networks of ATM for communication between ATM
endpoints and ATM switches. It includes the Generic Flow Control (GFC) field.
2. NNI Header: is used for communication between ATM switches, and it does not include the
Generic Flow Control(GFC) instead it includes a Virtual Path Identifier (VPI) which occupies the
first 12 bits.
Video Content / Details of website for further learning (if any):
https://www.geeksforgeeks.org/difference-between-frame-relay-and-atm/
https://www.youtube.com/watch?v=RP4OAbyVjV8&t=597s
Important Books/Journals for further learning including the page nos.:
William Stallings, ”High speed networks and internet”, Pearson Education 2002,Page No: (46-49)

Course Faculty

Verified by HOD
MUTHAYAMMAL ENGINEERING COLLEGE
(An Autonomous Institution)
(Approved by AICTE, New Delhi, Accredited by NAAC & Affiliated to Anna University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu

LECTURE HANDOUTS L-2

CSE III/VI
Course Name with Code :High Speed Networks-16CSE10

Course Faculty : C.Nithya

Unit : I- High Speed Networks Date of Lecture:

Topic of Lecture: ATM Protocol Architecture, ATM logical Connection

Introduction :
• A streamlined packet transfer interface
• Similarities to packet switching
• Transfers data in discrete chunks
• Supports multiple logical connections over a single physical interface
Prerequisite knowledge for Complete understanding and learning of Topic:
• Protocol
• Host name
• IP Address
Detailed content of the Lecture:
ATM Protocol Architecture
• The asynchronous transfer mode (ATM) protocol architecture is designed to support the
transfer of data with a range of guarantees for quality of service.
• The user data is divided into small, fixed-length packets, called cells, and transported over
virtual connections.
• ATM operates over high data rate physical circuits, and the simple structure of ATM cells
allows switching to be performed in hardware, which improves the speed and efficiency of
ATM switches.
• The first thing to notice is that, as well as layers, the model has planes.
• The functions for transferring user data are located in the user plane; the functions
associated with the control of connections are located in the control plane; and the co-
ordination functions associated with the layers and planes are located in the management
planes.
• The three-dimensional representation of the ATM protocol architecture is intended to
portray the relationship between the different types of protocol. An advantage of dividing
the functions into control and user planes is that it introduces a degree of independence in
the definition of the functions: the protocols for transferring user data (user plane) are
separated from the protocols for controlling connections (control plane).
• The protocols in the ATM layer provide communication between ATM switches while the
protocols in the ATM adaptation layer (AAL) operate end-to-end between users.
ATM logical Connection
• Each ATM cell consists of 53 bytes: the header is five bytes long and the remaining 48
bytes (the cell payload) carry information from higher layers. The only difference between
the two types of ATM cell is that the cells at the user-network interface carry a data field for
the flow control of data from users. This means that only eight bits are available for virtual
path identifiers, rather than 12 bits at the network-node interface.
• The virtual connections set up in ATM networks are identified by the combination of
the virtual path identifier and virtual channel identifier. These two fields provide a
hierarchy in the numbering of virtual connections, whereby a virtual path contains a number
of virtual channels. An advantage of this hierarchy is that in some cases the switching of
ATM cells may be based on the virtual path identifier alone.

Video Content / Details of website for further learning (if any):

https://www.youtube.com/watch?v=P6Q5QZAZ8yc
https://www.youtube.com/watch?v=RzcFTVf-mCE&t=17s
Important Books/Journals for further learning including the page nos.:
William Stallings, ”High speed networks and internet”, Pearson Education 2002-Pg.No (46-49)

Course Faculty

L-3
LECTURE HANDOUTS

CSE III/VI

Course Name with Code :High Speed Networks-16CSE10

Course Faculty : C.Nithya

Unit : I- High Speed Networks Date of Lecture:

Topic of Lecture: ATM Cell

Introduction :
• ATM networks are connection oriented networks for cell relay that supports voice, video and
data communications.
• It encodes data into small fixed - size cells so that they are suitable for TDM and transmits them
over a physical medium. The size of an ATM cell is 53 bytes: 5 byte header and 48 byte
payload.
Prerequisite knowledge for Complete understanding and learning of Topic:
• Open system
• Closed system
• Protocol
• Host name
• IP Address
Detailed content of the Lecture:
ATM Cells
• Cell in ATM is a 53-byte packet of data, the standard packet size used by Asynchronous
Transfer Mode (ATM) communication technologies.
• Cells are to ATM technologies what frames are to Ethernet networking.
• In other words, they form the smallest element of data for transmission over the network.

•
• ATM cells are standardized at a fixed-length size of 53 bytes to enable faster switching than
is possible on networks using variable-packet sizes (such as Ethernet).
• It is much easier to design a device to quickly switch a fixed-length packet than to design a
device to switch a variable-length packet.
• Using fixed-length cells also makes it possible to control and allocate ATM bandwidth
more effectively, making support for different quality of service (QoS) levels for ATM
possible.
ATM Cell Format

Header Format
– Generic flow control
– Only at user to network interface
– Controls flow only at this point
– Virtual path identifier
– Virtual channel identifier
– Payload type
– e.g. user info or network management
– Cell loss priority
– Header error control
Generic Flow Control (GFC)
– Control traffic flow at user to network interface (UNI) to alleviate short term overload
– Two sets of procedures
– Uncontrolled transmission
– Controlled transmission
– Every connection either subject to flow control or not
– Subject to flow control
– May be one group (A) default
– May be two groups (A and B)
– Flow control is from subscriber to network
– Controlled by network side
Single Group of Connections (1)
– Terminal equipment (TE) initializes two variables
– TRANSMIT flag to 1
– GO_CNTR (credit counter) to 0
– If TRANSMIT=1 cells on uncontrolled connection may be sent any time
– If TRANSMIT=0 no cells may be sent (on controlled or uncontrolled connections)
– If HALT received, TRANSMIT set to 0 and remains until NO_HALT
Single Group of Connections (2)
– If TRANSMIT=1 and no cell to transmit on any uncontrolled connection:
– If GO_CNTR>0, TE may send cell on controlled connection
– Cell marked as being on controlled connection
– GO_CNTR decremented
– If GO_CNTR=0, TE may not send on controlled connection
– TE sets GO_CNTR to GO_VALUE upon receiving SET signal
– Null signal has no effect
Header Error Control
– 8 bit error control field
– Calculated on remaining 32 bits of header
– Allows some error correction

HEC Operation at Receiver

Real Time Services
• Amount of delay
• Variation of delay (jitter)
CBR
– Fixed data rate continuously available
– Tight upper bound on delay
– Uncompressed audio and video
– Video conferencing
– Interactive audio
– A/V distribution and retrieval
rt-VBR
– Time sensitive application
– Tightly constrained delay and delay variation
– rt-VBR applications transmit at a rate that varies with time
– e.g. compressed video
– Produces varying sized image frames
– Original (uncompressed) frame rate constant
– So compressed data rate varies
– Can statistically multiplex connections
Video Content / Details of website for further learning (if any):
https://www.geeksforgeeks.org/difference-between-frame-relay-and-atm/
https://www.youtube.com/watch?v=RP4OAbyVjV8&t=597s
Important Books/Journals for further learning including the page nos.:
William Stallings, ”High speed networks and internet”, Pearson Education 2002-Pg.No (63-70)

Course Faculty

L-4
LECTURE HANDOUTS

CSE III/VI

Course Name with Code :High Speed Networks-16CSE10

Course Faculty : C.Nithya

Unit : I- High Speed Networks Date of Lecture:

Topic of Lecture: ATM Service Categories : AAL

Introduction :
Asynchronous Transfer Mode (ATM) is a telecommunications standard defined by ANSI and ITU
(formerly CCITT) for digital transmission of multiple types of traffic, including telephony (voice),
data, and video signals in one network without the use of separate overlay networks.
Prerequisite knowledge for Complete understanding and learning of Topic:
• Open system
• Closed system
• Protocol
• Host name
• IP Address
Detailed content of the Lecture:
• In Asynchronous Transfer Mode (ATM) networks, the ATM Adaptation Layer (AAL) provides
facilities for non-ATM based networks to connect to ATM network and use its services.
• AAL is basically a software layer that accepts user data, which may be digitized voice, video or
computer data, and makes them suitable for transmission over an ATM network. The
transmissions can be of fixed or variable data rate.
• AAL accepts higher layer packets and segments them into fixed sized ATM cells before
transmission via ATM. It also reassembles the received segments to the higher layer packets.
Function of AAL

This layer has two sub layers −

• Convergence sub layer

• Segmentation and Reassembly sub layer.

Some networks that need AAL services are Gigabit Ethernet, IP, Frame Relay, SONET/SDH and
UMTS/Wireless.

AAL Protocols
International Telecommunication Union Telecommunication Standardization Sector (ITU-T) has
defined five AAL protocols to provide the range of services.

• AAL Type 0 − This is the simplest service that provides direct interface to ATM services
without any restrictions. These cells are called raw cells that contain 48-byte payload field
without any special fields. It lacks guaranteed delivery and interoperability.

• AAL Type 1 − This service provides interface for synchronous, connection oriented traffic. It
supports constant rate bit stream between the two ends of an ATM link. An AAL 1 cell contains
a 4-bit sequence number, a 4-bit sequence number protection and a 47-byte payload field.

• AAL Type 2 − This service also provides interface for synchronous, connection oriented
traffic. However, this is for variable rate bit stream between the two ends of an ATM link. It is
used in wireless applications.

Topic of Lecture: Fibre Channel

Introduction :
Fibre Channel is a high-speed networking technology primarily used for transmitting data among data
centers, computer servers, switches and storage at data rates of up to 128 Gbps.
Prerequisite knowledge for Complete understanding and learning of Topic:
• I/O Operations
• Buffers
• Error Detection
• Error Recovery
Detailed content of the Lecture:
I/O channel communication
Direct point-to-point or multipoint communications link and mainly designed for high speed over very
short distance.
Network channel communication
• A Network is a collection of interconnected access points with a software protocol structure. It
allows different types of data rates and provide flow and error control. Fiber channel is
designed to combine the best features of both technologies .
• In fiber channel , the simplicity and speed of channel communication is combined with
flexibility and
inter connectivity of network based communication .
Requirements of fibre channel
• Full duplex links with two fibers per link
• Performance from 100 Mbps TO 800 Mbps on a single line
• Support for distances up to 10 km
• High capacity utilization with distance insensitivity
• Broad availability
• Support multicost
Fibre channel elements
1. N- ports
Key elements of a fiber channel network are the end systems called nodes. Each node include one or
more ports called N-ports.
2. F-ports
Network itself consists of one or more Switching elements. The collection of switching elements is
referred to as a fabric. Each fabric switching elements include multiple ports called F-ports.
Fibre Channel Protocol Architecture

Fiber channel organized into five levels

1. FC-0 Physical Media -includes optical fiber for long distance, coaxial cable for high speed
over short distance, shielded twisted pair for lower speeds over short distance.
2. FC-1 Transmission Protocol -it defines the signal encoding scheme.
3. FC-2 Framing Protocol –deals with topologies frame format, flow of error control
4. FC-3 Common Services –it includes multicasting
5. FC-4 Mapping –mapping of various channel and network protocols to fiber channel including
IEEE 802.11, ATM, IP, SCSI
Transmission Media
• Video coaxial cable
• optical fiber
• shielded twisted pair
Topologies
Topology supported by fiber channel are
• Fabric or switcher topology
• point –to-point topology
• o arbitrated loop topology.
Video Content / Details of website for further learning (if any):
https://www.youtube.com/watch?v=Ac8824tFS8c
https://www.youtube.com/watch?v=Ish0DOaTdtA
Important Books/Journals for further learning including the page nos.:
William Stallings, ”High speed networks and internet”, Pearson Education 2002-Pg.No(140-144)

Course Faculty

L-8
LECTURE HANDOUTS

CSE III/VI

Course Name with Code :High Speed Networks-16CSE10

Course Faculty : C.Nithya

Unit : I- High Speed Networks Date of Lecture:

Topic of Lecture: Wireless LAN’s :WiFi and WiMax Networks applications, Requirements
Introduction :
• WiMax can be used to provide internet services such as mobile data and WiFi spots. Wifi is
defined under IEEE 802.11x standards where x stands for various WiFi versions.
• WiMax is defined under IEEE 802.16y standards where y stands for various WiMax versions.
• WiFi is for LAN (Local Area Network) applications.
Prerequisite knowledge for Complete understanding and learning of Topic:
• LAN
• Ethernet
• Wireless Communication
Detailed content of the Lecture:
• WiMAX (Worldwide Interoperability for Microwave Access) is a family of wireless
broadband communication standards based on the IEEE 802.16 set of standards, which provide
multiple physical layer (PHY) and Media Access Control (MAC) options.
• The name "WiMAX" was created by the WiMAX Forum, which was formed in June 2001 to
promote conformity and interoperability of the standard, including the definition of predefined
system profiles for commercial vendors.
• The forum describes WiMAX as "a standards-based technology enabling the delivery of last
mile wireless broadband access as an alternative to cable and DSL.
• IEEE 802.16m or Wireless MAN-Advanced was a candidate for the 4G, in competition with
the LTE Advanced standard.
• WiMAX was initially designed to provide 30 to 40 megabit-per-second data rates, with the
2011 update providing up to 1 Gbit/s for fixed stations.
• The latest version of WiMAX, WiMAX release 2.1, popularly branded as/known as WiMAX
2+, is a smooth, backwards-compatible transition from previous WiMAX generations. It is
compatible and inter-operable with TD-LTE.
Uses of WiMAX
The scalable physical layer architecture that allows for data rate to scale easily with available channel
bandwidth and range of WiMAX make it suitable for the following potential applications:
• Providing portable mobile broadband connectivity across cities and countries through various
devices.
• Providing a wireless alternative to cable and digital subscriber line (DSL) for "last mile" broadband
access.
• Providing data, telecommunications (VoIP) and IPTV services (triple play).
• Providing Internet connectivity as part of a business continuity plan.
• Smart grids and metering.
Requirements
A wireless LAN must meet the same sort of requirements typical of any LAN, including high capacity,
ability to cover short distances, full connectivity among attached stations, and broadcast capability. In
addition, a number of requirements are specific to the wireless LAN environment. The following are
among the most important requirements for wireless LANs:
Throughput: The medium access-control (MAC) protocol should make as efficient use as possible of
the wireless medium to maximize capacity.
Number of nodes: Wireless LANs may need to support hundreds of nodes across multiple cells.
Connection to backbone LAN: In most cases, interconnection with stations on a wired backbone
LAN is required. For infrastructure wireless LANs, this is easily accomplished through the use of
control modules that connect to both types of LANs. There may also need to be accommodation for
mobile users and ad hoc wireless networks.
Service area. A typical coverage area for a wireless LAN has a diameter of 100 to 300 m.
Battery power consumption: Mobile workers use battery-powered workstations that need to have a
long battery life when used with wireless adapters. This suggests that a MAC protocol that requires
mobile nodes to monitor access points constantly or engage in frequent handshakes with a base station
is inappropriate. Typical wireless LAN implementations have features to reduce power consumption
while not using the network, such as a sleep mode.
Transmission robustness and security: Unless properly designed, a wireless LAN may be
interference-prone and easily eavesdropped. The design of a wireless LAN must permit reliable
transmission even in a noisy environment and should provide some level of security from
eavesdropping.
Collocated network operation: As wireless LANs become more popular, it's quite likely that two or
more wireless LANs will operate in the same area or in some area where interference between the
LANs is possible. Such interference may thwart the normal operation of a MAC algorithm and may
allow unauthorized access to a particular LAN.
License-free operation: Users would prefer to buy and operate wireless LAN products without
having to secure a license for the frequency band used by the LAN.
Handoff/roaming: The MAC protocol used in the wireless LAN should enable mobile stations to
move from one cell to another.
Dynamic configuration: The MAC addressing and network management aspects of the LAN should
permit dynamic and automated addition, deletion, and relocation of end systems without disruption to
other users.
Video Content / Details of website for further learning (if any):
https://en.wikipedia.org/wiki/WiMAX
https://www.youtube.com/watch?v=RP4OAbyVjV8&t=597s
Important Books/Journals for further learning including the page nos.:
William Stallings, ”High speed networks and internet”, Pearson Education 2002-Pg.No (173-181)

Course Faculty

L-9
LECTURE HANDOUTS

CSE III/VI

Course Name with Code : High Speed Networks-16CSE10

Course Faculty : C.Nithya

Course Name with Code : High Speed Networks-16CSE10

Course Faculty : C.Nithya

Unit : II- Congestion and Traffic Management Date of Lecture:

Topic of Lecture: Effects of Congestion

Introduction :
• Congestion shrinks the delivery area that any one driver and vehicle can serve. This means that
there is a need to have more vehicles on the road (to maintain and grow distribution and
trucking markets), and routes need to be changed more often.
Prerequisite knowledge for Complete understanding and learning of Topic:
• Queuing theory
• Networking
• Queuing Networks
Detailed content of the Lecture:
Effects of Congestion
Congestion-Control Mechanisms
Backpressure
– Request from destination to source to reduce rate
– Useful only on a logical connection basis
– Requires hop-by-hop flow control mechanism
Policing
– Measuring and restricting packets as they enter the network
Choke packet
– Specific message back to source
E.g., ICMP Source Quench
Implicit congestion signaling
– Source detects congestion from transmission delays and lost packets and reduces flow
Explicit congestion signaling
• Frame Relay reduces network overhead by implementing simple congestion notification
mechanisms rather than explicit, per-virtual-circuit flow control.
• Frame Relay typically is implemented on reliable network media, so data integrity is not
sacrificed because flow control can be left to higher-layer protocols.
o Frame Relay implements two congestion-notification mechanisms:
o Forward-explicit congestion notification (FECN)
• Backward-explicit congestion notification (BECN) FECN and BECN each is controlled by a
single bit contained in the Frame Relay frame header.
• The Frame Relay frame header also contains a Discard Eligibility (DE) bit, which is used to
identify less important traffic that can be dropped during periods of congestion.
• If the network is congested, DCE devices (switches) set the value of the frames' FECN bit to 1.
• When the frames reach the destination DTE device, the Address field (with the FECN bit set)
indicates that the frame experienced congestion in the path from source to destination.
• The DTE device then can relay this information to a higher-layer protocol for processing.
Depending on the implementation, flow-control may be initiated, or the indication may be
ignored.
Frame Relay Discard Eligibility
• The Discard Eligibility (DE) bit is used to indicate that a frame has lower importance than other
frames. The DE bit is part of the Address field in the Frame Relay frame header.
• DTE devices can set the value of the DE bit of a frame to 1 to indicate that the frame has lower
importance than other frames. When the network becomes congested, DCE devices will
discard frames with the DE bit set before discarding those that do not.
• This reduces the likelihood of critical data being dropped by Frame Relay DCE devices during
periods of congestion.
Frame Relay Error Checking
• Frame Relay uses a common error-checking mechanism known as the cyclic redundancy check
(CRC). The CRC compares two calculated values to determine whether errors occurred during
the transmission from source to destination.
• Frame Relay reduces network overhead by implementing error checking rather than error
correction. Frame Relay typically is implemented on reliable network media, so data integrity is
not sacrificed because error correction can be left to higher-layer protocols running on top of
Frame Relay.

Video Content / Details of website for further learning (if any):

https://www.youtube.com/watch?v=6Ojr-Q319r8
Important Books/Journals for further learning including the page nos.:
William Stallings, ”High speed networks and internet”, Pearson Education 2002.Page No: (146-147)

Course Faculty

L-14
LECTURE HANDOUTS

CSE III/VI

Course Name with Code : High Speed Networks-16CSE10

Course Faculty : C.Nithya

Unit : II- Congestion and Traffic Management Date of Lecture:

Topic of Lecture: Congestion Control

Introduction :
• Congestion control refers to techniques and mechanisms that can either prevent congestion,
before it happens, or remove congestion, after it has happened.
• In general, we can divide congestion control mechanisms into two broad categories: open-loop
congestion control (prevention) and closed-loop congestion control (removal)
Prerequisite knowledge for Complete understanding and learning of Topic:
• Queuing theory
• Networking
• Protocols
Detailed content of the Lecture:
Congestion Control
Backpressure:
• If node becomes congested it can slow down or halt flow of packets from other nodes
• May mean that other nodes have to apply control on incoming packet rates
• Propagates back to source
• Can restrict to logical connections generating most traffic
• Used in connection oriented that allow hop by hop congestion control (e.g. X.25)
• Not used in ATM nor frame relay

Choke Packet :A choke packet is a packet sent by a node to the source to inform it of congestion. Note
the difference between the backpressure and choke packet methods. In backpressure choke packet send
the message to source directly.
Control packet
— Generated at congested node
— Sent to source node
— e.g. ICMP source quench
• From router or destination
• Source cuts back until no more source quench message
• Sent for every discarded packet, or anticipated
• Rather crude mechanism
Implicit Signaling
In implicit signaling, there is no communication between the congested node or nodes and the source.
The source guesses that there is a congestion somewhere in the network from other symptoms.
Explicit Signaling
The node that experiences congestion can explicitly send a signal to the source or destination. The
explicit signaling method, however, is different from the choke packet method.
Explicit congestion signaling approaches can work in one of two directions:
• Backward and
• Forward.
We can divide explicit congestion signaling approaches into three general
categories:
• BINARY: A bit is set in a data packet as it is forwarded.
• CREDIT BASED: by the congested node.
These schemes are based on providing an explicit credit to a source
over a logical connection.
Video Content / Details of website for further learning (if any):
https://www.youtube.com/watch?v=6Ojr-Q319r8
Important Books/Journals for further learning including the page nos.:
William Stallings, ”High speed networks and internet”, Pearson Education 2002,Page No: (146-147)

Course Faculty

L-15
LECTURE HANDOUTS

CSE III/VI

L-20
LECTURE HANDOUTS

CSE III/VI
Course Name with Code : High Speed Networks-16CSE10

Course Faculty : C.Nithya

Unit : III- TCP and ATM Congestion Control Date of Lecture:

Topic of Lecture: Retransmission – Timer Management- Exponential RTO- backoff – KARN’s

Algorithm
Introduction :
To retransmit lost segments, TCP uses retransmission timeout (RTO). When TCP sends a segment
the timer starts and stops when the acknowledgment is received. If the timer expires timeout occurs and
the segment is retransmitted.
Prerequisite knowledge for Complete understanding and learning of Topic:
• Acknowledgement
• TCP
• Retransmission
• Segment
Detailed content of the Lecture:
Retransmission Strategy
• TCP relies exclusively on positive acknowledgements and retransmission on acknowledgement
timeout
• There is no explicit negative acknowledgement
• Retransmission required when:
• Segment arrives damaged, as indicated by checksum error, causing receiver to discard segment
• Segment fails to arrive
Timers
• A timer is associated with each segment as it is sent
• If timer expires before segment acknowledged, sender must retransmit
Key Design Issue:
• value of retransmission timer
• Too small: many unnecessary retransmissions, wasting network bandwidth
• Too large: delay in handling lost segment
Two Strategies
• Timer should be longer than round-trip delay (send segment, receive ack)
• Delay is variable
Strategies:
• Fixed timer
• Adaptive
Problems with Adaptive Scheme
• Peer TCP entity may accumulate acknowledgements and not acknowledge immediately
• For retransmitted segments, can’t tell whether acknowledgement is response to original
transmission or retransmission
• Network conditions may change suddenly
Implementation Policy Options
• Send
• Deliver
• Accept
• In-order
• In-window
• Retransmit
• First-only
• Batch
• individual
• Acknowledge
• immediate
• cumulative
TCP Flow and Congestion Control

Retransmission Timer Management

Three Techniques to calculate retransmission timer (RTO):
• RTT Variance Estimation
• Exponential RTO Backoff
• Karn’s Algorithm
RTT Variance Estimation (Jacobson’s Algorithm)
3 sources of high variance in RTT
• If data rate relative low, then transmission delay will be relatively large, with larger variance
due to variance in packet size
• Load may change abruptly due to other sources
• Peer may not acknowledge segments immediately
Two Other Factors
Jacobson’s algorithm can significantly improve TCP performance
Exponential RTO Backoff
• Increase RTO each time the same segment retransmitted – backoff process
Multiply RTO by constant:
RTO = q × RTO
q = 2 is called binary exponential backoff
Which Round-trip Samples?
• If an ack is received for retransmitted segment, there are 2 possibilities:
• Ack is for first transmission
• Ack is for second transmission
• TCP source cannot distinguish 2 cases
• No valid way to calculate RTT:
–From first transmission to ack, or
–From second transmission to ack?
Karn’s Algorithm
The first part of Karn's Algorithm states that when there is retransmission ambiguity, RTT
measurements are ignored instead of being incorporated into SRTT to avoid skewing SRTT
incorrectly. For each retransmission, TCP applies a "backoff factor" to each RTO, so
subsequent retransmission timers are double the previous
• Do not use measured RTT to update SRTT and SDEV
• Calculate backoff RTO when a retransmission occurs
• Use backoff RTO for segments until an ack arrives for a segment that has not been
retransmitted
• Then use Jacobson’s algorithm to calculate RTO
Video Content / Details of website for further learning (if any):
https://en.wikipedia.org/wiki/Karn%27s_algorithm
https://www.extrahop.com/company/blog/2016/retransmission-timeouts-rtos-application-performance-
degradation/
https://www.youtube.com/watch?v=Og4Br2Jog5Y
https://www.geeksforgeeks.org/tcp-timers/
Important Books/Journals for further learning including the page nos.:
William Stallings, “High speed networks and internet”, Pearson Education 2002.Page No: (190-193)

Course Faculty

L-21
LECTURE HANDOUTS

CSE III/VI

Course Name with Code : High Speed Networks-16CSE10

Course Faculty : C.Nithya

Unit : III- TCP and ATM Congestion Control Date of Lecture:

Topic of Lecture: Window management- Performance of TCP over ATM

Introduction :
Sliding windows, a technique also known as windowing, is used by the Internet's Transmission
Control Protocol (TCP) as a method of controlling the flow of packets between two computers or
network hosts. TCP requires that all transmitted data be acknowledged by the receiving host.
Prerequisite knowledge for Complete understanding and learning of Topic:
• Window size
• Asynchronous Transfer Mode
• Fast recovery
• Congestion
Detailed content of the Lecture:
Window Management
• Slow start
• Dynamic window sizing on congestion
• Fast retransmit
• Fast recovery
• Limited transmit
Slow Start
awnd = MIN[ credit, cwnd]
where
awnd = allowed window in segments
cwnd = congestion window in segments
credit = amount of unused credit granted in most recent ack
cwnd = 1 for a new connection and increased by 1 for each ack received, up to a maximum
Dynamic Window Sizing on Congestion
• A lost segment indicates congestion
• Prudent to reset cwsd = 1 and begin slow start process
• May not be conservative enough: ― easy to drive a network into saturation but hard for the net to
recover‖ (Jacobson)
• Instead, use slow start with linear growth in cwnd
Fast Retransmit
• RTO is generally noticeably longer than actual RTT
• If a segment is lost, TCP may be slow to retransmit
• TCP rule: if a segment is received out of order, an ack must be issued immediately for the last in-
order segment
• Fast Retransmit rule: if 4 acks received for same segment, highly likely it was lost, so retransmit
immediately, rather than waiting for timeout
Effect of Slow Start

Fast Recovery
• When TCP retransmits a segment using Fast Retransmit, a segment was assumed lost
• Congestion avoidance measures are appropriate at this point
• E.g., slow-start/congestion avoidance procedure
• This may be unnecessarily conservative since multiple acks indicate segments are getting through
• Fast Recovery: retransmit lost segment, cut cwnd in half, proceed with linear increase of cwnd
• This avoids initial exponential slow-start
Limited Transmit
• If congestion window at sender is small, fast retransmit may not get triggered, e.g., cwnd = 3
• Under what circumstances does sender have small congestion window?
• If the problem is common, why not reduce number of duplicate acks needed to trigger retransmit?
Limited Transmit Algorithm
Sender can transmit new segment when 3 conditions are met:
• Two consecutive duplicate acks are received
• Destination advertised window allows transmission of segment
• Amount of outstanding data after sending is less than or equal to cwnd + 2
Performance of TCP over ATM
• How best to manage TCP’s segment size, window management and congestion control at the same
time as ATM’s quality of service and traffic control policies
• TCP may operate end-to-end over one ATM network, or there may be multiple ATM LANs or
WANs with non-ATM networks
TCP/IP over AAL5/ATM

Topic of Lecture: ABR traffic Management – ABR rate control

Introduction :
• QoS for CBR, VBR based on traffic contract and UPC described previously
• No congestion feedback to source
• Open-loop control
• Not suited to non-real-time applications
– File transfer, web access, RPC, distributed file systems
– No well defined traffic characteristics except PCR
– PCR not enough to allocate resources
• Use best efforts or closed-loop control
Prerequisite knowledge for Complete understanding and learning of Topic:
• Loop Control
• Quality of Service
Detailed content of the Lecture:
Best Efforts
• Share unused capacity between applications
• As congestion goes up:
– Cells are lost
– Sources back off and reduce rate
– Fits well with TCP techniques
– Inefficient
Closed-Loop Control
• Sources share capacity not used by CBR and VBR
• Provide feedback to sources to adjust load
• Avoid cell loss
• Share capacity fairly
• Used for ABR
Characteristics of ABR
ABR connections share available capacity
– Access instantaneous capacity unused by CBR/VBR
– Increases utilization without affecting CBR/VBR QoS
Share used by single ABR connection is dynamic
– Varies between agreed MCR and PCR
Network gives feedback to ABR sources
– ABR flow limited to available capacity
– Buffers absorb excess traffic prior to arrival of feedback
Low cell loss
– Major distinction from UBR
Feedback Mechanisms
Cell transmission rate characterized by:
– Allowable cell rate
Current rate
– Minimum cell rate
Min for ACR
May be zero
– Peak cell rate
Max for ACR
– Initial cell rate
Start with ACR=ICR
Adjust ACR based on feedback
Feedback in resource management (RM) cells
– Cell contains three fields for feedback
Congestion indicator bit (CI)
No increase bit (NI)
Explicit cell rate field (ER)
Source Reaction to Feedback
If CI=1
– Reduce ACR by amount proportional to current ACR but not less than CR
Else if NI=0
– Increase ACR by amount proportional to PCR but not more than PCR
If ACR>ER set ACR<-max[ER,MCR]
Cell Flow on ABR
Two types of cell
– Data & resource management (RM)
Source receives regular RM cells
– Feedback
Bulk of RM cells initiated by source
– One forward RM cell (FRM) per (Nrm-1) data cells
Nrm preset – usually 32
– Each FRM is returned by destination as backwards RM (BRM) cell
– FRM typically CI=0, NI=0 or 1 ER desired transmission rate in range ICR<=ER<=PCR
– Any field may be changed by switch or destination before return
ATM Switch Rate Control Feedback
• EFCI marking
• Explicit forward congestion indication
• Causes destination to set CI bit in ERM
• Relative rate marking
• Switch directly sets CI or NI bit of RM
• If set in FRM, remains set in BRM
• Faster response by setting bit in passing BRM
• Fastest by generating new BRM with bit set
• Explicit rate marking
• Switch reduces value of ER in FRM or BRM
ARB Feedback v TCP ACK
ABR feedback controls rate of transmission
– Rate control
TCP feedback controls window size
– Credit control
ARB feedback from switches or destination
TCP feedback from destination only
RM Cell Format

RM Cell Format Notes

• ATM header has PT=110 to indicate RM cell
• On virtual channel VPI and VCI same as data cells on connection
• On virtual path VPI same, VCI=6
• Protocol id identifies service using RM (ARB=1)
• Message type
– Direction FRM=0, BRM=1
– BECN cell. Source (BN=0) or switch/destination (BN=1)
– CI (=1 for congestion)
– NI (=1 for no increase)
– Request/Acknowledge (not used in ATM forum spec)
Video Content / Details of website for further learning (if any):
https://www.youtube.com/watch?v=oMpj4a6Qhm0
Important Books/Journals for further learning including the page nos.:
William Stallings, ”High speed networks and internet”, Pearson Education 2002,Page No: 293-297

Course Faculty

L-25
LECTURE HANDOUTS

CSE III/VI

Course Name with Code : High Speed Networks-16CSE10

Course Faculty : C.Nithya

Unit : III- TCP and ATM Congestion Control Date of Lecture:

Topic of Lecture: RM Cell Formats

Introduction :
• In asynchronous transfer mode (ATM), a flow control feedback mechanism that communicates
to the originating end-user device to change the transfer characteristics of the connection during
periods of congestion.
• ABR is a best-effort category in which the network attempts to pass the maximum number of
cells, but with no absolute guarantees. See also ABR, best effort, buffer, cell, congestion,
and flow control.
Prerequisite knowledge for Complete understanding and learning of Topic:
• Loop Control
• Quality of Service
Detailed content of the Lecture:
RM Cell Format
• In asynchronous transfer mode (ATM), a flow control feedback mechanism that communicates to
the originating end-user device to change the transfer characteristics of the connection during
periods of congestion.
• The network buffers cells and advises the sender to throttle back on the rate of transmission. The
available bit rate (ABR) class of service makes use of RMCells.
• ABR is a best-effort category in which the network attempts to pass the maximum number of
cells, but with no absolute guarantees. See also ABR, best effort, buffer, cell, congestion,
and flow control.
RM Cell Format Notes
• ATM header has PT=110 to indicate RM cell
• On virtual channel VPI and VCI same as data cells on connection
• On virtual path VPI same, VCI=6
• Protocol id identifies service using RM (ARB=1)
• Message type
– Direction FRM=0, BRM=1
– BECN cell. Source (BN=0) or switch/destination (BN=1)
– CI (=1 for congestion)
– NI (=1 for no increase)
– Request/Acknowledge (not used in ATM forum spec)

Video Content / Details of website for further learning (if any):

https://www.youtube.com/watch?v=oMpj4a6Qhm0
Important Books/Journals for further learning including the page nos.:
William Stallings, ”High speed networks and internet”, Pearson Education 2002,Page No: 293-297

Course Faculty

L-26
LECTURE HANDOUTS

CSE III/VI

Unit : IV- Integrated and Differentiated Services Date of Lecture:

Topic of Lecture: Approach, Components, Services
Introduction:
ISA Approach
• Provision of QoS over IP
• Sharing available capacity when congested
• Router mechanisms
–Routing Algorithms
• Select to minimize delay
–Packet discard
• Causes TCP sender to back off and reduce load
• Enhanced by ISA
Prerequisite knowledge for Complete understanding and learning of Topic:
• Routing
• Quality of Service
• Packet Scheduler
Detailed content of the Lecture:
ISA Approach
• Provision of QoS over IP
• Sharing available capacity when congested
• Router mechanisms
–Routing Algorithms
• Select to minimize delay
–Packet discard
• Causes TCP sender to back off and reduce load
• Enahnced by ISA
Flow
IP packet can be associated with a flow
–Distinguishable stream of related IP packets
–From single user activity
–Requiring same QoS
–E.g. one transport connection or one video stream
–Unidirectional
–Can be more than one recipient
Multicast
–Membership of flow identified by source and destination IP address, port numbers, protocol
type
–IPv6 header flow identifier can be used but is not necessarily equivalent to ISA flow
ISA Functions
Admission control
–For QoS, reservation required for new flow
–RSVP used
Routing algorithm
–Base decision on QoS parameters
Queuing discipline
–Take account of different flow requirements
Discard policy
ISA Implementation in Router
• Background Functions
• Forwarding functions
ISA Components – Background Functions
Reservation Protocol
–RSVP
Admission control
Management agent
–Can use agent to modify traffic control database and direct admission control
Routing protocol
ISA Components – Forwarding
Classifier and route selection
–Incoming packets mapped to classes
Single flow or set of flows with same QoS
–E.g. all video flows
Based on IP header fields
–Determines next hop
Packet scheduler
–Manages one or more queues for each output
–Order queued packets sent
Based on class, traffic control database, current and past activity on outgoing port
–Policing
Video Content / Details of website for further learning (if any):
https://www.slideshare.net/ayyakathir/unit4-29753244
https://www.slideshare.net/ThamerAlamery/intserv-diffserv
https://www.sciencedirect.com/topics/computer-science/integrated-service
Important Books/Journals for further learning including the page nos.:
William Stallings, “High Speed Networks and Internet”, Pearson Education, Second Edition, 2002.
(247-251)

Course Faculty

L-30
LECTURE HANDOUTS

CSE III/VI

Course Name with Code : High Speed Networks-16CSE10

Course Faculty :C.Nithya

Unit : IV- Integrated and Differentiated Services Date of Lecture:

Topic of Lecture: Queuing Discipline

Introduction :
The queuing discipline allows high-priority packets to cut to the front of the line. The idea of the
fair queuing (FQ) discipline is to maintain a separate queue for each flow currently being handled by
the router. The router then services these queues in a round-robin manner.
Prerequisite knowledge for Complete understanding and learning of Topic:
• Queue
• Priority
• Packet Scheduler
Detailed content of the Lecture:
• Population of Customers can be considered either limited (closed systems) or unlimited (open
systems).
• Example of a limited population may be a number of processes to be run (served) by a
computer or a certain number of machines to be repaired by a service man. It is necessary to
take the term "customer" very generally. Customers may be people, machines of various nature,
computer processes, telephone calls, etc.
• Arrival defines the way customers enter the system. Mostly the arrivals are random with
random intervals between two adjacent arrivals. Typically the arrival is described by a random
distribution of intervals also called Arrival Pattern.
• Queue represents a certain number of customers waiting for service (of course the queue may
be empty). Typically the customer being served is considered not to be in the queue. Sometimes
the customers form a queue literally (people waiting in a line for a bank teller). Sometimes the
queue is an abstraction (planes waiting for a runway to land). There are two important
properties of a queue: Maximum Size and Queuing Discipline.
• Maximum Queue Size (also called System capacity) is the maximum number of customers that
may wait in the queue (plus the one(s) being served). Queue is always limited, but some
theoretical models assume an unlimited queue length. If the queue length is limited, some
customers are forced to renounce without being served.
• Queuing Discipline represents the way the queue is organized (rules of inserting and removing
customers to/from the queue). There are these ways:
1) FIFO (First In First Out) also called FCFS (First Come First Serve) - orderly queue.
2) LIFO (Last In First Out) also called LCFS (Last Come First Serve) - stack.
3) SIRO (Serve In Random Order).
4) Priority Queue that may be viewed as a number of queues for various priorities.
5) Many other more complex queuing methods that typically change the customer’s
position in the queue according to the time spent already in the queue, expected service
duration, and/or priority. These methods are typical for computer multi-access systems.
• Most quantitative parameters (like average queue length, average time spent in the system) do
not depend on the queuing discipline. That’s why most models either do not take the queuing
discipline into account at all or assume the normal FIFO queue. In fact the only parameter that
depends on the queuing discipline is the variance (or standard deviation) of the waiting time.
There is this important rule (that may be used for example to verify results of a simulation
experiment):
• The two extreme values of the waiting time variance are for the FIFO queue (minimum) and the
LIFO queue (maximum).
• Theoretical models (without priorities) assume only one queue. This is not considered as a
limiting factor because practical systems with more queues (bank with several tellers with
separate queues) may be viewed as a system with one queue, because the customers always
select the shortest queue. Of course, it is assumed that the customers leave after being served.
Systems with more queues (and more servers) where the customers may be served more times
are called Queuing Networks.
• Service represents some activity that takes time and that the customers are waiting for. Again
take it very generally. It may be a real service carried on persons or machines, but it may be a
CPU time slice, connection created for a telephone call, being shot down for an enemy plane,
etc. Typically a service takes random time.
• Theoretical models are based on random distribution of service duration also called Service
Pattern. Another important parameter is the number of servers. Systems with one server only
are called Single Channel Systems, systems with more servers are called Multi Channel
Systems.
Video Content / Details of website for further learning (if any):
https://www.slideshare.net/ayyakathir/unit4-29753244
https://www.youtube.com/results?search_query=Queuing+Discipline+&pbjreload=10
Important Books/Journals for further learning including the page nos.:
William Stallings, “High Speed Networks and Internet”, Pearson Education, Second Edition, 2002.
(270-274)

Course Faculty

L-31
LECTURE HANDOUTS

CSE III/VI

Course Name with Code : High Speed Networks-16CSE10

Course Faculty :C.Nithya

Unit : IV- Integrated and Differentiated Services Date of Lecture:

Topic of Lecture :FQ – PS

Introduction :
• Fair queuing is a family of scheduling algorithms used in some process and network
schedulers. The algorithm is designed to achieve fairness when a limited resource is shared, for
example to prevent flows with large packets or processes that generate small jobs from
consuming more throughput or CPU time than other flows or processes.
• Processor sharing is a service policy where the customers, clients or jobs are all served
simultaneously, each receiving an equal fraction of the service capacity available. In such a
system all jobs start service immediately.
Prerequisite knowledge for Complete understanding and learning of Topic:
 Queue
 Process Sharing
 Scheduling Algorithm
Detailed content of the Lecture:
Fair Queuing (FQ)
• Multiple queues for each port
–One for each source or flow
–Queues services round robin
–Each busy queue (flow) gets exactly one packet per cycle
–Load balancing among flows
–No advantage to being greedy
Your queue gets longer, increasing your delay
–Short packets penalized as each queue sends one packet per cycle
• A simple variation on basic FIFO queuing is priority queuing.
• The idea is to mark each packet with a priority; the mark could be carried, for example, in the
IP header, as we’ll discuss in a later section. The routers then implement multiple FIFO queues,
one for each priority class.
• The router always transmits packets out of the highest-priority queue if that queue is nonempty
before moving on to the next priority queue.
• Within each priority, packets are still managed in a FIFO manner.
• This idea is a small departure from the best-effort delivery model, but it does not go so far as to
make guarantees to any particular priority class. It just allows high-priority packets to cut to the
front of the line.
FIFO and FQ

Processor Sharing
• Multiple queues as in FQ
• Send one bit from each queue per round
Longer packets no longer get an advantage
• Can work out virtual (number of cycles) start and finish time for a given packet
• However, we wish to send packets, not bits
Video Content / Details of website for further learning (if any):
https://en.wikipedia.org/wiki/Processor_sharing
https://en.wikipedia.org/wiki/Fair_queuing
https://www.youtube.com/watch?v=6MyBum2Njls
Important Books/Journals for further learning including the page nos.:
William Stallings, “High Speed Networks and Internet”, Pearson Education, Second Edition, 2002.
(261-269)

Course Faculty

L-32

LECTURE HANDOUTS

CSE III/VI
Course Name with Code : High Speed Networks-16CSE10

Course Faculty :C.Nithya

Unit : IV- Integrated and Differentiated Services Date of Lecture:

Topic of Lecture: BRFQ
Introduction :
In BRFQ (Bit Round Fair Queuing) a bit-by-bit round robin discipline is followed. In this we set up
multiple queues and transmit one bit from each queue on each round. In this way, longer packets no
longer receive an advantage and each busy source receives exactly the same amount of capacity.
Prerequisite knowledge for Complete understanding and learning of Topic:
• Queuing
• FIFO
• Reservation
Detailed content of the Lecture:
Bit-Round Fair Queuing (BRFQ)
• Compute virtual start and finish time as before
• When a packet finished, the next packet sent is the one with the earliest virtual finish time
• Good approximation to performance of PS
–Throughput and delay converge as time increases
Generalized Processor Sharing (GPS)
• Generalized processor sharing (GPS) is an ideal scheduling algorithm for process schedulers
and network schedulers.
• It is related to the fair-queuing principle which groups packets into classes and shares the
service capacity between them. GPS shares this capacity according to some fixed weights.
• BRFQ cannot provide different capacities to different flows
Enhancement called Weighted fair queue (WFQ)
• From PS, allocate weighting to each flow that determines how many bots are sent during each
round
• If weighted 5, then 5 bits are sent per round
• Gives means of responding to different service requests
• Guarantees that delays do not exceed bounds
Weighted Fair Queue
Weighted fair queueing (WFQ) is a network scheduler scheduling algorithm. WFQ is both a packet-
based implementation of the generalized processor sharing (GPS) policy, and a natural extension
of fair queuing (FQ).
• Emulates bit by bit GPS
• Same strategy as BRFQ
Comparison of FIFO, FQ and BRFQ

Unit : IV- Integrated and Differentiated Services Date of Lecture:

Topic of Lecture: Random Early Detection
Introduction :
• Random early detection (RED), also known as random early discard or random early drop is
a queuing discipline for a network scheduler suited for congestion avoidance.
• In the conventional tail drop algorithm, a router or other network component buffers as many
packets as it can, and simply drops the ones it cannot buffer.
Prerequisite knowledge for Complete understanding and learning of Topic:
• Queuing
• FIFO
• Buffering
Detailed content of the Lecture:
Random Early Detection(RED)
• Random Early Discard (RED) is a data queue control mechanism to improve data utilization
during network congestion. Some radio vendors have made exaggerated claims about its
capacity to improve “radio link utilization.”
• To help control network congestion (i.e., overloading) the Internet Transmission Control
Protocol (TCP) uses a mechanism known as the TCP sliding window, which is designed to
maximize bandwidth usage while avoiding traffic congestion. Under control of the sliding
window, TCP connections have their window size (i.e., share of bandwidth) increased as
acknowledgements (ACKs) are received.
• With multiple connections in play this can reach a point where all bandwidth is consumed,
resulting in network congestion and the dropping (i.e., tail dropping) of frames. At this point,
the sliding window mechanism initiates a simultaneous reduction in window size for all TCP
connections after which, with a return to stability, it ramps up their window size to create an
oscillating traffic pattern. It results in inefficient use of the available traffic bandwidth. This
oscillating behavior is termed “TCP global synchronization.”
Counteracting the Problem
• In order to counteract this problem different queuing control mechanisms have been devised. A
common one is Random Early Discard or Radom Early Detection (RED). In RED, when the
queue exceeds a certain size the network component marks each arriving packet with a
probability that depends on the queue size.
• When the buffer is full, the probability reaches 1 and all incoming packets are dropped. The
chance that the network component notifies a particular sender to reduce its data transmission
rate is proportional to the sender’s share of the bandwidth of the link—an improvement over
tail dropping.
• An issue that was realized early on about the RED algorithm is that it cannot differentiate
between traffic types. A variation of the RED algorithm that addresses this problem is
called Weighted Random Early Detection (WRED). In WRED, the probability of dropping
packets is based on the size of the queue and the traffic flow type (IP precedence).
Improvement Using RED and WRED Algorithms are Modest
• Although some microwave vendors claim to obtain up to 25 percent improvement in “radio link
utilization” with the RED and WRED algorithms, independent studies show that RED
improvement for real data applications is more modest. Also, bear in mind that RED is only
beneficial where the bulk of the traffic is TCP/IP.
• The first study from AT&T Labs and Stanford University using a simple analytical model
showed that although RED may prevent the traffic rate from moving in lockstep, this algorithm
was not enough to prevent the traffic rate from oscillating and wasting throughput for all traffic
flows.
• The study suggests that if the buffer sizes are small, then randomized policies are unlikely to
reduce aggregated periodic behavior.
• A second study from the University of North Carolina concludes that below saturation point (90
percent utilization) there is little difference in performance between RED and tail dropping.
• For loads from 90 percent to 100 percent, RED can be tuned to outperform tail dropping but
only with careful RED parameter settings and degradation in latency response. Analyzing the
results from these and other studies it becomes clear that the claim of 25 percent link
improvement is not realistic.
RED Buffer

RED Algorithm Detail

The general RED algorithm:
procedure REDAlgorithm() is
avg ← average queue size
while packetsarrive
if (minth <= avg AND avg < maxth)
Compute pa with probability pb
mark the arriving packet
else if (maxth < avg)
mark the arriving packet
endif
end REDAlgorithm
RED algorithm contains two main sub algorithms (parts):
1. For computing the average queue size that determines the degree of burstiness that will be allowed
in the routers queue.
2. For calculating the packet-*marking probability that determines how frequently the router *marks
packets, given the current level of congestion. The goal is for the router to *mark packets at fairly even
spaced intervals, in order to avoid biases and to avoid global synchronization, and to *mark packets
sufficiently frequently to control the average queue size.
Video Content / Details of website for further learning (if any):
https://www.networkfashion.net/random-early-detection-red/
Important Books/Journals for further learning including the page nos.:
William Stallings, “High Speed Networks and Internet”, Pearson Education, Second Edition, 2002.
(301-304)

Course Faculty

L-36
LECTURE HANDOUTS

CSE III/VI

Course Name with Code : High Speed Networks-16CSE10

Course Faculty :C.Nithya

Unit : IV- Integrated and Differentiated Services Date of Lecture:

Topic of Lecture: Differentiated Services.
Introduction :
Differentiated services or DiffServ is a computer networking architecture that specifies a simple and
scalable mechanism for classifying and managing network traffic and providing quality of service (QoS)
on modern IP networks.
Prerequisite knowledge for Complete understanding and learning of Topic:
• Mathematical Model
• Queuing
• Network analytics
Detailed content of the Lecture:
Differentiated Services
• In the DiffServ model a packet's "class" can be marked directly in the packet, which contrasts with
the IntServ model where a signaling protocol is required to tell the routers which flows of packets
requires special QoS treatment.
• DiffServ achieves better QoS scalability, while IntServ provides a tighter QoS mechanism for real-
time traffic. These approaches can be complimentary and are not mutually exclusive.
• The DiffServ architecture model (RFC 2475, December 1998) divides traffic into a small number
of classes, and allocates resources on a per-class basis. Because DiffServ has only a few classes of
traffic, a packet's "class" can be marked directly in the packet.
• In the DiffServ model, packets are classified and marked to receive a particular forwarding
treatment (per-hop behavior or PHB) on nodes along their path. Sophisticated classification,
marking, policing, and shaping operations need only be implemented at network boundaries or
hosts, enabling greater scalability than other models of service differentiation.
• Modern data networks carry many different types of services, including voice, video, streaming
music, web pages and email.
• Many of the proposed QoS mechanisms that allowed these services to co-exist were both complex
and failed to scale to meet the demands of the public Internet.
• In December 1998, the IETF published RFC 2474 - Definition of the Differentiated services field
(DS field) in the IPv4 and IPv6 headers, which replaced the IPv4 TOS field with the DS field.
• In the DS field, a range of eight values (Class Selectors) is used for backward compatibility with
the IP precedence specification in the former TOS field.
• Today, DiffServ has largely supplanted TOS and other layer-3 QoS mechanisms, such as integrated
services (IntServ), as the primary architecture routers use to provide QoS.

DS Traffic Conditioner

Per Hop Behaviour – Expedited forwarding

Premium service
– Low loss, delay, jitter; assured bandwidth end-to-end service through domains
– Looks like point to point or leased line
– Difficult to achieve
– Configure nodes so traffic aggregate has well defined minimum departure rate
EF PHB
–Condition aggregate so arrival rate at any node is always less that minimum departure rate
Boundary conditioners
Video Content / Details of website for further learning (if any):
https://en.wikipedia.org/wiki/Differentiated_services
Important Books/Journals for further learning including the page nos.:
William Stallings, “High Speed Networks and Internet”, Pearson Education, Second Edition, 2002. (305-
315)

Course Faculty

Verified by HOD
MUTHAYAMMAL ENGINEERING COLLEGE
(An Autonomous Institution)
(Approved by AICTE, New Delhi, Accredited by NAAC & Affiliated to
Anna University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu

LECTURE HANDOUTS L-37

CSE III/VI

Course Name with Code : High Speed Networks-16CSE10

Course Faculty : C.Nithya

Unit : V- Protocols for QoS Support Date of Lecture:

Topic of Lecture: Multiprotocol Label Switching

Introduction :
• Multiprotocol Label Switching (MPLS) is a routing technique in telecommunications
networks that directs data from one node to the next based on short path labels rather than long
network addresses, thus avoiding complex lookups in a routing table and speeding traffic flows.
• The labels identify virtual links (paths) between distant nodes rather than endpoints. MPLS can
encapsulate packets of various network protocols, hence the "multiprotocol" reference on its
name
Prerequisite knowledge for Complete understanding and learning of Topic:
• Routing
• Paths
• ATM
• Traffic
Detailed content of the Lecture:
Multiprotocol Label Switching:
Routing algorithms provide support for performance goals
• Distributed and dynamic
• React to congestion
• Load balance across network
• Based on metrics
• Develop information that can be used in handling different service needs
• Enhancements provide direct support to IS, DS, RSVP
• Nothing directly improves throughput or delay
• MPLS tries to match ATM QoS support
• Background
• Efforts to marry IP and ATM
• IP switching (Ipsilon)
• Tag switching (Cisco)
• Aggregate route based IP switching (IBM)
• Cascade (IP navigator)
• All use standard routing protocols to define paths between end points
• Assign packets to path as they enter network
• Use ATM switches to move packets along paths
• ATM switching (was) much faster than IP routers
• Use faster technology
• Developments
• IETF working group in 1997, proposed standard 2001
• Routers developed to be as fast as ATM switches
Remove the need to provide both technologies in same network
MPLS does provide new capabilities
– QoS support
– Traffic engineering
– Virtual private networks
– Multiprotocol support
– Connection Oriented QoS Support
– Guarantee fixed capacity for specific applications
– Control latency/jitter
– Ensure capacity for voice
– Provide specific, guaranteed quantifiable SLAs
– Configure varying degrees of QoS for multiple customers
– MPLS imposes connection oriented framework on IP based internets
– Traffic Engineering
– Ability to dynamically define routes, plan resource commitments based on known
– demands and optimize network utilization
– Basic IP allows primitive traffic engineering
– E.g. dynamic routing
– MPLS makes network resource commitment easy
– Able to balance load in face of demand
– Able to commit to different levels of support to meet user traffic
– requirements
– Aware of traffic flows with QoS requirements and predicted demand
– Intelligent re-routing when congested
Video Content / Details of website for further learning (if any):
https://en.wikipedia.org/wiki/Multiprotocol_Label_Switching
https://www.youtube.com/watch?v=ZwK3c3UR4zk
Important Books/Journals for further learning including the page nos.:
William Stallings, “High Speed Networks and Internet”, Pearson Education, Second Edition, 2002.
(361-365)

Course Faculty

Verified by HOD
MUTHAYAMMAL ENGINEERING COLLEGE
(An Autonomous Institution)
(Approved by AICTE, New Delhi, Accredited by NAAC & Affiliated to
Anna University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu
LECTURE HANDOUTS L-41

CSE III/VI

Course Name with Code : High Speed Networks-16CSE10

Course Faculty : C.Nithya

Unit : V- Protocols for QoS Support Date of Lecture:

Topic of Lecture: Operations

MPLS Operation
• Label switched routers capable of switching and routing packets based on label appended to
packet
• Labels define a flow of packets between end points or multicast destinations
• Each distinct flow (forward equivalence class – FEC) has specific path through LSRs defined
Connection oriented
• Each FEC has QoS requirements
• IP header not examined
• Forward based on label value
PLS works by prefixing packets with an MPLS header, containing one or more labels. This is called a
label stack. Each entry in the label stack contains four fields:
• A 20-bit label value. A label with the value of 1 represents the router alert label.
• a 3-bit Traffic Class field for QoS (quality of service) priority and ECN (Explicit Congestion
Notification). Prior to 2009 this field was called EXP.
• a 1-bit bottom of stack flag. If this is set, it signifies that the current label is the last in the stack.
• an 8-bit TTL (time to live) field.

• These MPLS-labeled packets are switched after a label lookup/switch instead of a lookup into
the IP table. As mentioned above, when MPLS was conceived, label lookup and label
switching were faster than a routing table or RIB (Routing Information Base) lookup because
they could take place directly within the switched fabric and avoid having to use the OS.
• The presence of such a label, however, has to be indicated to the router/switch. In the case of
Ethernet frames this is done through the use of EtherType values 0x8847 and 0x8848,
for unicast and multicast connections respectively.
Video Content / Details of website for further learning (if any):
https://towardsdatascience.com/multiprotocol-label-switching-mpls-explained-aac04f3c6e94
https://www.youtube.com/watch?v=LYFUfywvjhs
Important Books/Journals for further learning including the page nos.:
William Stallings, “High Speed Networks and Internet”, Pearson Education, Second Edition, 2002.
(352-353)

Course Faculty

Verified by HOD
MUTHAYAMMAL ENGINEERING COLLEGE
(An Autonomous Institution)
(Approved by AICTE, New Delhi, Accredited by NAAC & Affiliated to
Anna University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu
LECTURE HANDOUTS L-42

CSE III/VI

Course Name with Code : High Speed Networks-16CSE10

Course Faculty : C.Nithya

Unit : V- Protocols for QoS Support Date of Lecture:

Topic of Lecture: Label Stacking-Protocol details

Introduction :
• MPLS works by prefixing packets with an MPLS header, containing one or more labels. This is
called a label stack.
Prerequisite knowledge for Complete understanding and learning of Topic:
• Stack
• Data Flow
• Routers
Detailed content of the Lecture:
Label Stacking
• Packet may carry number of labels
LIFO (stack)
– Processing based on top label
– Any LSR may push or pop label
Unlimited levels
– Allows aggregation of LSPs into single LSP for part of route
– C.f. ATM virtual channels inside virtual paths
– E.g. aggregate all enterprise traffic into one LSP for access provider to handleReduces size of
tables
Label Format Diagram

Time to Live Processing

– Needed to support TTL since IP header not read
– First label TTL set to IP header TTL on entry to MPLS domain
– TTL of top entry on stack decremented at internal LSR
– If zero, packet dropped or passed to ordinary error processing (e.g. ICMP)
– If positive, value placed in TTL of top label on stack and packet forwarded
– At exit from domain, (single stack entry) TTL decremented
– If zero, as above
– If positive, placed in TTL field of Ip header and
– Label Stack
– Appear after data link layer header, before network layer header
– Top of stack is earliest (closest to network layer header)
– Network layer packet follows label stack entry with S=1
– Over connection oriented services
– Topmost label value in ATM header VPI/VCI field
– Facilitates ATM switching
– Top label inserted between cell header and IP header
– In DLCI field of Frame Relay
– Note: TTL problem
Position of MPLS Label Stack

FECs, LSPs, and Labels

• Traffic grouped into FECs
• Traffic in a FEC transits an MLPS domain along an LSP
• Packets identified by locally significant label
• At each LSR, labelled packets forwarded on basis of label.
• LSR replaces incoming label with outgoing label
• Each flow must be assigned to a FEC
• Routing protocol must determine topology and current conditions so LSP can be
• assigned to FEC
– Must be able to gather and use information to support QoS
– LSRs must be aware of LSP for given FEC, assign incoming label to LSP,communicate label to
other LSRs
Explanation – Setup
Labelled switched path established prior to routing and delivery of packets
QoS parameters established along path
– Resource commitment
– Queuing and discard policy at LSR
– Interior routing protocol e.g. OSPF used
– Labels assigned
– Source/destination IP address or network IP address
– Port numbers
– IP protocol id
– Differentiated services codepoint
– IPv6 flow label
Forwarding is simple lookup in predefined table
– Map label to next hop
Can define PHB at an LSR for given FEC
Packets between same end points may belong to different FEC
Video Content / Details of website for further learning (if any):
https://www.dummies.com/programming/networking/juniper/label-stacking-in-mpls-networks/
Important Books/Journals for further learning including the page nos.:
William Stallings, “High Speed Networks and Internet”, Pearson Education, Second Edition, 2002.
(352-353)

Course Faculty

Verified by HOD
MUTHAYAMMAL ENGINEERING COLLEGE
(An Autonomous Institution)
(Approved by AICTE, New Delhi, Accredited by NAAC & Affiliated to
Anna University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu
LECTURE HANDOUTS L-43

CSE III/VI

Course Name with Code : High Speed Networks-16CSE10

Course Faculty : C.Nithya

Unit : V- Protocols for QoS Support Date of Lecture:

Topic of Lecture: RTP – Protocol Architecture

Introduction :
The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video
over IP networks. RTP is used in communication and entertainment systems that involve streaming
media,such as telephony, video teleconference applications including WebRTC, television services and
web-based push-to-talk features.
Prerequisite knowledge for Complete understanding and learning of Topic:
• TCP
• UDP
• Segments
Detailed content of the Lecture:
Real-time Transport Protocol (RTP)
TCP not suited to real time distributed application
– Point to point so not suitable for multicast
– Retransmitted segments arrive out of order
– No way to associate timing with segments
UDP does not include timing information nor any support for real time
applications
Solution is real-time transport protocol RTP
RTP Architecture
• Close coupling between protocol and application layer functionality
• Framework for application to implement single protocol
• Application level framing
• Integrated layer processing
• Application Level Framing
• Recovery of lost data done by application rather than transport layer
• Application may accept less than perfect delivery
• Real time audio and video
• Inform source about quality of delivery rather than retransmit
• Source can switch to lower quality
• Application may provide data for retransmission
• Sending application may recompute lost values rather than storing them
• Sending application can provide revised values
• Can send new data to ―fix‖ consequences of loss
• Lower layers deal with data in units provided by application
• Application data units (ADU)
Integrated Layer Processing
– Adjacent layers in protocol stack tightly coupled
– Allows out of order or parallel functions from different layers

Video Content / Details of website for further learning (if any):

https://en.wikipedia.org/wiki/Real-time_Transport_Protocol
https://www.youtube.com/watch?v=CttYdLpGeGA
https://www.youtube.com/watch?v=bEYwHvLuy80
Important Books/Journals for further learning including the page nos.:
William Stallings, “High Speed Networks and Internet”, Pearson Education, Second Edition, 2002.
(353-355)

Course Faculty

Verified by HOD
MUTHAYAMMAL ENGINEERING COLLEGE
(An Autonomous Institution)
(Approved by AICTE, New Delhi, Accredited by NAAC & Affiliated to
Anna University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu
LECTURE HANDOUTS L-44

CSE III/VI

Course Name with Code : High Speed Networks-16CSE10

Course Faculty : C.Nithya

Unit : V- Protocols for QoS Support Date of Lecture:

Topic of Lecture: Data Transfer Protocol

Introduction :
• RTP – short for Real-time Transport Protocol defines a standard packet format for delivering audio
and video over the Internet. It is defined in RFC 1889.
• It was developed by the Audio Video Transport Working group and was first published in 1996.
Prerequisite knowledge for Complete understanding and learning of Topic:
• TCP
• UDP
• Network Layer
Detailed content of the Lecture:
Data Transfer Protocol
• RTP applications can use the Transmission Control Protocol (TCP), but most use the User
Datagram protocol (UDP) instead because UDP allows for faster delivery of data.
RTP Data Transfer Protocol
• Transport of real time data among number of participants in a session, defined by:
• RTP Port number
• UDP destination port number if using UDP
• RTP Control Protocol (RTCP) port number
• Destination port address used by all participants for RTCP transfer
• IP addresses
• Multicast or set of unicast
• Multicast Support
• Each RTP data unit includes:
• Source identifier
• Timestamp
• Payload format
• Relays
• Intermediate system acting as receiver and transmitter for given protocol layer
• Mixers
– Receives streams of RTP packets from one or more sources
– Combines streams
– Forwards new stream
– Translators
Video Content / Details of website for further learning (if any):
https://www.youtube.com/watch?v=rMuhgZegNio
https://www.youtube.com/results?search_query=+RTCP.
Important Books/Journals for further learning including the page nos.:
William Stallings, “High Speed Networks and Internet”, Pearson Education, Second Edition, 2002 (365-
368)

Course Faculty

Verified by HOD
MUTHAYAMMAL ENGINEERING COLLEGE
(An Autonomous Institution)
(Approved by AICTE, New Delhi, Accredited by NAAC & Affiliated to
Anna University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu
LECTURE HANDOUTS L-45

CSE III/VI

Course Name with Code : High Speed Networks-16CSE10

Course Faculty : C.Nithya

Unit : V- Protocols for QoS Support Date of Lecture:

Topic of Lecture: RTCP

Introduction :
• The RTP Control Protocol (RTCP) is a sister protocol of the Real-time Transport Protocol (RTP).
Its basic functionality and packet structure is defined in RFC 3550. RTCP provides out-of-
band statistics and control information for an RTP session.
• It partners with RTP in the delivery and packaging of multimedia data, but does not transport any
media data itself.
Prerequisite knowledge for Complete understanding and learning of Topic:
• TCP
• UDP
• Network Layer
Detailed content of the Lecture:
RTP Control Protocol (RTCP)
• RTP is for user data
• RTCP is multicast provision of feedback to sources and session participants
• Uses same underlying transport protocol (usually UDP) and different port number
• RTCP packet issued periodically by each participant to other session members
RTP Header
RTCP Functions
• QoS and congestion control
• Identification
• Session size estimation and scaling
• Session control
RTCP Transmission
Number of separate RTCP packets bundled in single UDP datagram
– Sender report
– Receiver report
– Source description
– Goodbye
– Application specific
Packet Fields (All Packets)
Version (2 bit) currently version 2
Padding (1 bit) indicates padding bits at end of control information, with number of octets as last octet of
padding
Count (5 bit) of reception report blocks in SR or RR, or source items in SDES or BYE
Packet type (8 bit)
Length (16 bit) in 32 bit words minus 1
In addition Sender and receiver reports have:
–Synchronization Source Identifier
Packet Fields (Sender Report) Sender Information Block
• NTP timestamp: absolute wall clock time when report sent
• RTP Timestamp: Relative time used to create timestamps in RTP packets
• Sender’s packet count (for this session)
• Sender’s octet count (for this session)
Packet Fields (Sender Report) Reception Report Block
• SSRC_n (32 bit) identifies source refered to by this report block
• Fraction lost (8 bits) since previous SR or RR
• Cumulative number of packets lost (24 bit) during this session
• Extended highest sequence number received (32 bit)
–Least significant 16 bits is highest RTP data sequence number received from SSRC_n
–Most significant 16 bits is number of times sequence number has wrapped to zero
• Interarrival jitter (32 bit)
• Last SR timestamp (32 bit)
• Delay since last SR (32 bit)
Receiver Report
• Same as sender report except:
– Packet type field has different value
– No sender information block
Source Description Packet
• Used by source to give more information
• 32 bit header followed by zero or more additional information chunks
E.g.:
 0 END End of SDES list
 1 CNAME Canonical name
 2 NAME Real user name of source
 3 EMAIL Email address
Goodbye (BYE)
• Indicates one or more sources no linger active
–Confirms departure rather than failure of network
Application Defined Packet
• Experimental use
• For functions & features that are application specific
Video Content / Details of website for further learning (if any):
https://www.youtube.com/watch?v=rMuhgZegNio
https://www.youtube.com/results?search_query=+RTCP.
Important Books/Journals for further learning including the page nos.:
William Stallings, “High Speed Networks and Internet”, Pearson Education, Second Edition, 2002 (365-
368)

Course Faculty

Verified by HOD

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