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net

Unit I
Frame relay Networks
Frame Relay often is described as a streamlined version of X.25, offering fewer of the
robust capabilities, such as windowing and retransmission of last data that are offered in
X.25.
Frame Relay Devices
Devices attached to a Frame Relay WAN fall into the following two general categories:

Data terminal equipment (DTE)

Data circuit-terminating equipment (DCE)

DTEs generally are considered to be terminating equipment for a specific network and
typically are located on the premises of a customer. In fact, they may be owned by the
customer. Examples of DTE devices are terminals, personal computers, routers, and
bridges.
DCEs are carrier-owned internetworking devices. The purpose of DCE equipment is to
provide clocking and switching services in a network, which are the devices that actually
transmit data through the WAN. In most cases, these are packet switches. Figure 10-1
shows the relationship between the two categories of devices.

Standard Frame Relay Frame

Standard Frame Relay frames consist of the fields illustrated in Figure 10-4.
Figure Five Fields Comprise the Frame Relay Frame

Each frame relay PDU consists of the following fields:

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1. Flag Field. The flag is used to perform high level data link synchronization which
indicates the beginning and end of the frame with the unique pattern 01111110.
To ensure that the 01111110 pattern does not appear somewhere inside the frame,
bit stuffing and destuffing procedures are used.
2. Address Field. Each address field may occupy either octet 2 to 3, octet 2 to 4, or
octet 2 to 5, depending on the range of the address in use. A two-octet address
field comprising the EA=ADDRESS FIELD EXTENSION BITS and the
C/R=COMMAND/RESPONSE BIT.
3. DLCI-Data Link Connection Identifier Bits. The DLCI serves to identify the
virtual connection so that the receiving end knows which information connection
a frame belongs to. Note that this DLCI has only local significance. A single
physical channel can multiplex several different virtual connections.
4. FECN, BECN, DE bits. These bits report congestion:
o FECN=Forward Explicit Congestion Notification bit
o BECN=Backward Explicit Congestion Notification bit
o DE=Discard Eligibility bit
5. Information Field. A system parameter defines the maximum number of data
bytes that a host can pack into a frame. Hosts may negotiate the actual maximum
frame length at call set-up time. The standard specifies the maximum information
field size (supportable by any network) as at least 262 octets. Since end-to-end
protocols typically operate on the basis of larger information units, frame relay
recommends that the network support the maximum value of at least 1600 octets
in order to avoid the need for segmentation and reassembling by end-users.
Frame Check Sequence (FCS) Field. Since one cannot completely ignore the bit errorrate of the medium, each switching node needs to implement error detection to avoid
wasting bandwidth due to the transmission of erred frames. The error detection
mechanism used in frame relay uses the cyclic redundancy check (CRC) as its basis.
Congestion-Control Mechanisms
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. Frame Relay implements two congestionnotification mechanisms:

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.

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Frame Relay versus X.25
The design of X.25 aimed to provide error-free delivery over links with high error-rates.
Frame relay takes advantage of the new links with lower error-rates, enabling it to
eliminate many of the services provided by X.25. The elimination of functions and fields,
combined with digital links, enables frame relay to operate at speeds 20 times greater
than X.25.
X.25 specifies processing at layers 1, 2 and 3 of the OSI model, while frame relay
operates at layers 1 and 2 only. This means that frame relay has significantly less
processing to do at each node, which improves throughput by an order of magnitude.
X.25 prepares and sends packets, while frame relay prepares and sends frames. X.25
packets contain several fields used for error and flow control, none of which frame relay
needs. The frames in frame relay contain an expanded address field that enables frame
relay nodes to direct frames to their destinations with minimal processing .
X.25 has a fixed bandwidth available. It uses or wastes portions of its bandwidth as the
load dictates. Frame relay can dynamically allocate bandwidth during call setup
negotiation at both the physical and logical channel level.

Asynchronous Transfer Mode (ATM)

Asynchronous Transfer Mode (ATM) is an International Telecommunication UnionTelecommunications Standards Section (ITU-T) standard for cell relay wherein
information for multiple service types, such as voice, video, or data, is conveyed in small,
fixed-size cells. ATM networks are connection-oriented.
ATM is a cell-switching and multiplexing technology that combines the benefits of
circuit switching (guaranteed capacity and constant transmission delay) with those of
packet switching (flexibility and efficiency for intermittent traffic). It provides scalable
bandwidth from a few megabits per second (Mbps) to many gigabits per second (Gbps).
Because of its asynchronous nature, ATM is more efficient than synchronous
technologies, such as time-division multiplexing (TDM).
With TDM, each user is assigned to a time slot, and no other station can send in that time
slot. If a station has much data to send, it can send only when its time slot comes up, even
if all other time slots are empty. However, if a station has nothing to transmit when its
time slot comes up, the time slot is sent empty and is wasted. Because ATM is
asynchronous, time slots are available on demand with information identifying the source
of the transmission contained in the header of each ATM cell.
ATM transfers information in fixed-size units called cells. Each cell consists of 53
octets, or bytes. The first 5 bytes contain cell-header information, and the remaining 48
contain the payload (user information). Small, fixed-length cells are well suited to

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transferring voice and video traffic because such traffic is intolerant of delays that result
from having to wait for a large data packet to download, among other things. Figure
illustrates the basic format of an ATM cell. Figure :An ATM Cell Consists of a Header
and Payload Data

ATM Protocol architecture:

ATM is almost similar to cell relay and packets witching using X.25and framerelay.like
packet switching and frame relay,ATM involves the transfer of data in discrete
pieces.also,like packet switching and frame relay ,ATM allows multiple logical
connections to multiplexed over a single physical interface. in the case of ATM,the
information flow on each logical connection is organised into fixed-size packets, called
cells. ATM is a streamlined protocol with minimal error and flow control capabilities
:this reduces the overhead of processing ATM cells and reduces the number of overhead
bits required with each cell, thus enabling ATM to operate at high data rates.the use of
fixed-size cells simplifies the processing required at each ATM node,again supporting the
use of ATM at high data rates. The ATM architecture uses a logical model to describe the
functionality that it supports. ATM functionality corresponds to the physical layer and
part of the data link layer of the OSI reference model. . the protocol referencce model
shown makes reference to three separate planes:
user plane provides for user information transfer ,along with associated controls
(e.g.,flow control ,error control).
control plane performs call control and connection control functions.
management plane includes plane management ,which performs management function
related to a system as a whole and provides coordination between all the planes ,and layer
management which performs management functions relating to resource and parameters
residing in its protocol entities .
The ATM reference model is composed of the following ATM layers:
Physical layerAnalogous to the physical layer of the OSI reference model, the
ATM physical layer manages the medium-dependent transmission.

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ATM layerCombined with the ATM adaptation layer, the ATM layer is roughly
analogous to the data link layer of the OSI reference model. The ATM layer is
responsible for the simultaneous sharing of virtual circuits over a physical link (cell
multiplexing) and passing cells through the ATM network (cell relay). To do this, it uses
the VPI and VCI information in the header of each ATM cell.
ATM adaptation layer (AAL)Combined with the ATM layer, the AAL is roughly
analogous to the data link layer of the OSI model. The AAL is responsible for isolating
higher-layer protocols from the details of the ATM processes. The adaptation layer
prepares user data for conversion into cells and segments the data into 48-byte cell
payloads.
Finally, the higher layers residing above the AAL accept user data, arrange it into
packets, and hand it to the AAL. Figure :illustrates the ATM reference model.

Structure of an ATM cell

An ATM cell consists of a 5 byte header and a 48 byte payload. The payload size of 48
bytes was a compromise between the needs of voice telephony and packet networks,
obtained by a simple averaging of the US proposal of 64 bytes and European proposal of
32, said by some to be motivated by a European desire not to need echo-cancellers on
national trunks.
ATM defines two different cell formats: NNI (Network-network interface) and UNI
(User-network interface). Most ATM links use UNI cell format.

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Diagram of the UNI ATM Cell

Diagram of the NNI ATM Cell

7
GFC

7
VPI

4 3
VPI

VPI

VCI

4 3

VPI

VCI

VCI

VCI

VCI

PT

CLP

HEC

VCI

PT

CLP

(48

bytes)

HEC

Payload

(48

bytes)

Payload

GFC = Generic Flow Control (4 bits) (default: 4-zero bits)

VPI = Virtual Path Identifier (8 bits UNI) or (12 bits NNI)
VCI = Virtual channel identifier (16 bits)
PT = Payload Type (3 bits)
CLP = Cell Loss Priority (1-bit)
HEC = Header Error Correction (8-bit CRC, polynomial = X8 + X2 + X + 1)
The PT field is used to designate various special kinds of cells for Operation and
Management (OAM) purposes, and to delineate packet boundaries in some AALs.
Several of ATM's link protocols use the HEC field to drive a CRC-Based Framing
algorithm, which allows the position of the ATM cells to be found with no overhead
required beyond what is otherwise needed for header protection. The 8-bit CRC is used to
correct single-bit header errors and detect multi-bit header errors. When multi-bit header
errors are detected, the current and subsequent cells are dropped until a cell with no
header errors is found.
In a UNI cell the GFC field is reserved for a local flow control/submultiplexing system
between users. This was intended to allow several terminals to share a single network
connection, in the same way that two ISDN phones can share a single basic rate ISDN
connection. All four GFC bits must be zero by default.The NNI cell format is almost
identical to the UNI format, except that the 4-bit GFC field is re-allocated to the VPI
field, extending the VPI to 12 bits. Thus, a single NNI ATM interconnection is capable of

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addressing almost 212 VPs of up to almost 216 VCs each (in practice some of the VP and
VC numbers are reserved).

A Virtual Channel (VC) denotes the transport of ATM cells which have the same
unique identifier, called the Virtual Channel Identifier (VCI). This identifier is encoded in
the cell header. A virtual channel represents the basic means of communication between
two end-points, and is analogous to an X.25 virtual circuit.
A Virtual Path (VP) denotes the transport of ATM cells belonging to virtual channels
which share a common identifier, called the Virtual Path Identifier (VPI), which is also
encoded in the cell header. A virtual path, in other words, is a grouping of virtual
channels which connect the same end-points. This two layer approach results in improved
network performance. Once a virtual path is set up, the addition/removal of virtual
channels is straightforward

ATM Classes of Services

ATM is connection oriented and allows the user to specify the resources required on a per-connection
basis (per SVC) dynamically. There are the five classes of service defined for ATM (as per ATM Forum
UNI 4.0 specification). The QoS parameters for these service classes are summarized in Table 1.

Service Class

Quality of Service Parameter

This class is used for emulating circuit switching. The cell rate is
constant bit rate constant with time. CBR applications are quite sensitive to cell-delay
(CBR)
variation. Examples of applications that can use CBR are telephone
traffic (i.e., nx64 kbps), videoconferencing, and television.
This class allows users to send traffic at a rate that varies with time
variable bit rate
depending on the availability of user information. Statistical
non-real
time
multiplexing is provided to make optimum use of network resources.
(VBRNRT)
Multimedia e-mail is an example of VBRNRT.

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This class is similar to VBRNRT but is designed for applications that
variable bit rate
are sensitive to cell-delay variation. Examples for real-time VBR are
real time (VBR
voice with speech activity detection (SAD) and interactive compressed
RT)
video.
This class of ATM services provides rate-based flow control and is
aimed at data traffic such as file transfer and e-mail. Although the
standard does not require the cell transfer delay and cell-loss ratio to
available bit rate be guaranteed or minimized, it is desirable for switches to minimize
(ABR)
delay and loss as much as possible. Depending upon the state of
congestion in the network, the source is required to control its rate.
The users are allowed to declare a minimum cell rate, which is
guaranteed to the connection by the network.
unspecified
rate (UBR)

bit This class is the catch-all, other class and is widely used today for
TCP/IP.

Technical
Parameter

Definition

CLR is the percentage of cells not delivered at their destination

cell loss ratio
because they were lost in the network due to congestion and
(CLR)
buffer overflow.
The delay experienced by a cell between network entry and
cell
transfer exit points is called the CTD. It includes propagation delays,
delay (CTD)
queuing delays at various intermediate switches, and service
times at queuing points.
cell
delay CDV is a measure of the variance of the cell transfer delay.
variation
High variation implies larger buffering for delay-sensitive
(CDV)
traffic such as voice and video.
peak cell rate The maximum cell rate at which the user will transmit. PCR is
(PCR)
the inverse of the minimum cell inter-arrival time.
sustained
cell This is the average rate, as measured over a long interval, in
rate (SCR)
the order of the connection lifetime.
This parameter determines the maximum burst that can be sent
burst tolerance at the peak rate. This is the bucket-size parameter for the
(BT)
enforcement algorithm that is used to control the traffic
entering the network.
Benefits of ATM

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The benefits of ATM are the following:

high performance via hardware switching

dynamic bandwidth for bursty traffic
class-of-service support for multimedia
scalability in speed and network size
common LAN/WAN architecture
opportunities for simplification via VC architecture
international standards compliance

ATM Adaptation Layers (AAL)

The use of Asynchronous Transfer Mode (ATM) technology and services creates the
need for an adaptation layer in order to support information transfer protocols, which are
not based on ATM. This adaptation layer defines how to segment and reassemble higherlayer packets into ATM cells, and how to handle various transmission aspects in the
ATM layer.
Examples of services that need adaptations are Gigabit Ethernet, IP, Frame Relay,
SONET/SDH, UMTS/Wireless, etc.
The main services provided by AAL (ATM Adaptation Layer) are:

Segmentation and reassembly

Handling of transmission errors
Handling of lost and misinserted cell conditions
Timing and flow control

The following ATM Adaptation Layer protocols (AALs) have been defined by the ITUT. It is meant that these AALs will meet a variety of needs. The classification is based on
whether a timing relationship must be maintained between source and destination,
whether the application requires a constant bit rate, and whether the transfer is connection
oriented or connectionless.

AAL Type 1 supports constant bit rate (CBR), synchronous, connection oriented
traffic. Examples include T1 (DS1), E1, and x64 kbit/s emulation.
AAL Type 2 supports time-dependent Variable Bit Rate (VBR-RT) of
connection-oriented, synchronous traffic. Examples include Voice over ATM.
AAL2 is also widely used in wireless applications due to the capability of
multiplexing voice packets from different users on a single ATM connection.
AAL Type 3/4 supports VBR, data traffic, connection-oriented, asynchronous
traffic (e.g. X.25 data) or connectionless packet data (e.g. SMDS traffic) with an
additional 4-byte header in the information payload of the cell. Examples include
Frame Relay and X.25.

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AAL Type 5 is similar to AAL 3/4 with a simplified information header scheme.
This AAL assumes that the data is sequential from the end user and uses the
Payload Type Indicator (PTI) bit to indicate the last cell in a transmission.
Examples of services that use AAL 5 are classic IP over ATM, Ethernet Over
ATM, SMDS, and LAN Emulation (LANE). AAL 5 is a widely used ATM
adaptation layer protocol. This protocol was intended to provide a streamlined
transport facility for higher- layer protocols that are connection oriented.

AAL 5 was introduced to:

reduce protocol processing overhead.

reduce transmission overhead.
ensure adaptability to existing transport protocols.

T AAL1 PDU
The structure of the AAL1 PDU is given in the following illustration:
SN

SNP

CSI

SC

CRC

1 bit

3 bits 3 bits

EPC

SAR PDU Payload

1 bit

47 bytes

AAL1 PDU
SN

Sequence number. Numbers the stream of SAR PDUs of a CPCS PDU (modulo 16). The
sequence number is comprised of the CSI and the SN.
CSI

Convergence sublayer indicator. Used for residual time stamp for clocking.
SC

Sequence count. The sequence number for the entire CS PDU, which is generated by the
Convergence Sublayer.
SNP

Sequence number protection. Comprised of the CRC and the EPC.

CRC

Cyclic redundancy check calculated over the SAR header.

EPC

Even parity check calculated over the CRC.

SAR

PDU

payload

47-byte user information field.

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AAL2
AAL2 provides bandwidth-efficient transmission of low-rate, short and variable packets
in delay sensitive applications. It supports VBR and CBR. AAL2 also provides for
variable payload within cells and across cells. AAL type 2 is subdivided into the
Common Part Sublayer (CPS ) and the Service Specific Convergence Sublayer (SSCS ).
AAL2 CPS Packet

The CPS packet consists of a 3 octet header followed by a payload. The structure of the
AAL2 CPS packet is shown in the following illustration.
CID
8 bits

LI
6 bits

UUI HEC Information payload

5 bits 5 bits 1-45/64 bytes

AAL2 CPS packet

CID Channelidentification.
LI
Length indicator. This is the length of the packet payload associated with each individual
user. Value is one less than the packet payload and has a default value of 45 bytes (may
be set to 64 bytes).
UUI
User-to-user indication. Provides a link between the CPS and an appropriate SSCS that
satisfies the higher layer application
HEC
Header error control.
AAL2
The structure of the AAL2 SAR PDU is given in the following illustration.
Start
OSF SN
6 bits 1 bit

field
P
1 bit

CPS-PDU payload
AAL2 PDU payload

PAD
0-47
bytes

AAL2 CPS PDU

OSF
Offset field. Identifies the location of the start of the next CPS packet within the CPSPDU.

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SN
Sequence number. Protects data integrity.
P
Parity. Protects the start field from errors.
SAR
Information field of the SAR PDU.

PDU

payload

PAD
Padding.
AAL2 SSCS Packet
The SSCS conveys narrowband calls consisting of voice, voiceband data or circuit mode
data. SSCS packets are transported as CPS packets over AAL2 connections. The CPS
packet contains a SSCS payload. There are 3 SSCS packet types.
Type 1 Unprotected; this is used by default.
Type 2 Partially protected.
Type 3 Fully protected: the entire payload is protected by a 10-bit CRC which is
computed as for OAM cells. The remaining 2 bits of the 2-octet trailer consist of the
message type field.
AAL2 SSCS Type 3 Packets:
The type 3 packets are used for the following:

Dialled digits
Channel associated signalling bits
Facsimile demodulated control data
Alarms
User state control operations.

The following illustration gives the general sturcture of AAL2 SSCS Type 3 PDUs. The
format varies and each message has its own format according to the actual message type.
Redundancy

Time
stamp

Message
dependant
information

Message
type

CRC10

14

16

10 bits

AAL2 SSCS Type 3 PDU

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Redundancy
Packets are sent 3 times to ensure error correction. The value in this field signifies the
transmission number.
Time
stamp
Counters packet delay variation and allows a receiver to accurately reproduce the relative
timing of successive events separated by a short interval.
Message
dependant
Packet content that varies, depending on the message type.

information

Message
The message type code.

type

CRC-10
The 10-bit CRC.
AAL3/4
AAL3/4 consists of message and streaming modes. It provides for point-to-point and
point-to-multipoint (ATM layer) connections. The Convergence Sublayer (CS) of the
ATM Adaptation Layer (AAL) is divided into two parts: service specific (SSCS ) and
common part (CPCS ). This is illustrated in the following diagram:
AAL3/4 packets are used to carry computer data, mainly SMDS traffic.
AAL3/4 CPCS PDU
The functions of the AAL3/4 CPCS include connectionless network layer (Class D),
meaning no need for an SSCS; and frame relaying telecommunication service in Class C.
The CPCS PDU is composed of the following fields:
Header

Info

Trailer

CPI

Btag

Basize CPCS
SDU

Pad

Etag Length

0-65535 0-3

2 bytes

AAL3/4 CPCS PDU

CPI
Message type. Set to zero when the BAsize and Length fields are encoded in bytes.

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Btag
Beginning tag. This is an identifier for the packet. It is repeated as the Etag.
BAsize
Buffer allocation size. Size (in bytes) that the receiver has to allocate to capture all the
data.
CPCS
Variable information field up to 65535 bytes.

SDU

PAD
Padding field which is used to achieve 32-bit alignment of the length of the packet.
0
All-zero.
Etag
End tag. Must be the same as Btag.
Length
Must be the same as BASize.
AAL3/4 SAR PDU
The structure of the AAL3/4 SAR PDU is illustrated below:
ST

SN

MID

Information

LI

CRC

10

352

10 bits

44 bytes

2-byte trailer

2-byte header
48 bytes
AAL3/4 SAR PDU

ST
Segment type. Values may be as follows:
SN
Sequence number. Numbers the stream of SAR PDUs of a CPCS PDU (modulo 16).
MID
Multiplexing identification. This is used for multiplexing several AAL3/4 connections
over one ATM link.

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Information
This field has a fixed length of 44 bytes and contains parts of CPCS PDU.
LI
Length indication. Contains the length of the SAR SDU in bytes, as follows:
CRC
Cyclic redundancy check.
Functions of AAL3/4 SAR include identification of SAR SDUs; error indication and
handling; SAR SDU sequence continuity; multiplexing and demultiplexing.
AAL5 The type 5 adaptation layer is a simplified version of AAL3/4. It also consists of
message and streaming modes, with the CS divided into the service specific and common
part. AAL5 provides point-to-point and point-to-multipoint (ATM layer) connections.
AAL5 is used to carry computer data such as TCP/IP. It is the most popular AAL and is
sometimes referred to as SEAL (simple and easy adaptation layer).
AAL5 CPCS PDU
The AAL5 CPCS PDU is composed of the following fields:
Trailer

Info
CPCS payload
0-65535

Pad
0-47

UU CPI Length CRC

1 1
2
4 bytes

AAL5 CPCS PDU

CPCS
The actual information that is sent by the user. Note that the information comes before
any length indication (as opposed to AAL3/4 where the amount of memory required is
known in advance).
Pad
Padding bytes to make the entire packet (including control and CRC) fit into a 48-byte
boundary.
UU
CPCS user-to-user indication to transfer one byte of user information.
CPI
Common part indicator is a filling byte (of value 0). This field is to be used in the future
for layer management message indication.

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Length
Length of the user information without the Pad.
CRC
CRC-32. Used to allow identification of corrupted transmission.
AAL5 SAR PDU The structure of the AAL5 CS PDU is as follows:
Information
PAD
UU CPI Length CRC-32
1-48

0-47

4 bytes

8-byte trailer
AAL5 SAR PDU

High-Speed LANs
Emergence of High-Speed LANs
2 Significant trends
Computing power of PCs continues to grow rapidly
Network computing
Examples of requirements
Centralized server farms
Power workgroups
High-speed local backbone
Classical Ethernet
Bus topology LAN
10 Mbps
CSMA/CD medium access control protocol
2 problems:
A transmission from any station can be received by all stations
How to regulate transmission

Solution to First Problem

Data transmitted in blocks called frames:
User data
Frame header containing unique address of destination station
CSMA/CD
Carrier Sense Multiple Access/ Carrier Detection
If the medium is idle, transmit.
If the medium is busy, continue to listen until the channel is idle, then transmit

immediately.
If a collision is detected during transmission, immediately cease transmitting.

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After a collision, wait a random amount of time, then attempt to transmit again (repeat

from step 1).

Medium Options at 10Mbps

<data rate> <signaling method> <max length>
10Base5
10 Mbps
50-ohm coaxial cable bus
Maximum segment length 500 meters
10Base-T
Twisted pair, maximum length 100 meters

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Star
point)

topology

(hub

or

multipoint

repeater

at

central

Hubs and Switches

Hub
Transmission from a station received by central hub and retransmitted on all outgoing
lines
Only one transmission at a time
Layer 2 Switch
Incoming frame switched to one outgoing line
Many transmissions at same time

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Bridge
Frame handling done in software
Analyze and forward one frame at a time
Store-and-forward
Layer 2 Switch
Frame handling done in hardware
Multiple data paths and can handle multiple frames at a time
Can do cut-through
Layer 2 Switches
Flat address space
Broadcast storm
Only one path between any 2 devices
Solution 1: subnetworks connected by routers
Solution 2: layer 3 switching, packet-forwarding logic in hardware
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Benefits of 10 Gbps Ethernet over ATM

No expensive, bandwidth consuming conversion between Ethernet packets and ATM
cells
Network is Ethernet, end to end
IP plus Ethernet offers QoS and traffic policing capabilities approach that of ATM
Wide variety of standard optical interfaces for 10 Gbps Ethernet
Fibre Channel
2 methods of communication with processor:
I/O channel
Network communications
Fibre channel combines both
Simplicity and speed of channel communications
Flexibility and interconnectivity of network communications

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I/O channel
Hardware based, high-speed, short distance
Direct point-to-point or multipoint communications link
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Data type qualifiers for routing payload
Link-level constructs for individual I/O operations
Protocol specific specifications to support e.g. SCSI
Fibre Channel Network-Oriented Facilities
Full multiplexing between multiple destinations
Peer-to-peer connectivity between any pair of ports
Internetworking with other connection technologies
Fibre Channel Requirements
Full duplex links with 2 fibres/link
100 Mbps 800 Mbps
Distances up to 10 km
Small connectors
high-capacity
Greater connectivity than existing multidrop channels
Broad availability
Support for multiple cost/performance levels
Support for multiple existing interface command sets
Fibre Channel Protocol Architecture
FC-0 Physical Media
FC-1 Transmission Protocol
FC-2 Framing Protocol
FC-3 Common Services
FC-4 Mapping

Wireless LAN Requirements

Throughput
Number of nodes
Connection to backbone
Service area
Battery power consumption
Transmission robustness and security
Collocated network operation
License-free operation
Handoff/roaming
Dynamic configuration

IEEE 802.11 Services

Association
Reassociation
Disassociation
Authentication
Privacy

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Unit II
Queing analysis
In queueing theory, a queueing model is used to approximate a real queueing
situation or system, so the queueing behaviour can be analysed
mathematically. Queueing models allow a number of useful steady state
performance measures to be determined, including:

the
the
the
the
the

average number in the queue, or the system,

average time spent in the queue, or the system,
statistical distribution of those numbers or times,
probability the queue is full, or empty, and
probability of finding the system in a particular state.

These performance measures are important as issues or problems caused by

queueing situations are often related to customer dissatisfaction with service or
may be the root cause of economic losses in a business. Analysis of the
relevant queueing models allows the cause of queueing issues to be identified
and the impact of any changes that might be wanted to be assessed.
Notation

Queueing models can be represented using Kendall's notation:

A/B/S/K/N/Disc
where:

A is the interarrival time distribution

B is the service time distribution
S is the number of servers
K is the system capacity
N is the calling population
Disc is the service discipline assumed
Some standard notation for distributions (A or B) are:

M for a Markovian (exponential) distribution

E for an Erlang distribution with phases
D for Deterministic (constant)
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G for General distribution

PH for a Phase-type distribution
Models
Construction and analysis

Queueing models are generally constructed to represent the steady state of a

queueing system, that is, the typical, long run or average state of the system.
As a consequence, these are stochastic models that represent the probability
that a queueing system will be found in a particular configuration or state.
A general procedure for constructing and analysing such queueing models is:
1. Identify the parameters of the system, such as the arrival rate, service time,
Queue capacity, and perhaps draw a diagram of the system.
2. Identify the system states. (A state will generally represent the integer number
of customers, people, jobs, calls, messages, etc. in the system and may or may
not be limited.)
3. Draw a state transition diagram that represents the possible system states and
identify the rates to enter and leave each state. This diagram is a representation
of a Markov chain.
4. Because the state transition diagram represents the steady state situation
between state there is a balanced flow between states so the probabilities of
being in adjacent states can be related mathematically in terms of the arrival
and service rates and state probabilities.
5. Express all the state probabilities in terms of the empty state probability, using
the inter-state transition relationships.
6. Determine the empty state probability by using the fact that all state
probabilities always sum to 1.
Whereas specific problems that have small finite state models are often able to
be analysed numerically, analysis of more general models, using calculus,
yields useful formulae that can be applied to whole classes of problems.

Single-server queue
Single-server queues are, perhaps, the most commonly encountered queueing
situation in real life. One encounters a queue with a single server in many
situations, including business (e.g. sales clerk), industry (e.g. a production
line), transport (e.g. a bus, a taxi rank, an intersection), telecommunications
(e.g. Telephone line), computing (e.g. processor sharing). Even where there are
multiple servers handling the situation it is possible to consider each server
individually as part of the larger system, in many cases. (e.g A supermarket
checkout has several single server queues that the customer can select from.)

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Consequently, being able to model and analyse a single server queue's
behaviour is a particularly useful thing to do.

Poisson arrivals and service

M/M/1// represents a single server that has unlimited queue capacity and
infinite calling population, both arrivals and service are Poisson (or random)
processes, meaning the statistical distribution of both the inter-arrival times
and the service times follow the exponential distribution. Because of the
mathematical nature of the exponential distribution, a number of quite simple
relationships are able to be derived for several performance measures based on
knowing the arrival rate and service rate.
This is fortunate because, an M/M/1 queuing model can be used to
approximate many queuing situations.

Poisson arrivals and general service

M/G/1// represents a single server that has unlimited queue capacity and
infinite calling population, while the arrival is still Poisson process, meaning
the statistical distribution of the inter-arrival times still follow the exponential
distribution, the distribution of the service time does not. The distribution of
the service time may follow any general statistical distribution, not just
exponential. Relationships are still able to be derived for a (limited) number of
performance measures if one knows the arrival rate and the mean and variance
of the service rate. However the derivations a generally more complex.
A number of special cases of M/G/1 provide specific solutions that give broad
insights into the best model to choose for specific queueing situations because
they permit the comparison of those solutions to the performance of an M/M/1
model.

Multiple-servers queue
Multiple (identical)-servers queue situations are frequently encountered in
telecommunications or a customer service environment. When modelling these
situations care is needed to ensure that it is a multiple servers queue, not a
network of single server queues, because results may differ depending on how
the queuing model behaves.
One observational insight provided by comparing queuing models is that a
single queue with multiple servers performs better than each server having
their own queue and that a single large pool of servers performs better than two
or more smaller pools, even though there are the same total number of servers
in the system.

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One simple example to prove the above fact is as follows: Consider a system
having 8 input lines, single queue and 8 servers.The output line has a capacity
of 64 kbit/s. Considering the arrival rate at each input as 2 packets/s. So, the
total arrival rate is 16 packets/s. With an average of 2000 bits per packet, the
service rate is 64 kbit/s/2000b = 32 packets/s. Hence, the average response
time of the system is 1/(-) = 1/(32-16) = 0.0667 sec. Now, consider a second
system with 8 queues, one for each server. Each of the 8 output lines has a
capacity of 8 kbit/s. The calculation yields the response time as 1/(-) = 1/(42) = 0.5 sec. And the average waiting time in the queue in the first case is /(1) = 0.25, while in the second case is 0.03125.

Infinitely many servers

While never exactly encountered in reality, an infinite-servers (e.g. M/M/)
model is a convenient theoretical model for situations that involve storage or
delay, such as parking lots, warehouses and even atomic transitions. In these
models there is no queue, as such, instead each arriving customer receives
service. When viewed from the outside, the model appears to delay or store
each customer for some time.
Queueing System Classification

With Little's Theorem, we have developed some basic understanding of a

queueing system. To further our understanding we will have to dig deeper into
characteristics of a queueing system that impact its performance. For example,
queueing requirements of a restaurant will depend upon factors like:

How do customers arrive in the restaurant? Are customer arrivals more during
lunch and dinner time (a regular restaurant)? Or is the customer traffic more
uniformly distributed (a cafe)?
How much time do customers spend in the restaurant? Do customers typically
leave the restaurant in a fixed amount of time? Does the customer service time
vary with the type of customer?
How many tables does the restaurant have for servicing customers?
The above three points correspond to the most important characteristics of a
queueing system. They are explained below:
Arrival Process

Service Process

The
probability
density
distribution
that
determines the customer arrivals in the system.
In a messaging system, this refers to the message
arrival probability distribution.
The
probability
density
distribution
that
determines the customer service times in the

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Number
Servers

of

system.
In a messaging system, this refers to the message
transmission time distribution. Since message
transmission is directly proportional to the length
of the message, this parameter indirectly refers to
the message length distribution.
Number of servers available to service the
customers.
In a messaging system, this refers to the number
of links between the source and destination nodes.

Based on the above characteristics, queueing systems can be classified by the

following convention:
A/S/n
Where A is the arrival process, S is the service process and n is the number of
servers. A and S are can be any of the following:
M (Markov)
D (Deterministic)
G (General)

Exponential probability density

All customers have the same value
Any arbitrary probability distribution

Examples of queueing systems that can be defined with this convention are:

M/M/1: This is the simplest queueing system to analyze. Here the arrival and
service time are negative exponentially distributed (poisson process). The
system consists of only one server. This queueing system can be applied to a
wide variety of problems as any system with a very large number of
independent customers can be approximated as a Poisson process. Using a
Poisson process for service time however is not applicable in many
applications and is only a crude approximation. Refer to M/M/1 Queueing
System for details.
M/D/n: Here the arrival process is poisson and the service time distribution is
deterministic. The system has n servers. (e.g. a ticket booking counter with n
cashiers.) Here the service time can be assumed to be same for all customers)
G/G/n: This is the most general queueing system where the arrival and service
time processes are both arbitrary. The system has n servers. No analytical
solution is known for this queueing system.

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Markovian arrival processes
In queuing theory, Markovian arrival processes are used to model the arrival
customers to queue.
Some of the most common include the Poisson process, Markovian arrival
process and the batch Markovian arrival process.
Markovian arrival processes has two processes. A continuous-time Markov
process j(t), a Markov process which is generated by a generator or rate
matrix, Q. The other process is a counting process N(t), which has state space
(where
is the set of all natural numbers). N(t) increases
every time there is a transition in j(t) which marked.

Poisson process
The Poisson arrival process or Poisson process counts the number of arrivals,
each of which has a exponentially distributed time between arrival. In the most
general case this can be represented by the rate matrix,

M arkov arrival process

The Markov arrival process (MAP) is a generalisation of the Poisson process
by having non-exponential distribution sojourn between arrivals. The
homogeneous case has rate matrix,
Little's law
In queueing theory, Little's result, theorem, lemma, or law says:
The average number of customers in a stable system (over some time interval),
N, is equal to their average arrival rate, , multiplied by their average time in
the system, T, or:

Although it looks intuitively reasonable, it's a quite remarkable result, as it

implies that this behavior is entirely independent of any of the detailed
probability distributions involved, and hence requires no assumptions about the
schedule according to which customers arrive or are serviced, or whether they
are served in the order in which they arrive.
It is also a comparatively recent result - it was first proved by John Little, an
Institute Professor and the Chair of Management Science at the MIT Sloan
School of Management, in 1961.

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Handily his result applies to any system, and particularly, it applies to systems
within systems. So in a bank, the queue might be one subsystem, and each of
the tellers another subsystem, and Little's result could be applied to each one,
as well as the whole thing. The only requirement is that the system is stable -it can't be in some transition state such as just starting up or just shutting down.
Mathematical formalization of Little's theorem

Let (t) be to some system in the interval [0, t]. Let (t) be the number of
departures from the same system in the interval [0, t]. Both (t) and (t) are
integer valued increasing functions by their definition. Let Tt be the mean time
spent in the system (during the interval [0, t]) for all the customers who were in
the system during the interval [0, t]. Let Nt be the mean number of customers
in the system over the duration of the interval [0, t].
If the following limits exist,

and, further, if = then Little's theorem holds, the limit

exists and is given by Little's theorem,

Ideal Performance

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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

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Explicit congestion signaling

Frame Relay reduces network overhead by implementing simple congestionnotification 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. Frame Relay
implements two congestion-notification mechanisms:

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.
The FECN bit is part of the Address field in the Frame Relay frame header. The
FECN mechanism is initiated when a DTE device sends Frame Relay frames into the
network. 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 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.
The BECN bit is part of the Address field in the Frame Relay frame header. DCE
devices set the value of the BECN bit to 1 in frames traveling in the opposite direction
of frames with their FECN bit set. This informs the receiving DTE device that a
particular path through the network is congested. The DTE device then can relay this

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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.

Traffic Management
Considerations

in

Congested Network

Some

Fairness

Various flows should suffer equally

Last-in-first-discarded may not be fair

Quality of Service (QoS)

Flows treated differently, based on need

Voice, video: delay sensitive, loss insensitive

File transfer, mail: delay insensitive, loss sensitive

Interactive computing: delay and loss sensitive

Reservations

Policing: excess traffic discarded or handled on best-effort basis

Frame Relay Congestion Control

Minimize frame discard
Maintain QoS (per-connection bandwidth)

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Minimize monopolization of network

Simple to implement, little overhead
Minimal additional network traffic
Resources distributed fairly
Limit spread of congestion
Operate effectively regardless of flow
Have minimum impact other systems in network
Minimize variance in QoS

Congestion Avoidance with Explicit Signaling

Two general strategies considered:
Hypothesis 1: Congestion always occurs slowly, almost always at egress nodes

forward explicit congestion avoidance

Hypothesis 2: Congestion grows very quickly in internal nodes and requires
quick action

backward explicit congestion avoidance

Explicit Signaling Response
Network Response

each frame handler monitors its queuing behavior and takes action

use FECN/BECN bits

some/all connections notified of congestion

User (end-system) Response

receipt of BECN/FECN bits in frame

BECN at sender: reduce transmission rate

FECN at receiver: notify peer (via LAPF or higher layer) to restrict

flow

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Frame Relay Traffic Rate Management Parameters
Committed Information Rate (CIR)

Average data rate in bits/second that the network agrees to support for a
connection
Data Rate of User Access Channel (Access Rate)

Fixed rate link between user and network (for network access)
Committed Burst Size (Bc)

Maximum data over an interval agreed to by network

Excess Burst Size (Be)

Maximum data, above Bc, over an interval that network will attempt to
transfer

Relationship of Congestion Parameters

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Unit III

TCP Flow Control

Uses a form of sliding window
Differs from mechanism used in LLC, HDLC, X.25, and others:
Decouples acknowledgement of received data units from granting
permission to send more
TCPs flow control is known as a credit allocation scheme:
Each transmitted octet is considered to have a sequence number
TCP Header Fields for Flow Control
Sequence number (SN) of first octet in data segment
Acknowledgement number (AN)
Window (W)
Acknowledgement contains AN = i, W = j:
Octets through SN = i - 1 acknowledged
Permission is granted to send W = j more octets,
i.e., octets i through i + j - 1
TCP Credit Allocation Mechanism

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Credit Allocation is Flexible

Suppose last message B issued was AN = i, W = j
To increase credit to k (k > j) when no new data, B issues AN = i, W = k
To acknowledge segment containing m octets (m < j), B issues AN = i + m, W = j m
Flow Control Perspectives

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Credit

Policy
Receiver needs a policy for how much credit to give sender
Conservative approach: grant credit up to limit of available buffer space
May limit throughput in long-delay situations
Optimistic approach: grant credit based on expectation of freeing space before
data arrives
Effect of Window Size
W = TCP window size (octets)
R = Data rate (bps) at TCP source
D = Propagation delay (seconds)

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After TCP source begins transmitting, it takes D seconds for first octet to arrive,
and D seconds for acknowledgement to return
TCP source could transmit at most 2RD bits, or RD/4 octets
Normalized Throughput S
1

W > RD / 4

4W/RD

W < RD / 4

S =

Window Scale Parameter

Complicating Factors
Multiple TCP connections are multiplexed over same network interface, reducing
R and efficiency
For multi-hop connections, D is the sum of delays across each network plus
delays at each router
If source data rate R exceeds data rate on one of the hops, that hop will be a
bottleneck
Lost segments are retransmitted, reducing throughput. Impact depends on
retransmission policy
Retransmission Strategy
TCP relies exclusively on positive acknowledgements and retransmission on
acknowledgement timeout
There is no explicit negative acknowledgement
Retransmission required when:

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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, cant tell whether acknowledgement is response to original
transmission or retransmission
Network conditions may change suddenly

Adaptive Retransmission Timer

Average Round-Trip Time (ARTT)
K+1
ARTT(K + 1) =
1
RTT(i)
K+1
i=1

=
K+1

K ART(K) +
K+1

RTT(K + 1)

RFC 793 Exponential Averaging

Smoothed Round-Trip Time (SRTT)
SRTT(K + 1) = SRTT(K)
+ (1 ) SRTT(K + 1)
The older the observation, the less it is counted in the average.

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RFC 793 Retransmission Timeout
RTO(K + 1) =
Min(UB, Max(LB, SRTT(K + 1)))
UB, LB: prechosen fixed upper and lower bounds
Example values for , :
0.8 < < 0.9

1.3 < < 2.0

Implementation Policy Options

Send
Deliver
Accept
In-order
In-window
Retransmit
First-only
Batch
individual
Acknowledge
immediate
cumulative

TCP Congestion Control

Dynamic routing can alleviate congestion by spreading load more evenly
But only effective for unbalanced loads and brief surges in traffic
Congestion can only be controlled by limiting total amount of data entering network
ICMP source Quench message is crude and not effective
RSVP may help but not widely implemented

TCP Congestion Control is Difficult

IP is connectionless and stateless, with no provision for detecting or controlling
congestion
TCP only provides end-to-end flow control
No cooperative, distributed algorithm to bind together various TCP entities

TCP Flow and Congestion Control

The rate at which a TCP entity can transmit is determined by rate of incoming ACKs to
previous segments with new credit
Rate of Ack arrival determined by round-trip path between source and destination
Bottleneck may be destination or internet

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Sender cannot tell which
Only the internet bottleneck can be due to congestion

TCP Segment Pacing

TCP Flow and Congestion Control

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Retransmission Timer Management
Three Techniques to calculate retransmission timer (RTO):
RTT Variance Estimation
Exponential RTO Backoff
Karns Algorithm

RTT Variance Estimation

(Jacobsons 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
Jacobsons Algorithm
SRTT(K + 1) = (1 g) SRTT(K) + g RTT(K + 1)
SERR(K + 1) = RTT(K + 1) SRTT(K)
SDEV(K + 1) = (1 h) SDEV(K) + h |SERR(K + 1)|
RTO(K + 1) = SRTT(K + 1) + f SDEV(K + 1)
g = 0.125
h = 0.25
f = 2 or f = 4 (most current implementations use f = 4)

Two Other Factors

Jacobsons algorithm can significantly improve TCP performance, but:
What RTO to use for retransmitted segments?
ANSWER: exponential RTO backoff algorithm
Which round-trip samples to use as input to Jacobsons algorithm?
ANSWER: Karns algorithm
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:

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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?

Karns Algorithm
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 Jacobsons algorithm to calculate RTO

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
Effect of Slow Start

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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
Illustration of Slow Start and Congestion Avoidance

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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
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?
Is the problem common?
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
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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 TCPs segment size, window management and congestion
control
at the same time as ATMs 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

Performance of TCP over UBR

Buffer capacity at ATM switches is a critical parameter in assessing TCP throughput
performance
Insufficient buffer capacity results in lost TCP segments and retransmissions
Effect of Switch Buffer Size
Data rate of 141 Mbps
End-to-end propagation delay of 6 s
IP packet sizes of 512 octets to 9180
TCP window sizes from 8 Kbytes to 64 Kbytes
ATM switch buffer size per port from 256 cells to 8000
One-to-one mapping of TCP connections to ATM virtual circuits
TCP sources have infinite supply of data ready
Observations
If a single cell is dropped, other cells in the same IP datagram are unusable, yet ATM
network forwards these useless cells to destination

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Smaller buffer increase probability of dropped cells
Larger segment size increases number of useless cells transmitted if a single cell
dropped
Partial Packet and Early Packet Discard
Reduce the transmission of useless cells
Work on a per-virtual circuit basis
Partial Packet Discard
If a cell is dropped, then drop all subsequent cells in that segment (i.e., look for cell with
SDU type bit set to one)
Early Packet Discard
When a switch buffer reaches a threshold level, preemptively discard all cells in a
segment
Selective Drop
Ideally, N/V cells buffered for each of the V virtual circuits
W(i) = N(i) = N(i) V
N/V
N
If N > R and W(i) > Z
then drop next new packet on VC i
Z is a parameter to be chosen
ATM Switch Buffer Layout

Fair Buffer Allocation

More aggressive dropping of packets as congestion increases
Drop new packet when:
N > R and W(i) > Z B R
N-R
TCP over ABR

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Good performance of TCP over UBR can be achieved with minor adjustments to switch

mechanisms
This reduces the incentive to use the more complex and more expensive ABR service
Performance and fairness of ABR quite sensitive to some ABR parameter settings
Overall, ABR does not provide significant performance over simpler and less expensive
UBR-EPD or UBR-EPD-FBA

Traffic and Congestion Control in ATM Networks

Introduction
Control needed to prevent switch buffer overflow
High speed and small cell size gives different problems from other networks
Limited number of overhead bits
ITU-T specified restricted initial set
I.371
ATM forum Traffic Management Specification 41
Overview
Congestion problem
Framework adopted by ITU-T and ATM forum
Control schemes for delay sensitive traffic
Voice & video
Not suited to bursty traffic
Traffic control
Congestion control
Bursty traffic
Available Bit Rate (ABR)
Guaranteed Frame Rate (GFR)
Requirements for ATM Traffic and Congestion Control
Most packet switched and frame relay networks carry non-real-time bursty data
No need to replicate timing at exit node
Simple statistical multiplexing
User Network Interface capacity slightly greater than average of channels
Congestion control tools from these technologies do not work in ATM
Problems with ATM Congestion Control
Most traffic not amenable to flow control
Voice & video can not stop generating
Feedback slow
Small cell transmission time v propagation delay
Wide range of applications
From few kbps to hundreds of Mbps
Different traffic patterns
Different network services
High speed switching and transmission
Volatile congestion and traffic control
Key Performance Issues-Latency/Speed Effects
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E.g. data rate 150Mbps
Takes (53 x 8 bits)/(150 x 106) =2.8 x 10-6 seconds to insert a cell
Transfer time depends on number of intermediate switches, switching time and
propagation delay. Assuming no switching delay and speed of light propagation,
round trip delay of 48 x 10-3 sec across USA
A dropped cell notified by return message will arrive after source has transmitted
N further cells
N=(48 x 10-3 seconds)/(2.8 x 10-6 seconds per cell)
=1.7 x 104 cells = 7.2 x 106 bits
i.e. over 7 Mbits
Cell Delay Variation
For digitized voice delay across network must be small
Rate of delivery must be constant
Variations will occur
Dealt with by Time Reassembly of CBR cells (see next slide)
Results in cells delivered at CBR with occasional gaps due to dropped cells
Subscriber requests minimum cell delay variation from network provider
Increase data rate at UNI relative to load
Increase resources within network
Time Reassembly of CBR Cells

Network Contribution to Cell Delay Variation

In packet switched network
Queuing effects at each intermediate switch
Processing time for header and routing
Less for ATM networks
Minimal processing overhead at switches
Fixed cell size, header format
No flow control or error control processing
ATM switches have extremely high throughput

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Congestion can cause cell delay variation

Build up of queuing effects at switches
Total load accepted by network must be controlled
Cell Delay Variation at UNI
Caused by processing in three layers of ATM model
See next slide for details
None of these delays can be predicted
None follow repetitive pattern
So, random element exists in time interval between reception by ATM stack and
transmission
ATM Traffic-Related Attributes
Six service categories (see chapter 5)
Constant bit rate (CBR)
Real time variable bit rate (rt-VBR)
Non-real-time variable bit rate (nrt-VBR)
Unspecified bit rate (UBR)
Available bit rate (ABR)
Guaranteed frame rate (GFR)
Characterized by ATM attributes in four categories
Traffic descriptors
QoS parameters
Congestion
Other
Traffic Parameters
Traffic pattern of flow of cells
Intrinsic nature of traffic
Source traffic descriptor
Modified inside network
Connection traffic descriptor
Source Traffic Descriptor
Peak cell rate
Upper bound on traffic that can be submitted
Defined in terms of minimum spacing between cells T
PCR = 1/T
Mandatory for CBR and VBR services
Sustainable cell rate
Upper bound on average rate
Calculated over large time scale relative to T
Required for VBR
Enables efficient allocation of network resources between VBR sources
Only useful if SCR < PCR
Maximum burst size
Max number of cells that can be sent at PCR
If bursts are at MBS, idle gaps must be enough to keep overall rate below
SCR

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Required for VBR
Minimum cell rate
Min commitment requested of network
Can be zero
Used with ABR and GFR
ABR & GFR provide rapid access to spare network capacity up to PCR
PCR MCR represents elastic component of data flow
Shared among ABR and GFR flows
Maximum frame size
Max number of cells in frame that can be carried over GFR connection
Only relevant in GFR
Connection Traffic Descriptor
Includes source traffic descriptor plus:Cell delay variation tolerance
Amount of variation in cell delay introduced by network interface and UNI
Bound on delay variability due to slotted nature of ATM, physical layer
overhead and layer functions (e.g. cell multiplexing)
Represented by time variable
Conformance definition
Specify conforming cells of connection at UNI
Enforced by dropping or marking cells over definition
Quality of Service Parameters-maxCTD
Cell transfer delay (CTD)
Time between transmission of first bit of cell at source and reception of last
bit at destination
Typically has probability density function (see next slide)
Fixed delay due to propagation etc.
Cell delay variation due to buffering and scheduling
Maximum cell transfer delay (maxCTD)is max requested delay for connection
Fraction of cells exceed threshold
Discarded or delivered late
Peak-to-peak CDV & CLR
Peak-to-peak Cell Delay Variation
Remaining (1-) cells within QoS
Delay experienced by these cells is between fixed delay and maxCTD
This is peak-to-peak CDV
CDVT is an upper bound on CDV
Cell loss ratio
Ratio of cells lost to cells transmitted

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Cell Transfer Delay PDF

Congestion Control Attributes

Only feedback is defined
ABR and GFR
Actions taken by network and end systems to regulate traffic submitted
ABR flow control
Adaptively share available bandwidth
Other Attributes
Behaviour class selector (BCS)
Support for IP differentiated services (chapter 16)
Provides different service levels among UBR connections
Associate each connection with a behaviour class
May include queuing and scheduling
Minimum desired cell rate
Traffic Management Framework
Objectives of ATM layer traffic and congestion control
Support QoS for all foreseeable services
Not rely on network specific AAL protocols nor higher layer application
specific protocols
Minimize network and end system complexity
Maximize network utilization
Timing Levels
Cell insertion time
Round trip propagation time
Connection duration
Long term
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Traffic Control and Congestion Functions

Traffic Control Strategy

Determine whether new ATM connection can be accommodated
Agree performance parameters with subscriber
Traffic contract between subscriber and network
This is congestion avoidance
If it fails congestion may occur
Invoke congestion control
Traffic Control
Resource management using virtual paths
Connection admission control
Usage parameter control
Selective cell discard
Traffic shaping
Explicit forward congestion indication
Resource Management Using Virtual Paths
Allocate resources so that traffic is separated according to service characteristics
Virtual path connection (VPC) are groupings of virtual channel connections
(VCC)
Applications
User-to-user applications
VPC between UNI pair

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No knowledge of QoS for individual VCC
User checks that VPC can take VCCs demands
User-to-network applications
VPC between UNI and network node
Network aware of and accommodates QoS of VCCs
Network-to-network applications
VPC between two network nodes
Network aware of and accommodates QoS of VCCs

Resource Management Concerns

Cell loss ratio
Max cell transfer delay
Peak to peak cell delay variation
All affected by resources devoted to VPC
If VCC goes through multiple VPCs, performance depends on consecutive VPCs
and on node performance
VPC performance depends on capacity of VPC and traffic characteristics
of VCCs
VCC related function depends on switching/processing speed and priority
VCCs and VPCs Configuration

Allocation of Capacity to VPC

Aggregate peak demand
May set VPC capacity (data rate) to total of VCC peak rates
Each VCC can give QoS to accommodate peak demand
VPC capacity may not be fully used
Statistical multiplexing

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VPC capacity >= average data rate of VCCs but < aggregate peak demand
Greater CDV and CTD
May have greater CLR
More efficient use of capacity
For VCCs requiring lower QoS
Group VCCs of similar traffic together

Connection Admission Control

User must specify service required in both directions
Category
Connection traffic descriptor
Source traffic descriptor
CDVT
Requested conformance definition
QoS parameter requested and acceptable value
Network accepts connection only if it can commit resources to support requests
Procedures to Set Traffic Control Parameters

Cell Loss Priority

Two levels requested by user
Priority for individual cell indicated by CLP bit in header
If two levels are used, traffic parameters for both flows specified
High priority CLP = 0
All traffic CLP = 0 + 1
May improve network resource allocation
Usage Parameter Control
UPC
Monitors connection for conformity to traffic contract
Protect network resources from overload on one connection
Done at VPC or VCC level

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VPC level more important
Network resources allocated at this level

Location of UPC Function

Peak Cell Rate Algorithm

How UPC determines whether user is complying with contract
Control of peak cell rate and CDVT
Complies if peak does not exceed agreed peak
Subject to CDV within agreed bounds
Generic cell rate algorithm
Leaky bucket algorithm

Generic Cell Rate Algorithm

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Virtual Scheduling Algorithm

Leaky Bucket Algorithm

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Continuous Leaky Bucket Algorithm

Sustainable Cell Rate Algorithm

Operational definition of relationship between sustainable cell rate and burst
tolerance
Used by UPC to monitor compliance
Same algorithm as peak cell rate

UPC Actions
Compliant cell pass, non-compliant cells discarded
If no additional resources allocated to CLP=1 traffic, CLP=0 cells C
If two level cell loss priority cell with:
CLP=0 and conforms passes
CLP=0 non-compliant for CLP=0 traffic but compliant for CLP=0+1 is
tagged and passes
CLP=0 non-compliant for CLP=0 and CLP=0+1 traffic discarded

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CLP=1 compliant for CLP=0+1 passes
CLP=1 non-compliant for CLP=0+1 discarded
Possible Actions of UPC

Explicit Forward Congestion Indication

Essentially same as frame relay
If node experiencing congestion, set forward congestion indication is cell headers
Tells users that congestion avoidance should be initiated in this direction
User may take action at higher level
ABR Traffic Management
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
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 (chapter 12)
Inefficient
Cells dropped causing re-transmission
Closed-Loop Control
Sources share capacity not used by CBR and VBR
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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

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ATM Switch

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
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
Flow of Data and RM Cells

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

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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)

ARB Parameters

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ARB Capacity Allocation

ATM switch must perform:
Congestion control
Monitor queue length
Fair capacity allocation
Throttle back connections using more than fair share
ATM rate control signals are explicit
TCP are implicit
Increasing delay and cell loss
Congestion Control Algorithms-Binary Feedback
Use only EFCI, CI and NI bits
Switch monitors buffer utilization
When congestion approaches, binary notification
Set EFCI on forward data cells or CI or NI on FRM or BRM
Three approaches to which to notify
Single FIFO queue
Multiple queues
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Fair share notification

Single FIFO Queue

When buffer use exceeds threshold (e.g. 80%)
Switch starts issuing binary notifications
Continues until buffer use falls below threshold
Can have two thresholds
One for start and one for stop
Stops continuous on/off switching
Biased against connections passing through more switches
Multiple Queues
Separate queue for each VC or group of VCs
Separate threshold on each queue
Only connections with long queues get binary notifications
Fair
Badly behaved source does not affect other VCs
Delay and loss behaviour of individual VCs separated
Can have different QoS on different VCs
Fair Share

Selective feedback or intelligent marking

Try to allocate capacity dynamically
E.g.
fairshare =(target rate)/(number of connections)
Mark any cells where CCR>fairshare
Explicit Rate Feedback Schemes
Compute fair share of capacity for each VC
Determine current load or congestion
Compute explicit rate (ER) for each connection and send to source
Three algorithms
Enhanced proportional rate control algorithm
EPRCA
Explicit rate indication for congestion avoidance
ERICA
Congestion avoidance using proportional control
CAPC
Enhanced Proportional Rate Control Algorithm(EPRCA
Switch tracks average value of current load on each connection
Mean allowed cell rate (MARC)
MACR(I)=(1-)*(MACR(I-1) + *CCR(I)
CCR(I) is CCR field in Ith FRM
Typically =1/16
Bias to past values of CCR over current
Gives estimated average load passing through switch
If congestion, switch reduces each VC to no more than DPF*MACR

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DPF=down pressure factor, typically 7/8
ER<-min[ER, DPF*MACR]
Load Factor
Adjustments based on load factor
LF=Input rate/target rate
Input rate measured over fixed averaging interval
Target rate slightly below link bandwidth (85 to 90%)
LF>1 congestion threatened
VCs will have to reduce rate
Explicit Rate Indication for Congestion Avoidance (ERICA)
Attempt to keep LF close to 1
Define:
fairshare = (target rate)/(number of connections)
VCshare = CCR/LF
= (CCR/(Input Rate)) *(Target Rate)
ERICA selectively adjusts VC rates
Total ER allocated to connections matches target rate
Allocation is fair
ER = max[fairshare, VCshare]
VCs whose VCshare is less than their fairshare get greater increase
Congestion Avoidance Using Proportional Control (CAPC)
If LF<1 fairshare<- fairshare*min[ERU,1+(1-LF)*Rup]
If LF>1 fairshare<- fairshare*min[ERU,1-(1-LF)*Rdn]
ERU>1, determines max increase
Rup between 0.025 and 0.1, slope parameter
Rdn, between 0.2 and 0.8, slope parameter
ERF typically 0.5, max decrease in allottment of fair share
If fairshare < ER value in RM cells, ER<-fairshare
Simpler than ERICA
Can show large rate oscillations if RIF (Rate increase factor) too high
Can lead to unfairness
GRF Overview
Simple as UBR from end system view
End system does no policing or traffic shaping
May transmit at line rate of ATM adaptor
Modest requirements on ATM network
No guarantee of frame delivery
Higher layer (e.g. TCP) react to congestion causing dropped frames
User can reserve cell rate capacity for each VC
Application can send at min rate without loss
Network must recognise frames as well as cells
If congested, network discards entire frame
All cells of a frame have same CLP setting
CLP=0 guaranteed delivery, CLP=1 best efforts
GFR Traffic Contract

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Peak cell rate PCR

Minimum cell rate MCR
Maximum burst size MBS
Maximum frame size MFS
Cell delay variation tolerance CDVT

Mechanisms for supporting Rate Guarantees

Tagging and policing
Buffer management
Scheduling

Tagging and Policing

Tagging identifies frames that conform to contract and those that dont
CLP=1 for those that dont
Set by network element doing conformance check
May be network element or source showing less important frames
Get lower QoS in buffer management and scheduling
Tagged cells can be discarded at ingress to ATM network or subsequent
switch
Discarding is a policing function
Buffer Management
Treatment of cells in buffers or when arriving and requiring buffering
If congested (high buffer occupancy) tagged cells discarded in preference to
untagged
Discard tagged cell to make room for untagged cell
May buffer per-VC
Discards may be based on per queue thresholds
Scheduling
Give preferential treatment to untagged cells
Separate queues for each VC
Per VC scheduling decisions
E.g. FIFO modified to give CLP=0 cells higher priority
Scheduling between queues controls outgoing rate of VCs
Individual cells get fair allocation while meeting traffic contract

Components of GFR Mechanism

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GFR Conformance Definition

UPC function
UPC monitors VC for traffic conformance
Tag or discard non-conforming cells
Frame conforms if all cells in frame conform
Rate of cells within contract
Generic cell rate algorithm PCR and CDVT specified for
connection
All cells have same CLP
Within maximum frame size (MFS)
QoS Eligibility Test
Test for contract conformance
Discard or tag non-conforming cells
Looking at upper bound on traffic
Determine frames eligible for QoS guarantee
Under GFR contract for VC
Looking at lower bound for traffic
Frames are one of:
Nonconforming: cells tagged or discarded
Conforming ineligible: best efforts
Conforming eligible: guaranteed delivery

Simplified Frame Based GCRA

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Unit IV
Integrated and Differentiated Services
Introduction
New additions to Internet increasing traffic
High volume client/server application
Web
Graphics
Real time voice and video
Need to manage traffic and control congestion
IEFT standards
Integrated services
Collective service to set of traffic demands in domain
Limit demand & reserve resources
Differentiated services
Classify traffic in groups
Different group traffic handled differently

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Integrated Services Architecture (ISA)
IPv4 header fields for precedence and type of service usually ignored
ATM only network designed to support TCP, UDP and real-time traffic
May need new installation
Need to support Quality of Service (QoS) within TCP/IP
Add functionality to routers
Means of requesting QoS

Internet Traffic Elastic

Can

adjust to changes in delay and throughput

common TCP and UDP application
E-Mail insensitive to delay changes
FTP User expect delay proportional to file size
Sensitive to changes in throughput
SNMP delay not a problem, except when caused by congestion
Web (HTTP), TELNET sensitive to delay
Not per packet delay total elapsed time
E.g. web page loading time
For small items, delay across internet dominates
For large items it is throughput over connection
Need some QoS control to match to demand
Internet Traffic Inelastic
Does not easily adapt to changes in delay and throughput
Real time traffic
Throughput
Minimum may be required
Delay
E.g. stock trading
Jitter - Delay variation
More jitter requires a bigger buffer
E.g. teleconferencing requires reasonable upper bound
Packet loss
Inelastic Traffic Problems
Difficult to meet requirements on network with variable queuing delays and congestion
Need preferential treatment
Applications need to state requirements
Ahead of time (preferably) or on the fly
Using fields in IP header
Resource reservation protocol
Must still support elastic traffic
Deny service requests that leave too few resources to handle elastic traffic demands
E.g.

ISA Approach
Provision of QoS over IP
Sharing available capacity when congested

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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 isnot 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
Manage congestion
Meet QoS

ISA Implementation in Router

Background Functions

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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

ISA Services
Traffic specification (TSpec) defined as service for flow
On two levels
General categories of service
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Guaranteed
Controlled

load
effort (default)
Particular flow within category
TSpec is part of contract
Token Bucket
Many traffic sources can be defined by token bucket scheme
Provides concise description of load imposed by flow
Easy to determine resource requirements
Provides input parameters to policing function
Token Bucket Diagram
Best

ISA Services
Guaranteed Service
Assured capacity level or data rate
Specific upper bound on queuing delay through network
Must be added to propagation delay or latency to get total delay
Set high to accommodate rare long queue delays
No queuing losses
I.e. no buffer overflow
E.g. Real time play back of incoming signal can use delay buffer for incoming signal
but will not tolerate packet loss
ISA Services
Controlled Load
Tightly approximates to best efforts under unloaded conditions
No upper bound on queuing delay
High percentage of packets do not experience delay over minimum transit delay
Propagation plus router processing with no queuing delay
Very high percentage delivered
Almost no queuing loss
Adaptive real time applications
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Receiver measures jitter and sets playback point
Video can drop a frame or delay output slightly
Voice can adjust silence periods

Queuing Discipline
Traditionally first in first out (FIFO) or first come first served (FCFS) at each router
port
No special treatment to high priority packets (flows)
Small packets held up by large packets ahead of them in queue
Larger average delay for smaller packets
Flows of larger packets get better service
Greedy TCP connection can crowd out altruistic connections
If one connection does not back off, others may back off more
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
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
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

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Throughput and delay converge as time increases
Comparison of FIFO, FQ and BRFQ

Generalized Processor Sharing (GPS)

BRFQ can not 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
Emulates bit by bit GPS
Same strategy as BRFQ

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FIFO v WFQ

Proactive Packet Discard

Congestion management by proactive packet discard
Before buffer full
Used on single FIFO queue or multiple queues for elastic traffic
E.g. Random Early Detection (RED)
Random Early Detection (RED)
Motivation
Surges fill buffers and cause discards
On TCP this is a signal to enter slow start phase, reducing load
Lost packets need to be resent

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Adds to load and delay
Global synchronization
Traffic burst fills queues so packets lost
Many TCP connections enter slow start
Traffic drops so network under utilized
Connections leave slow start at same time causing burst
Bigger buffers do not help
Try to anticipate onset of congestion and tell one connection to slow down

RED Design Goals

Congestion avoidance
Global synchronization avoidance
Current systems inform connections to back off implicitly by dropping packets
Avoidance of bias to bursty traffic
Discard arriving packets will do this
Bound on average queue length
Hence control on average delay
RED Algorithm Overview
Calculate average queue size avg
if avg < THmin
queue packet
else if THmin avg Thmax
calculate probability Pa
with probability Pa
discard packet
else with probability 1-Pa
queue packet
else if avg THmax
discard packet
RED Buffer

RED Algorithm Detail

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Differentiated Services (DS)

ISA and RSVP complex to deploy
May not scale well for large volumes of traffic

Amount of control signals

Maintenance of state information at routers
DS architecture designed to provide simple, easy to implement, low overhead tool
Support range of network services

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Differentiated on basis of performance

Characteristics of DS
Use IPv4 header Type of Service or IPv6 Traffic Class field
No change to IP
Service level agreement (SLA) established between provider (internet domain) and
customer prior to use of DS
DS mechanisms not needed in applications
Build in aggregation
All traffic with same DS field treated same
E.g. multiple voice connections
DS implemented in individual routers by queuing and forwarding based on DS field
State information on flows not saved by routers

Services
Provided

within DS domain
Contiguous portion of Internet over which consistent set of DS policies administered
Typically under control of one administrative entity
Defined in SLA
Customer may be user organization or other DS domain
Packet class marked in DS field
Service provider configures forwarding policies routers
Ongoing measure of performance provided for each class
DS domain expected to provide agreed service internally
If destination in another domain, DS domain attempts to forward packets through other
domains
Appropriate service level requested from each domain
SLA Parameters
Detailed service performance parameters
Throughput, drop probability, latency
Constraints on ingress and egress points
Indicate scope of service
Traffic profiles to be adhered to
Token bucket
Disposition of traffic in excess of profile
Example Services
Qualitative
A: Low latency
B: Low loss
Quantitative
C: 90% in-profile traffic delivered with no more than 50ms latency
D: 95% in-profile traffic delivered
Mixed
E: Twice bandwidth of F
F: Traffic with drop precedence X has higher delivery probability than that with drop
precedence Y
DS Field Detail
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Leftmost 6 bits are DS codepoint

64 different classes available

3 pools
xxxxx0 : reserved for standards
000000 : default packet class
xxx000 : reserved for backwards compatibility with IPv4 TOS
xxxx11 : reserved for experimental or local use
xxxx01 : reserved for experimental or local use but may be allocated for future standards
if needed
Rightmost 2 bits unused
Configuration Diagram

Configuration Interior Routers

Domain consists of set of contiguous routers
Interpretation of DS codepoints within domain is consistent
Interior nodes (routers) have simple mechanisms to handle packets based on codepoints
Queuing gives preferential treatment depending on codepoint
Per Hop behaviour (PHB)
Must be available to all routers
Typically the only part implemented in interior routers
Packet dropping rule dictated which to drop when buffer saturated
Configuration Boundary Routers
Include PHB rules
Also traffic conditioning to provide desired service
Classifier
Separate packets into classes
Meter

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Measure traffic for conformance to profile
Marker
Policing by remarking codepoints if required
Shaper
Dropper
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
Per Hop Behaviour
Explicit Allocation
Superior to best efforts
Does not require reservation of resources
Does not require detailed discrimination among flows
Users offered choice of number of classes
Monitored at boundary node
In or out depending on matching profile or not
Inside network all traffic treated as single pool of packets, distinguished only as in or
out
Drop out packets before in packets if necessary
Different levels of service because different number of in packets for each user
PHB - Assured Forwarding
Four classes defined
Select one or more to meet requirements
Within class, packets marked by customer or provider with one of three drop
precedence values
Used to determine importance when dropping packets as result of congestion

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Codepoints for AF PHB

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Unit V
Protocols for QoS Support
Increased Demands

Need to incorporate bursty and stream traffic in TCP/IP architecture

Increase
capacity
Faster links, switches, routers
Intelligent routing policies
End-to-end flow control

Multicasting

Quality of Service (QoS) capability

Transport protocol for streaming

Resource Reservation - Unicast
Prevention as well as reaction to congestion required
Can do this by resource reservation
Unicast
End users agree on QoS for task and request from network
May reserve resources
Routers pre-allocate resources
If QoS not available, may wait or try at reduced QoS
Resource Reservation Multicast
Generate vast traffic
High volume application like video
Lots of destinations
Can reduce load
Some members of group may not want current transmission
Channels of video
Some members may only be able to handle part of transmission
Basic and enhanced video components of video stream
Routers can decide if they can meet demand
Resource Reservation Problems on an Internet
Must interact with dynamic routing
Reservations must follow changes in route
Soft state a set of state information at a router that expires unless refreshed
End users periodically renew resource requests
Resource ReSerVation Protocol (RSVP) Design Goals
Enable receivers to make reservations
Different reservations among members of same multicast group allowed
Deal gracefully with changes in group membership
Dynamic reservations, separate for each member of group
Aggregate for group should reflect resources needed
Take into account common path to different members of group
Receivers can select one of multiple sources (channel selection)
Deal gracefully with changes in routes
Re-establish reservations
Control protocol overheadIndependent of routing protocol
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RSVP Characteristics
Unicast and Multicast
Simplex
Unidirectional data flow
Separate reservations in two directions
Receiver initiated
Receiver knows which subset of source transmissions it wants
Maintain soft state in internet
Responsibility of end users
Providing different reservation styles
Users specify how reservations for groups are aggregated
Transparent operation through non-RSVP routers
Support IPv4 (ToS field) and IPv6 (Flow label field)
Data Flows - Session
Data flow identified by destination
Resources allocated by router for duration of session
Defined by
Destination IP address
Unicast or multicast
IP protocol identifier
TCP, UDP etc.
Destination port
May not be used in multicast
Flow Descriptor
Reservation Request
Flow spec
Desired QoS
Used to set parameters in nodes packet scheduler
Service class, Rspec (reserve), Tspec (traffic)
Filter spec
Set of packets for this reservation
Source address, source prot
Treatment of Packets of One Session at One Router

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RSVP Operation Diagram

RSVP Operation
G1, G2, G3 members of multicast group
S1, S2 sources transmitting to that group
Heavy black line is routing tree for S1, heavy grey line for S2
Arrowed lines are packet transmission from S1 (black) and S2 (grey)
All four routers need to know reservation s for each multicast address
Resource requests must propagate back through routing tree
Filtering
G3 has reservation filter spec including S1 and S2
G1, G2 from S1 only
R3 delivers from S2 to G3 but does not forward to R4
G1, G2 send RSVP request with filter excluding S2
G1, G2 only members of group reached through R4
R4 doesnt need to forward packets from this session
R4 merges filter spec requests and sends to R3
R3 no longer forwards this sessions packets to R4
Handling of filtered packets not specified
Here they are dropped but could be best efforts delivery
R3 needs to forward to G3
Stores filter spec but doesnt propagate it

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Reservation Styles
Determines manner in which resource requirements from members of group are
aggregated
Reservation attribute
Reservation shared among senders (shared)
Characterizing entire flow received on multicast address
Allocated to each sender (distinct)
Simultaneously capable of receiving data flow from each sender
Sender selection
List of sources (explicit)
All sources, no filter spec (wild card)

Reservation Attributes and Styles

Reservation Attribute
Distinct
Sender selection
Sender selection
Shared
Sender selection
Sender selection

explicit = Fixed filter (FF)

wild card = none
explicit= Shared-explicit (SE)
wild card = Wild card filter (WF)

Wild Card Filter Style

Single resource reservation shared by all senders to this address
If used by all receivers: shared pipe whose capacity is largest of resource requests
from receivers downstream from any point on tree
Independent of number of senders using it
Propagated upstream to all senders
WF(*{Q})
* = wild card sender
Q = flowspec
Audio teleconferencing with multiple sites

Fixed Filter Style

Distinct reservation for each sender

Explicit list of senders
FF(S1{Q!}, S2{Q2},)
Video distribution

Shared Explicit Style

Single reservation shared among specific list of senders
SE(S1, S2, S3, {Q})
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Multicast applications with multiple data sources but unlikely to transmit
simultaneously

RSVP Protocol Mechanisms

Two message
Resv

types
Originate at multicast group receivers
Propagate upstream
Merged and packet when appropriate
Create soft states
Reach sender
Allow host to set up traffic control for first hop

Path
Provide upstream routing information
Issued by sending hosts
Transmitted through distribution tree to all destinations

RSVP Host Model

Summary

RSVP is a transport layer protocol that enables a network to provide differentiated levels
of service to specific flows of data. Ostensibly, different application types have different
performance requirements. RSVP acknowledges these differences and provides the
mechanisms necessary to detect the levels of performance required by different applications and to modify network behaviors to accommodate those required levels. Over
time, as time and latency-sensitive applications mature and proliferate, RSVP's
capabilities will become increasingly important.
Review Questions

QIs it necessary to migrate away from your existing routing protocol to support
RSVP?

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ARSVP is not a routing protocol. Instead, it was designed to work in conjunction with
existing routing protocols. Thus, it is not necessary to migrate to a new routing protocol
to support RSVP.
QIdentify the three RSVP levels of service, and explain the difference among them.
ARSVP's three levels of service include best-effort, rate-sensitive, and delay-sensitive
service. Best-effort service is used for applications that require reliable delivery rather
than a timely delivery. Rate-sensitive service is used for any traffic that is sensitive to
variation in the amount of bandwidth available. Such applications include H.323
videoconferencing, which was designed to run at a nearly constant rate. RSVP's third
level of service is delay-sensitive service. Delay-sensitive traffic requires timely but not
reliable delivery of data.
QWhat are the two RSVP reservation classes, and how do they differ?
AA reservation style is a set of control options that defines how a reservation operates.
RSVP supports two primary types of reservation styles: distinct reservations and shared
reservations. A distinct reservation establishes a flow for each sending device in a
session. Shared reservations aggregate communications flows for a set of senders. Each
of these two reservation styles is defined by a series of filters.
QWhat are RSVP filters?
AA filter in RSVP is a specific set of control options that specifies operational
parameters for a reservation. RSVP's styles include wildcard-filter (WF), fixed-filter
(FF), and shared-explicit (SE) filters.
QHow can RSVP be used through network regions that do not support RSVP?
ARSVP supports tunneling through network regions that do not support RSVP. This
capability was developed to enable a phased-in implementation of RSVP.

Multiprotocol Label Switching (MPLS)

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
IS, DS, RSVP
Nothing directly improves throughput or delay
MPLS tries to match ATM QoS support
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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
VPN Support
Traffic from a given enterprise or group passes transparently through an internet
Segregated from other traffic on internet
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Performance guarantees
Security

Multiprotocol Support
MPLS can be used on different network technologies
IP
Requires router upgrades
Coexist with ordinary routers
ATM
Enables and ordinary switches co-exist
Frame relay
Enables and ordinary switches co-exist
Mixed network
MPLS Terminology
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
MPLS Operation Diagram

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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
Local significance only
Manually or using Label distribution protocol (LDP) or enhanced
version of RSVP
Explanation Packet Handling
Packet enters domain through edge LSR
Processed to determine QoS
LSR assigns packet to FEC and hence LSP
May need co-operation to set up new LSP
Append label
Forward packet
Within domain LSR receives packet
Remove incoming label, attach outgoing label and forward
Egress edge strips label, reads IP header and forwards
Notes
MPLS domain is contiguous set of MPLS enabled routers

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Traffic may enter or exit via direct connection to MPLS router or from non-MPLS
router
FEC determined by parameters, e.g.
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
MPLS Packet Forwarding

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
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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

Topology of LSPs
Unique ingress and egress LSR
Single path through domain
Unique egress, multiple ingress LSRs
Multiple paths, possibly sharing final few hops
Multiple egress LSRs for unicast traffic
Multicast

Route Selection

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Selection of LSP for particular FEC
Hop-by-hop
LSR independently chooses next hop
Ordinary routing protocols e.g. OSPF
Doesnt support traffic engineering or policy routing
Explicit
LSR (usually ingress or egress) specifies some or all LSRs in LSP for
given FEC
Selected by configuration,or dynamically
Constraint Based Routing Algorithm
Take in to account traffic requirements of flows and resources available along
hops
Current utilization, existing capacity, committed services
Additional metrics over and above traditional routing protocols (OSPF)
Max link data rate
Current capacity reservation
Packet loss ratio
Link propagation delay

Label Distribution
Setting up LSP
Assign label to LSP
Inform all potential upstream nodes of label assigned by LSR to FEC
Allows proper packet labelling
Learn next hop for LSP and label that downstream node has assigned to
FEC
Allow LSR to map incoming to outgoing label
Real Time Transport Protocol
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

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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

RTP Architecture Diagram

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

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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
Produce one or more outgoing RTP packets for each incoming packet
E.g. convert video to lower quality
RTP Header

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
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
RTCP Packet Formats

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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
Senders packet count (for this session)
Senders 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)

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Least significant
Most significant

16 bits is highest RTP data sequence number received from SSRC_n

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

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