computer network Module 3

COMPUTER NETWORKS (BCSE 3306)
Lecture Notes Module III Ajit K Nayak [email protected] Department of Computer Science Engineering & Application

Out Line of Module III
Network Layer, Network Layer Protocols Transport Layer, Congestion control Quality of service

Text: “Data Communications and Networking” Third Edition, Behrouz A Forcuzan, Tata Mc Graw-Hill. Chapter 19 - Chapter 23
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Lecture I
Network Layer
• Host-to-Host Delivery
• Addressing • Routing

•Network Layer Protocols
• IPV4 • ARP • ICMP

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Network Layer

Protocol used is IP for Network Layer Responsibility of this layer to deliver the datagram to the correct destination host. i.e. host-tohost delivery Computer Networking / Module III/ AKN / 4

Classful IP Addresses
Each host on a TCP/IP internet is assigned a unique 32-bit unicast Internet address that is used in all communication with that host. Each unicast IP address is a pair(netid, hostid), where netid identifies a network and hostid identifies a host on that network The total address space is 232=4,294,967,296. But all addresses are not usable It is represented in dotted decimal notation 128.11.3.31 1000000 00001011 00000011 00011111
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Type of communication
Unicast: one-to-one communication. i.e. One source sends to exactly one destination host Multicast: one-to-a group. i.e. one sources sends to a predefined group of destination hosts simultaneously Broadcast: one-to-all. i.e. one source sends to all other hosts available in that network. Broadcast in Internet is not allowed. Others: anycast, geocast, etc. read yourself!
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Classes of IP addresses
Class A
0 netid

0.0.0.0 – 127.255.255.255
hostid

Class B
1 0 netid

128.0.0.0 – 191.255.255.255
hostid

Class C
1 1 0

192.0.0.0 – 223.255.255.255
netid hostid

Class D
1 1 1 0

224.0.0.0 – 239.255.255.255
multicast address

Class E
1 1 1 1

240.0.0.0 – 255.255.255.255
reserved for future use
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IP Addresses
Class A
First octet defines the netid and first bit is fixed Max. no of network possible: 27-2=126 All zero and all one values can not be used 24 bits are used for hostid Max no of hosts 224-2=16,777,214 per network can be connected to a class A network

Class B
First two octet define the netid and two left bits are fixed : 214-2=16,382 networks and 216-2=65,534 hosts/network
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IP Addresses

Class C: First three octet defines netid and three bits fixed
221-2=2,097,151 networks 28-2=254 hosts/network

Class D: No net and host ids
First four bits are fixed, remaining 24 bits define multicast addresses?

Class E: No use

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Special Addresses
Network Addresses
Addresses having all zero hostids are used to identify a network and is not assigned to any host
Specific 123.50.16.90 123.65.7.34 All 0s 123.90.123.4 ...

123.0.0.0 Class A
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Network Address

Find Network addresses of the following IP addresses
24.32.3.29 190.234.211.21 200.23.31.6
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Special Addresses contd.
Direct Broadcast Addresses
Used by a router to broadcast a message to all hosts of a network Specific All 1s It can only be used as a destination address by specifying hostid as all 1s
221.45.71.20 221.45.71.64 ... 221.45.71.255 221.45.71.0 R 221.45.71.99

Class C network Computer Networking / Module III/ AKN / 12

Special Addresses contd.
Limited Broadcast Addresses
Used by a host to send a message to every other host in that network All 1s All 1s It can only be used as a destination address by specifying netid and hostid as all 1s Router blocks the packet and discards it. 221.45.71.20 255 .25 5 221.45.71.64
.25 5 .25 5

221.45.71.99 ... Blocked here

221.45.71.0 Class C network

R

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Special Addresses contd.
All 0s All 0s Used by a DHCP client at bootstrap as a source address to get a valid IP address from the DHCP server It is specified by all 0s. The destination is a limited broadcast address It is always a Class A address regardless of the network ?.?.?.? 221.45.71.64 221.45.71.99 255 .25 5.2 ... 55. 0.0 255 .0.0 Bootstrap server B 221.45.71.0 Class C network 221.45.71.1
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This Host Addresses

Special Addresses contd.
Loop Back Addresses
Used by a host to communicate with itself without a special network interface This is the address with first byte as 127 and the packet never goes out of the machine
127 Any P1 Host P2

127.0.0.1

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Private Network Addresses
These IPs should not be used in internet but one can use for hosts that do not require direct access to the Internet These addresses are filtered by Internet routers and therefore do not have to be globally unique 10.0.0.0 – 10.255.255.255 172.16.0.0 – 172.31.255.255 192.168.0.0 – 192.168.255.255 Automatic Private IP Addressing
Used by windows machine, if there is no DHCP available 169.254.0.0 – 169.254.255.255

Rfcs: 1466, 1918, 1597, 3927 etc.
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Masking
To reach at a host we have two level of hierarchy
1. Reach at destination network 2. Reach at host

Masking is a process that extracts the address of physical network from an IP address Mask is an IP having netid all ones and hostid all zeros 141.14.2.21 255.255.0.0 141.14.0.0
Mask

A bit wise and operation is performed 10001101 00001110 00000010 00010101 11111111 11111111 00000000 00000000 141 14 0 0
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Problems with classful
1.

2.

3.

There are three main problems with “classful” addressing, Lack of Internal Address Flexibility: Big organizations are assigned large, “monolithic” blocks of addresses that don't match well the structure of their underlying internal networks. Inefficient Use of Address Space: The existence of only three block sizes (classes A, B and C) leads to waste of limited IP address space. Proliferation of Router Table Entries: As the Internet grows, more and more entries are required for routers to handle the routing of IP datagrams, which causes performance problems for routers. Attempting to reduce inefficient address space allocation leads to even more router table entries.
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Subnetting
This technique helps to divide one physical network into some smaller subnets (i.e.to create hierarchies) Advantage:
Increasing popularity of LAN may exhaust the netids When many hosts connected to a single network the messages are overcrowded due to the broadcast nature of LANs

The scheme allows multiple physical networks to share a same prefix (1980s) A second extension is also available to divide suffix and prefix at an arbitrary point called classless addressing and supernetting (1990s)
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Subnetting an Example
.2.20 .7.96 ... R .2.20 Without subnet 141.14.0.0 141.14.2.0 .7.96 141.14.0.0 .22.90

.2 R

.7 .22

141.14.0.0 141.14.7.0 With subnet .22.90
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141.14.0.0 141.14.22.0

Subnetting
Rest of the Internet still fills as if one network. i.e packet destinated at 141.14.2.21 still reach at router R and it is aware of three subnets. Last two octets define two things
1.

subnetid

2.

hostid

Delivery of packets now involve three steps
1. Delivery to the network 2. Delivery to the subnet 3. Delivery to the host
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Example 1
Q. Design 8 subnets from 211.77.20.0 Ans. Taking 3 bits for subnet in last byte, remaining 5 bits are used for hostid

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Example 1 contd.

According to classic IP routing rules, it was not possible to use the subnets with all zero or all one values. i.e. subnet #0 and subnet #7 However, most modern machines have no troubles using uppermost or lowermost subnets
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Example 2
The network address is x.y.z.0, subnet mask is 255.255.255.248 then design the subnets From mask it is clear that first five bits of last byte is used as subnetid and last three bits are used as hostids i.e. 25=32 subnets and 23-2=6 hosts/subnets
Subnet #0: x.y.z.0, x.y.z.1, . . ., x.y.z.6, x.y.z.7 Subnet #1: x.y.z.9, x.y.z.10, . . ., x.y.z.14, x.y.z.15 Subnet #2: x.y.z.16, x.y.z.17, . . ., x.y.z.22, x.y.z.23 ...................................... Subnet #29: x.y.z.232, x.y.z.233, . . ., x.y.z.238, x.y.z.239 Subnet #30: x.y.z.240, x.y.z.241, . . ., x.y.z.246, x.y.z.247 Subnet #31: x.y.z.248, x.y.z.249, . . ., x.y.z.254, x.y.z.255 First column is used as subnet id, last column is used as Computer Networking / Module III/ AKN / 24 broadcast address.

IP addresses are used not only to uniquely identify IP addresses but also to facilitate the routing of IP datagrams over networks

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Problems with IP Addressing
If a host computer moves from one network to another, its IP address must change (manually) Because routing uses the network portion of the IP address, the path taken by packets traveling to a host with multiple IP address depends on the address used.
Network 1 I2 A I4 If link I3 fails than A cannot send to B Network 2 I2 R I3 B I5

Addressing Authorities
IANA: Internet Assigned Number Authority upto 1998 ICANN: Internet Corporation for Assigned Names and Numbers
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Dynamic Address configuration
Each computer that is connected to Internet must have following information
Its IP address Subnet mask Router/gateway’s IP address Name server’s IP address

These information are maintained in operating system and stored in disk These information may be acquired by assigning static values or can also be obtained dynamically when needed DHCP is designed to assign these information dynamically (on demand) It is a client/server program, when client sends a request to server, server selects an IP address from the pool of unused IP address for a negotiable period of time (lease time)
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Dynamic Host Configuration Protocol
TRANSITION STATES Initializing state
Client broadcasts a DHCPDISCOVER message

Selecting state

All the DHCP servers replies with a DHCPOFFER message, which contains IP address, lease time etc. client chooses on of the offers. Client now sends a DHCPREQUEST message

Requesting state

Remains in this state till it gets the DHCPACK, which creates a binding of physical and logical address
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DHCP contd.
Bound state
After using 50% of the time, client requests for renewal by sending another DHCPREQUEST, or client can cancel the lease and go back to the initializing state

Renewing state
If it receives the DHCPACK then the timer is reset or client goes again for rebinding. If not received till 87.5% of lease time then goes to rebinding state

Rebinding state
It remains in this state till it receives a DHCPNAK or lease expires, client goes to initializing state for a fresh process or goes to bound state if DHCPACK is received
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Network Address Translation
Home users and small business can be connected to Internet via an ADSL or cable modem and every body needs one or more IP addresses Due to shortage of IP addresses, the demand may be full filled by using the private network address through Network address translation method (NAT) NAT enables a user to have large set of addresses (private) internally and one or a small set of addresses externally (global)

Address translation
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NAT contd.
Address translation
All out going packets go through the NAT router, which replaces destination address in the packet with global NAT address. Similarly all incoming packets also pass through the NAT router, which replaces the destination address with appropriate private address using Translation table Private Address 172.18.3.1 172.18.3.2 ... Private Port 1400 1401 ... External Address 25.8.3.2 25.8.3.2 ... External Port 80 80 ... Transport Protocol TCP TCP ...

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Routing techniques
Usually routing uses an Internet routing table on each machine that stores information about possible destinations and how to reach them Next Hop Routing
20.0.0.5 network 10.0.0.0 10.0.0.5 Q network 20.0.0.0 20.0.0.6 R 30.0.0.6 network 30.0.0.0

Dest 10.0.0.0 20.0.0.0 30.0.0.0 40.0.0.0

Next hop 20.0.0.5 Direct Direct 30.0.0.7

30.0.0.7 40.0.0.7

S

network 40.0.0.0

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Network-Specific Routing
Instead of one entry for each destination host, we maintain one entry for total network

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Host-Specific Routing
Host-specific routes
Although all routing is based on networks and not on specific hosts, most software allows per-host routes as a special case. This is helpful for administration purposes like testing, controlling access and debugging etc.

A
Q

Net1

R

Destination B Net2 Net3
Net3

Next hop R Q R

Net2

P

Table for host A

B
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Default Routing
Default Routes
In this type of routing , it looks in the routing table for the destination network. If no route appears in the table, the routing routines send the datagram to a default router It is useful when the network has a small set of local addresses and only one connection to the rest of internet network 10.0.0.0 S Rest of Internet Q network 20.0.0.0

• Routing table for a host on network 10.0.0.0

Destination Next hop 20.0.0.0 Q Default S
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Static versus Dynamic Routing Tables
Routing tables may be constructed statically or dynamically. The success of routing depends on the consistency of routing table information Static Routing table
Information entered manually, can be used for small intranet that does not change very often. It is not a good choice in Internet where information changes very often

Dynamic Routing table
Updated periodically using the dynamic routing protocols like RIP, OSPF, or BGP etc. Dynamic routing is preferred over static routing as the updation of routing table is done dynamically thus providing a consistent routing mechanism.
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Hierarchical Routing
It is not possible to keep information about each host and or each network in the routing table of each Internet router To solve this problem we maintain hierarchical routing. According to this technique the we maintain partial information in routers e.g. if the block assigned to one ISP is a.b.c.d/n and it may create many subnets of e.f.g.h/m for each of its customers, the rest of the Internet does not have to be aware of this division. i.e. all customer of that ISP are defined as a.b.c.d/n to the rest of Internet There is only one entry needed for this ISP The router inside ISP recognizes the sub-blocks and routes the packets to the destination To reduce the size of table further the hierarchical routing may be included. i.e. The routers of ISPs outside Europe will have only one entry for packets to Europe in their routing tables.
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Internet Protocol (IPV4:RFC-791)
Connection less delivery system
Internet service consists of an unreliable, best-effort, connection less packet delivery system. Unreliable because delivery is not guaranteed. i.e.The packet may be lost, duplicated, delayed or delivered out of order but the service will not detect such conditions, nor will it inform the sender or receiver. A sequence of sent from one computer to another may travel over different paths, or some may be lost while others are delivered. It is best-effort delivery because the internet software makes an earnest attempt to delivery packets i.e. the internet does not discard packets always. Unreliability arises only when resources are exhausted or underlying networks fail.
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Internet Protocol (contd.)
The Internet protocol defines unreliable, connection less delivery mechanism ( IP ) It defines the basic unit of data transfer used throughout the internet by specifying the exact format of data It performs routing function, choosing the path over which the data will be sent It also includes a set of rules that embody the idea of unreliable packet delivery. i.e. It tells how to process the packets, how and when error message should be generated, and the conditions under which the packets can be discarded.

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Internet Protocol Datagram Format
0 4 8 16 19 24 31

Ver HLen Identification TTL

Service Type

Total length Flag Fragment offset Protocol Header checksum Source IP Destination IP Padding

IP Options if any Data ...

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IP Header

Ver: version of IP (4 or 6) HLen: total length of datagram header (20-60 bytes) Type of Service: how the datagram should be handled 4 7 by the router 0
Precedence D T R C Precedence: (3 bits) defines priorities in cases like congestion TOS bits: low delay, high throughput, high reliability, less cost. A hint to router as a decision making factor for routing algorithms. Internet does not guarantee to provide any 6 particular type of service 0 7 unused IETF redefined the meaning CODEPOINT If last three bits are zero than first three bits define precedence (backward compatibility)i.e. xxx000
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IP Header (contd.)
The 64 code point values maps to an underlying service definition and is divided into three groups Pool Codepoint Assigned by
1 2 3 xxxxx0 xxxx11 xxxx01 Standards Organization(IETF) Local or Experimental Local or experimental for now

If the standards bodies exhaust all values in pool 1, they may also choose to assign values in pool 3 Total Length: defines total length of the datagram in bytes. i.e. 216-1=65,535 bytes max. including header
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IP Header (contd.)
Fragmentation
Each datagram is encapsulated in a datalink frame before transmission. It has to travel through different networks and the frame size differs for different networks and is defined by MTU of that network Identification: IP software keeps a global counter and increments each time a new datagram created. if the datagram is fragmented then the identification is copied to each fragment of same datagram U D M Flags:
3 bit field, D:do not fragment M: more fragment
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IP Header (contd.)
D=1: datagram must not be fragmented D=0: datagram can be fragmented M=1: It is not the last fragment M=0: It is the last or only fragment Fragmentation offset: It shows the relative position of the fragment, w.r.t. whole datagram Offset measured in bytes 0 0 3999 1400 2800 1399 2799 2800/8 = 350 3999
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0/8 = 0 1400/8 = 175

IP Header (contd.)
Time to Live:It specifies how long in seconds, the datagram is allowed to remain in the internet system When a datagram arrives at a router, it records the time and before sending forward it decrements the time to live field. When it becomes zero, the datagram is discarded and an error message is sent to the source But to estimate exact time is difficult because routers do not usually know the transit time for physical networks. Thus in practice the time to live acts as a hop limit rather than an estimate of delay. Each router only decrements the value by one till it becomes zero.

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IP Header (contd.)
Protocol: It defines the higher level protocol that uses the IP layer service
ICMP- 1, IGMP-2, TCP-6, UDP-17 etc.

Header Checksum: Ensures the integrity of header values
Divide the packet in to k section of 16 bits each All sections are added using ones complement method The final result is complemented to make checksum Follow the same method at receiver. If the result is zero accept else discard the datagram
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IP Header Options
IP header is made of two parts: the fixed part and the variable part. Fixed part is 20 byte long; the variable part comprises the option which can be a max. of 40 bytes. These are included primarily for network testing and debugging Data (variable length) Format Code(8) Length(8) Code:
Copy Class Number

It contains copy(1), class(2), and number(5) Copy = 1: options should be copied to all fragment Copy = 0: options must be only copied to first fragment
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Options field of IP Datagram
Class
00 : used for datagram control, 01: reserved 10: Debugging and management, 11: reserved

Number
Defines the type of options

Length
It defines the total length of the option including the code field and the length field itself

Data
Contains the data that specific options require
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Types of Options
Options Data
0

0 : End of option, used if options do not end at end of header 1: no operation, used to align octets 7-byte opt 8-byte opt
1

7: Record Route, It is used to record the routers that handles the datagrams. It can list up to nine router addresses? The source creates empty fields for the IP addresses in the data field of the option

Code

Length Pointer First IP Address (empty) Second IP Address (empty) Third IP Address (empty)
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Types of Options
Whenever a router handles the datagram, it compares the pointer and length field. If the pointer field is greater than length field, the list is full. Else router inserts its IP address at the position specified by pointer and increments the pointer by four. This option requires that two machines must cooperate. i.e. source must enable record route and destination must agree to process the resultant list. 9: Strict source route, used by the source to predetermine a route for the datagram as it travels through internet
i.e. a source may choose a safer route to the destination
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Types of Options
If a datagram specifies a strict source route, all of the routers defined in the option must be visited in order by the datagram. If a datagram reaches at a router not in the list then it is discarded and error message is sent to the source. If a datagram reaches at the destination and some entries were not visited, it will also be discarded and error message is issued. i.e. The path between two successive addresses in the list must consists of a single physical network It is only useful when the network topology is known
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Types of Options
3: Loose source route, It is similar to strict source but allows multiple network hops between successive address in the list Both source route options requires routers along the path to overwrite the list with their local network address. 4: Timestamp, is used to record the time of datagram processing by the router. Code Length Pointer OFlow First IP Address First Timestamp ...
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Flags

Types of Options
Length and pointer fields are used to specify the length of the space reserved for the option and the location of the next unused slot. Oflow(4) contains an integer count of routers that couldnot supply timestamp because the option was too small Flag(4), controls the exact format of the option and tells how routers should supply timestamps.
0: Record timestamps only, omit IP addresses 1: Precede each timestamp by an IP address 3: IP addresses are specified by sender; a router only records a timestamp if the next IP address in the list matches the router’s IP address
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Routing IP Datagrams
Routing is the process of choosing a path over which to send packets, and router refers to a computer making the choice The goal of IP is to provide a virtual network that encompasses multiple physical network and offers a connection less datagram delivery service Routing is divided into two forms
1. Direct delivery: Transmission of a datagram from one computer across a single physical network directly to another 2. Indirect delivery: Transmission of datagram to a destination not attached directly to the senders network, thus forcing the sender to pass the datagram to a router for delivery
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Datagram delivery over a single Network
In this case the final destination of the datagram is a host connected to the same physical network
• Extraction of network address takes a few machine instructions making the process extremely efficient

R

• The sender extracts the network address of destination IP and compares it to the network portion of its own IP . • If a match is found then the delivery is direct and it does not involve routers • Now the destination IP address is used to find its physical address for actual datalink layer delivery?
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Indirect Delivery
It is more difficult because the sender must identify a router to which the datagram can be sent • The datagram goes from router to router until it reaches the destination network R • At the destination network it R performs direct delivery to reach at the host • How can a host know which router to use for a given destination? • How can a router know where to send datagrams?

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Mapping Internet Address to Physical Address
Delivery of a packet requires two levels of addressing. Hosts and routers are recognized at the network level by their logical addresses, which is universal and implemented in software But at physical level devices are recognized by their physical addresses Therefore, the packet to be sent from A to B should be mapped to the physical address of B Address mapping must be performed at each step along a path from original source to ultimate destination i.e 1. Last hop addressing 2. Intermediate addressing
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Mapping Internet Address Physical Address
Last hop addressing
Packet’s internet address is mapped to the final destinations physical address

Intermediate addressing
At any point along the path packet is mapped to intermediate routers physical address (as destination)

Address resolution problem
The problem of mapping logical to physical address is called the ‘address resolution problem’. There are two technologies followed by TCP/IP to resolve the problem. Resolution through direct mapping Resolution through Dynamic binding
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1. 2.

Mapping Internet Address Physical Address
Resolution through Direct Mapping
In proNET token ring network, the administrator chooses small integers for physical addresses while installing an interface. Now to have a efficient address resolution one can find a function PA = f (IA) to calculate the numbers. i.e. if f is simple then the mapping will be simple Another way is to keep a table containing address pairs (logical, physical) and a hash function may be used to search that table Another advantage in this method is, if one interface of a computer is changed then also the same physical address can be used for the new interface Also new computers can be added to the network without changing the existing assignments.

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Mapping Internet Address Physical Address
Resolution through dynamic binding
In Ethernet technology the 48 bit physical address is assigned when manufactured Thus the physical address of a computer changes each time an interface is changed. Because the physical address is 48 bit long and not assigned by the user thus it is impossible to devise a function for mapping as in previous case To avoid maintaining a mapping table (not possible !) the designers developed a protocol to bind addresses dynamically known as ‘Address Resolution Protocol’ ARP provides a mechanism that is both reasonably efficient and easy to maintain
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Resolution through dynamic Binding
Idea
Sender broadcasts a special packet that asks the destination about its physical address Destination recognizes the packet and sends a reply containing its physical address Now the sender uses physical address to send packets directly to destination A B C D

A

B

C

D

A

B

C

D

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ARP Packet Format (RFC-826)
Hardware Type
H/W length Protocol length

Protocol Type
Operation

Sender Hardware Address Sender Protocol Address Target Hardware Address Target Protocol Address
H/W Type: 16 bit field defines type of LAN e.g. Ethernet=1 Protocol Type: 16 bit field defining IP version e.g. IPV4=0080016 Hlen: 8 bit, length of hardware address e.g. Ethernet = 6 Plen : 16 bit, length of logical address Operation : 8 bit, request=1, reply 2
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Address Resolution Protocol
Encapsulation
ARP packet is encapsulated directly in to a datalink frame
SFD Dest Add Source Add Type Data CRC

Refinements

ARP Packet

If the target machine is down or too busy to accept the request? i.e sender may not receive a reply (1) or it is delayed(2) Retransmit the request for (1) or it restores the original outgoing packet till it resolves the address
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ARP Implementation
ARP Cache
After receiving an ARP reply, it saves the IP address and corresponding hardware address in its cache for successive lookups But problem occurs if receiver crashes in between and source gets no information but keep on sending To resolve above problem a timer is used, when it expires the information in the cache is erased and normal procedure starts again Another refinement possible is, senders IP-Physical address binding can also be updated in receivers cache before processing the ARP request
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Four cases using ARP

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Limitations with IP
A datagram travels from router to router till it reaches one that can deliver directly to its final destination If a router cannot route a datagram? If the router detects an unusual condition that affects its ability to forward the datagram? In an connectionless system, each router operates autonomously, i.e without coordination of sender. and IP fails to deliver the datagram if
The destination is temporarily or permanently disconnected The TTL expires The intermediate routers become so congested that they cannot process the incomingComputer Networking / Module III/ AKN / 66 traffic

The Internet Control Message Protocol
To allow routers in an internet to report errors or provide information about unexpected circumstances, one mechanism is attached with IP is called “The Internet Control Message Protocol”, ICMP ICMP allows routers to send error or control messages to other router or hosts; It provides communication between the IP software on one machine and the IP software on another i.e. The ultimate destination of an ICMP message is not an application program or user on destination but the IP software of that machine ICMP is not restricted only to routers but is allowed to be used by any arbitrary machine to get some information. ICMP messages travel across internet in the data portion of IP datagrams
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Error Reporting / Error Correction

When a datagram causes an error, ICMP can only report the error condition back to the original source of the datagram. The source must take some action to correct the error It cannot be used to inform intermediate routers about the problem An Example
If a datagram follows a path R1, R2, . . ., Rk and Rk has the incorrect information and mistakenly routes the datagram to Re Now Re cannot use ICMP to report the error back to Rk but it can send a report back to the original source And the original source has no control over the misbehaving router. In fact it is not possible for the source to know which router (Rk) causes the problem Computer Networking / Module III/ AKN / 68

ICMP Message
Message Delivery It requires two levels of encapsulation
Header Header Header ICMP Data

Datagram Data Frame Data

– Even though ICMP messages are encapsulated and sent using IP datagrams, it is not considered a higher level protocol, but a required part of IP – It is Because, it needs to travel across several physical networks to reach their final destination
Computer Networking / Module III/ AKN / 69

ICMP Message Format
Type (8 bit) Code (8 bit) Checksum (16 bit) Rest of Header Data . . . (Variable size)

Type : identifies the message type Code : provides further information about the message type Checksum : error detection ICMP messages that report errors always include the header and first 64 bit data bits of the datagram causing the problem
Computer Networking / Module III/ AKN / 70

ICMP Message Format (contd.)
Type 0 3 4 5 8 9 10 11 12 Message Echo Reply Destination unreachable Source Quench Redirect (change route) Echo Request Router Advertisement Router solicitation Time Exceeded for a datagram

Ping: One of the most frequently used debugging tool that invokes ICMP echo request and echo reply messages

- Any machine that receives an echo request formulates an echo reply and return it to the Parameter problem on a datagram original sender The total table is available in page 133 of D.E. Comer
Computer Networking / Module III/ AKN / 71

Echo Request and Reply Message
Type(8 / 0) Code (0) Identifier Data . . . (optional) Checksum Sequence no

Optional Data is a variable length field that contains data to be returned to sender Identifier and Sequence number are used by the sender to match replies to request. The Type field specifies whether the message is a request (8) or reply (0)
Computer Networking / Module III/ AKN / 72

Reports of Unreachable Destinations
Type-3 Code (0-15) Checksum Unused - all zeros Part of the received IP datagram including IP header + first 8 byte of datagram data When a router cannot forward or deliver an IP datagram, it sends a ‘destination unreachable’ message back to the original source The code field contains an integer that further describes the problem Code Meaning Cause
0: 1: 2: 3: 4: Network unreachable host unreachable Protocol unreachable Port unreachable fragmentation required (h/w failure) (do) (receiving protocol not running) (receiving appl. Prg not running) (D bit set) etc.
Computer Networking / Module III/ AKN / 73

Congestion and Datagram flow control
Type-4 Code -0 Checksum Unused - all zeros IP header + first 8 byte of datagram data

IP doesn't have a flow control (rate of sending and receiving) mechanism, which may lead to congestion. i.e The router eventually exhausts memory and discards additional datagrams arrived ‘Source quench’ message has been designed to add a kind flow control to IP. When a datagram is discarded, it sends a source quench message to the sender, which helps in
Reporting source that datagram is discarded Make the source aware of congestion and to slow down
Computer Networking / Module III/ AKN / 74

Route change requests
Type-5 Code (0-3) Checksum Router Internet Address IP header + first 8 byte of datagram data

Routers are assumed to know correct routes; hosts begin with minimal routing information and learn new routes from routers If a host sends a datagram to an incorrect router, then the router forwards the datagram in correct destination and sends a ‘redirect message’ to the host. Now host updates its table accordingly Code
0: redirection for the network 1 : redirection for the host
Computer Networking / Module III/ AKN / 75

Detecting Circular or long routes
Type-11 Code (0-1) Checksum Unused IP header + first 8 byte of datagram data

This message is generated in two cases
Code 0: TTL exceeded If there are errors in one or more routing table a datagram may travel in a loop. After some time when TTL becomes zero the datagram is discarded and a ‘Time exceeded’ message is sent to source Code 1: Fragment reassembly time exceeded If all fragments that belong to one datagram don’t arrive at the destination within a time limit then the fragments are discarded and a Time exceeded message is sent to the source
Computer Networking / Module III/ AKN / 76

Reporting Other Problems
Type-12 Code (0-1) Checksum Pointer Unused IP header + first 8 byte of datagram data

If a router or destination discovers an ambiguous or missing value in any field of the datagram header then it sends a ‘Parameter problem’ message back to source Code 0: Error in header fields
Pointer field points to the byte with problem

Code 1: Required part of option is missing
Pointer field not used in this case
Computer Networking / Module III/ AKN / 77

Clock Synchronization and Transit Time Estimation
Type(13-14) Code -0 Checksum Identifier Sequence number Source: Originate time stamp Destination: Receive time stamp Destination: Transmit time stamp (departure)
‘Time Stamp message’ is used by two machines to determine the round trip time needed for an IP datagram to travel between them

Each time the fields hold a no representing time measured in milliseconds from midnight in GMT Calculation:
Sending time = receive TS - Originate TS Receiving time = datagram return time - Trnsmit TS Round trip time = sending time + receiving time
Computer Networking / Module III/ AKN / 78

Obtaining a subnet mask
Type(17-18) Code -0 Checksum Identifier Sequence number Address Mask

‘Address mask request/reply’ message are used by a host to obtain its mask from a router

Router Discovery
Type(9) Nun addr Code -0 Checksum Addr size Life time Router Address 1 Preference level 1 Router Address 2 Preference level 2 . . .Networking / Module III/ AKN / 79 Computer

Router Solicitation/Advertisement
Type(10) Identifier Code -0 Checksum Sequence number

ICMP supports a router discovery scheme that allows hosts to discover router address. A host can broadcast a ‘router solicitation’ message. The routers that receive the message broad cast their routing information using ‘router advertisement’ message ICMP router discovery scheme helps in two ways 1. Instead of providing a statically configured router address via a boot strap protocol, the scheme allows a host to obtain information from router itself 2. The mechanism uses a soft state technique with timers to prevent hosts from retaining a route after a router crashes
Routers advertise their information periodically, and a host discards a route if the timer for a route expires (30min, 10min)
Computer Networking / Module III/ AKN / 80

Lecture II
Network Layer Protocols
• IPV6 • ICMPR6 • Unicast Routing protocols • RIP • OSPF

Computer Networking / Module III/ AKN / 81

IPv6: Need for an alternative
IPv4 has two level address structure (?) and categorized into 5 classes. The use of address space is inefficient The internet must accommodate realtime audio and video transmission, which requires min delay and reservation of resources The Internet must accommodate encryption and authentication of data for some application Not only the computers but various devices including house hold devices, hand held devices, telephones etc. needs IP address
Computer Networking / Module III/ AKN / 82

Characteristics of IPv6
Larger Address Space: 128 bit long
Huge increase in address space

Better header format
options are separated from base header

New options
To add new functionalities

Allowance for extension
To support new technologies

Support for resource allocation
To support traffic such as real-time audio and video

Support for more security
Encryption and authentication mechanism

RFCs
1365, 1550, 1678, . . .
Computer Networking / Module III/ AKN / 83

IPv6 address

• 128 bits are divided into eight sections of hexadecimal nos, each 2 byte long sections separated by colons • The address may be abbreviated, i.e the leading zeros can be omitted (not trailing zeros)

• consecutive sections consisting of zeros can be replaced with double semicolons • if there are two runs of zero section than only one of them can be abbreviated
Computer Networking / Module III/ AKN / 84

Unicast Addresses

Defines two types of unicast addresses
Geographically based unicast address (left for future definition) Provider based unicast address (discussed below)

Type identifier: 3 bit field defines the address as a providerbased address
Computer Networking / Module III/ AKN / 85

Unicast Addresses contd.
Registry identifier: 5bit field indicates the agency that has registered the address.currently three registry has been defined.
INTERNIC: center for North America RIPNIC: center for European registration APNIC: for Asian and Pacific countries

Provider indentifier: variable-length field identifies the provider for Internet access (like ISP). A 16 bit length is recommended for this field Subscriber identifier: a 24 bit is assigned to an organization subscribing to the Internet via provider Subnet identifier: a 32 bit is assigned to define a subnet under the territory of a subscriber Node identifier: a 48 bit is assigned for the identity of the node connected to subnet
Computer Networking / Module III/ AKN / 86

Multicast addresses
First 8 bits all 1s Flag: 4bit field that defines the group address as either permanent or transient Scope: 4 bit field defines scope of the group address Group ID: 112 bits identifies group

Anycast addresses
A packet destinated for anycast address is delivered to only one member of the anycast group. i.e. member having shortest route No block is assigned to for this anycast address
Computer Networking / Module III/ AKN / 87

Reserved addresses
Start with eight zeros

Unspecified address is used when a host does not know its own address Loopback address is used by a host to test itself Compatible address is used during the transition from IPv4 to IPv6. i.e. when passing from IPv6 to IPv6 via IPv4 network Mapped address is also used during transition when sending from Ipv6 to IPv4 computer
Computer Networking / Module III/ AKN / 88

Local addresses
Used when an organization wants to use IPv6 without being connected to Internet

Nobody outside the organization can send a message to the nodes using these addresses A link local address is used in an isolated subnet A site local address is used in an isolated site with several subnets
Computer Networking / Module III/ AKN / 89

Format of an IPv6 datagram
• Each packet is composed of a mandatory base header (40 bytes) followed by a payload. • Payload consists of two parts (65535 bytes)
Optional extension header • Data from an upper layer
Computer Networking / Module III/ AKN / 90

•

Base Header
Version(4): version of IP Priority(4): priority of the packet w.r.t. congestion Flow level(3byte): special handling for a particular flow of data Payload length(2 byte): total length of datagram excluding base header Next header(8): either one of the optional extension headers used by IP or the header for an upper layer protocol like UDP, TCP Hop Limit(8): same as TTL Source Address(16byte): IP of source Source Address(16byte): IP of destination
Computer Networking / Module III/ AKN / 91

Comparison between IPv4 and IPv6 packet headers

Computer Networking / Module III/ AKN / 92

Extension header
The base header can be followed by six extension headers Hop-by-hop Option
Is used when the source needs to pass information to all routers visited by the datagram. Three options are defined Pad1: 1 byte, designed for alignment purposes PadN: used when 2 or more bytes needed for alignment Jumbo payload: is used to define a payload longer than 65535 bytes

Fragmentation
Only original source can fragment after using a path MTU discovery to get the smallest MTU supported by any network on the path If it will not use the technique then it must fragment a datagram to a size <= 576 bytes
Computer Networking / Module III/ AKN / 93

Extension header contd.
Authentication
It validates sender, and ensures integrity of data

Encrypted Security Payload
It provides confidentiality and guards against eavesdropping

Source Routing
Uses the concept of strict/loose source routing

Destination Option
Is used when the source needs to pass information to the destination only. Intermediate routers are not permitted access too this information
Computer Networking / Module III/ AKN / 94

Comparison between IPv4 options and IPv6 extension headers

Computer Networking / Module III/ AKN / 95

Transition from IPv4 to IPv6
Because of huge systems using IPV4 that’s why three strategies were proposed for smooth transition

Dual stack
A station should run both IPv4 and IPv6 simultaneously until all the Internet uses IPv6 If DNS returns IPV4 address then source sends IPV4 packet else IPV6 packet

Tunneling
When two computers using IPV6 want to communicate with each other and the the packet has to pass through a region that uses IPV4 Therefore IPV6 packet is encapsulated in an IPV4 datagram when it enters that IPv4 region Computer Networking / Module III/ AKN / 96

Transition from IPv4 to IPv6
Header Translation
It is necessary when the majority of the Internet has moved to IPv6 i.e. If sender uses IPv6 but receiver uses IPv4 Header must be completely translated It uses mapped address of IPv6

Computer Networking / Module III/ AKN / 97

ICMPv6
Comparison of error-reporting messages in ICMPv4 and ICMPv6

Comparison of query messages in ICMPv4 and ICMPv6

Computer Networking / Module III/ AKN / 98

Unicast Routing Protocols
A routing protocol allows routers share their knowledge (routing information) about the network with other routers. They maintain a table to keep routing information. This table gets updated periodically after receiving information from neighbouring routers Routers use routing table to decide about the best route based on a cost metric Cost metric
Hop count: cost of passing through any network is same. i.e. passing through one network costs 1 hop Max throughput: throughput is more in passing through an fiber than in radio link Min delay: delay is less in fiber than satellite link Reliability: some networks may be more reliable than others, it is decided based on a policy.

Various routing protocols available are RIP, OSPF etc.
Computer Networking / Module III/ AKN / 99

Routing Information Protocol
It is based on Distance Vector routing, which uses BellmanFord algorithm for calculating the routing table Distance Vector Routing
In this scheme, each router periodically (30 s) shares (broadcasts) its own routing information with its neighbours Every router keeps a routing table that has three columns in its simplest form for each entry about a network

• A, B,C, D are (routers) • To: destination network • Cost: hop count • Next: next hop
Computer Networking / Module III/ AKN / 100

RIP Updating
Receive: a response RIP message
1. Add one hop to the hop count for each advertised destination. 2. Repeat the following steps for each advertised destination: 1. If (destination not in the routing table) 1. Add the advertised information to the table. 2. Else 1. If (next-hop field is the same) 1. Replace entry in the table with the advertised one. 2. Else 1. If (advertised hop count smaller than one in the table) 1. Replace entry in the routing table. 3. Return.
Computer Networking / Module III/ AKN / 101

Example of updating a routing table

Computer Networking / Module III/ AKN / 102

Initial and Final routing tables in an example network

Computer Networking / Module III/ AKN / 103

Problems with RIP: Count-to-infinity
Count to infinite
A B C D E F

Suppose there is a 1, 2, B 3, C 4, D 5, E Initially network as shown 2, B 3, C 4, D 5, E Each router keeps the After 1 exchange 3, C information about A 4, B 3, C 4, D 5, E After 2 exchanges 3, C initially as shown 4, B 5, C 4, D 5, E Now A goes down or link After 3 exchanges 5, C between A and B Brakes 6, B 5, C 6, D 5, E After 4 exchanges 5, C At the first packet After … exchanges . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . exchange B will not ∞ ∞ ∞ ∞ ∞ receive any message from A •The number of exchanges required depends But C tells B that it has a on the numerical value used for infinity. path to A of length 2 B now updates its own •In RIP the value is kept 16, that’s why it information about A can’t be used in large systems according updation algo and make it 3
Computer Networking / Module III/ AKN / 104

Open shortest path First (OSPF)
It is based on link state routing that uses dijkstra’s algorithm Link state routing
In this scheme, each router shares the knowledge about its own neighbours to all other routers using flooding Each router maintains a database about its neighbours and sends it when there is a change or after a large period. The idea is that all routers should have a complete topology of the network. From this topology the router can calculate the shortest path between itself and the destination network using dijkstra’s graph algorithm The topology is represented as a graph, where vertices are networks or routers and edges are links. A cost is associated with each link
Computer Networking / Module III/ AKN / 105

Link state Routing
Learning about neighbours
When router is booted, it sends a hello packet on each point-to-point line The router at the other end sends back a reply

Measuring Link cost
One echo packet is sent and its time is recorded, other side sends the packet back immediately and the time of receiving is recorded again The test is conducted several times and the average RTT is calculated for better result

Building the Link state packets
Identity of sender, sequence #, age, a list of neighbours with their link costs
Computer Networking / Module III/ AKN / 106

Link state Knowledge
Whole topology can be compiled from the partial knowledge of each node

Computer Networking / Module III/ AKN / 107

Formation of shortest path tree
The dijkstra’s algorithm creates a single source shortest path tree given a graph(topology), each node is assigned a cumulative cost from root to that node (called weight or total cost)

Computer Networking / Module III/ AKN / 108

Lecture III
Transport Layer
• User Datagram Protocol • Transmission Control protocol • Congestion Control and Quality of services

Computer Networking / Module III/ AKN / 109

Transport Layer

Protocols used for Transport Layer are UDP or TCP The responsibility of transport layer is to deliver the message to the receiving process/Application. i.e. process to process delivery
Computer Networking / Module III/ AKN / 110

Review
Internet layer provides a host-to-host packet delivery The next problem is turn this service to process-toprocess delivery The Transport layer supports communication between the end application programs, thus called end-to-end protocol The underlying networks upon which the transport protocol operates has certain limitations like, it may
Drop messages Reorder Messages Deliver duplicate copies of messages Limit messages to some finite size Delivery messages after a long delay Computer Networking / Module III/ AKN / 111

Review
The operating system supports multiprogramming But specifying that a particular process on a particular machine is the ultimate destination for a datagram is misleading, because
Processes are created and destroyed dynamically(pid), senders seldom know enough to identify a process on another machine Processes may be replaced without informing to the senders We need to identify destinations from the functions they implement without knowing the process

Instead of thinking a process as the ultimate destination, we will imagine that the machine contains a set of abstract points called protocol ports (integer nos.) Computer Networking / Module III/ AKN / 112

Review

Operating system provides two types of access to ports 1. Synchronous access
computation stops during a port access operation. i.e. if a process attempts to extract data from a port, then the operating system temporarily blocks the process till data is passed to the process and then restarts it

2. Asynchronous access
Ports are buffered, so that data arrives before a process is ready to access will not be lost To achieve buffering the protocol software places the packets that arrive for a particular protocol port in a (finite) queue

Each message must carry the destination port on Computer Networking / Module III/ AKN / 113 source

Types of data deliveries

Computer Networking / Module III/ AKN / 114

Port Addressing
At transport layer, port number is used to deliver a message to the correct process out of several processes running on destination host Port numbers are 16 bit integers between 0-65535. The client program defines itself with a port number, chosen randomly by transport layer called ephemeral port numbers The server program uses well known port number. i.e. client gets a new port number each time it runs, but the port number for server is fixed IANA defines some ranges
Well-know ports: 0-1023 are assigned and controlled by IANA for some well-know server processes Registered ports: 1024-49151 are not assigned or controlled by IANA, but can be used by processes Dynamic ports: 49151-65535 are neither controlled nor registered, called Computer Networking / Module III/ AKN / 115 ephemeral ports

Other features
Socket Address
The IP address and port number pair defines the socket address The client and server’s socket addresses define client and server processes uniquely A pair of socket address (client and server’s) uniquely defines a connection.

Multiplexing and demultiplexing
At the sender side, there may be several processes need to send packets, but there is one transport layer protocol. Therefore the protocol accepts messages from different processes differentiated by their port numbers and interleaves them At the receiver side, the transport layer receives interleaved packets from network layer and passes to appropriate application after processing Computer Networking / Module III/ AKN / 116

Other features contd.

Connection-less vs connection-oriented service
In a connection less service, packets are sent from one party to another, without establishing the connection In case of connection oriented, a connection is established, data transferred, then connection is released

Reliable vs unreliable
Reliability is achieved by providing error and flow control at transport layer (data transmission) It becomes a slower and more complex service Where as unreliable services are faster and simple to implement (real-time application)
Computer Networking / Module III/ AKN / 117

The User Datagram Protocol (UDP)
It is the simplest possible transport protocol that extends the host-to-host delivery into a process-toprocess communication service. It only adds a level of demultiplexing, s.t. multiple application process on each host are allowed to share the network. Aside from this requirement, UDP adds no other functionality to the best effort service. UDP provides an unreliable connection less delivery service. It uses IP to carry messages, but adds the ability to distinguish among multiple destinations within a given host computer. Computer Networking / Module III/ AKN / 118

The UDP message format
UDP Source Port UDP message length UDP Destination Port UDP Checksum Data . . .

Port nos may vary from 0-65535, and source port is optional. These are used to demultiplex datagrams The Length field contains a count of datagram in octets. Minimum length is 8 Checksum is optional and zero is kept if not computed The UDP checksum provides the only way to guarantee that data has arrived intact and should be used
Computer Networking / Module III/ AKN / 119

Checksum Calculation
UDP uses the same checksum algorithm as IP But UDP covers more information than is present in UDP datagram
It prepends a pseudo-header to the UDP datagram Appends an octets of zeros to pad the datagram to an exact multiple of 16 bits And computes checksum over entire object

UDP pseudo-Header Source IP Zero Destination IP Protocol UDP Length
Computer Networking / Module III/ AKN / 120

Checksum Calculation (contd.)
Checksum calculation at the Sender end.
Add pseudo-header to the user datagram Fill the checksum field with zeros Divide the total bits in to 16 bit words If total bytes are not even, add one byte of all zeros Add all 16-bit sections using one’s complement arithmetic Complement the result and insert the result in checksum field Drop the pseudo header and any padding used Deliver the datagram

Checksum calculation at the Receiver end.
Perform the operation same as above If complement is zero drop pseudo-header and padding and accept the datagram. Otherwise discard the datagram
Computer Networking / Module III/ AKN / 121

Checksum Calculation (contd.)
153.18.8.105 171.2.14.10 Zero 1027 15 U E Assignment
Calculate the checksum of the user datagram at sender side and also test it for the receiver side
Computer Networking / Module III/ AKN / 122

17

15 13 0 D S P T T padding

Checksum Calculation an example

Computer Networking / Module III/ AKN / 123

Problem with Checksum Calculation
Pseudo-header contains source and destination IP addresses i.e. IP addresses must be known at UDP layer Destination IP address is supplied by the user. But what about source IP, which is yet to be computed in IP layer?
Solution 1: UDP software asks the IP layer to compute addresses Solution 2: UDP software computes addresses and after checksum calculation sends it to IP layer. IP layer need to fill remaining IP header fields

But any of the solution violates the abstraction of layers i.e. It is clearly a compromise of pure separation needed for practical reasons

Computer Networking / Module III/ AKN / 124

UDP Operation
Connection less service
Each datagram sent by UDP is an independent datagram. Data grams are not numbered, also there is no connection establishment thus different datagrams may follow different path It cannot send a stream of data, i.e. each request must be small enough to fit into one user datagram

Flow and error control
No flow control hence no window mechanism. Receiver may overflow No error control hence sender does not know if a message is lost or duplicated
Computer Networking / Module III/ AKN / 125

Multiplexing and Demultiplexing
In a host running a TCP/IP software, there is only one UDP but possibly several processes, that need to use services of UDP Port1 Port2 Port3 Port1 Port2 Port3

UDP Multiplexer IP

UDP DeMultiplexer IP

• At sending side UDP accepts messages from different processes, differentiated by their port nos.Then it is passed to IP layer • At receiving side UDP receives datagrams from IP. After error checking drops the header and delivers to the appropriate processes
Computer Networking / Module III/ AKN / 126

Well known ports used for UDP

Computer Networking / Module III/ AKN / 127

Use of UDP
It is suitable for process that requires simple and fast request-response communication like DNS Suitable for process with internal flow and error control mechanism like tftp Suitable for multicasting Used for management process such as SNMP Used for route update protocols like RIP
Computer Networking / Module III/ AKN / 128

Reliable Stream Transport Service
Stream Orientation
Data is converted into stream of bits, divided into octets at source machines The stream delivery service on the destination machine passes to the receiver exactly the same sequence of octets that the sender has passed.

Virtual Circuit Connection
Before data transfer can start, both the applications interact with their respective OS for a connection
i.e. one application places a call, which must be accepted by the other
Computer Networking / Module III/ AKN / 129

Properties of Reliable Delivery Service
During transfer, protocol software on the two machines continue to communicate to verify that data is received correctly otherwise report the failure to appropriate S/W for necessary action Therefore, Application programs view the connection as a dedicated H/W circuit. The reliability is an illusion provided by the stream delivery service called virtual circuit

Buffered Transfer
The protocol software is free to divide/combine the stream into packets independent of pieces the application program transfers. At the sending side, a PUSH mechanism forces protocol S/W to transfer all the data that has been generated without waiting to fill a buffer. At the other end PUSH causes it to make the data available to application without delay Computer Networking / Module III/ AKN / 130

Properties of Reliable Delivery Service
Unstructured Stream
TCP/IP stream service doesn’t honour structured data stream i.e. There is no way for a payroll application to have the stream service mark the boundaries between employee records

Full Duplex Connection
Connections provided by TCP/IP stream service allow concurrent transfer on both directions The advantage is control information for one stream can be send back to the source in datagrams carrying data in the opposite direction
Computer Networking / Module III/ AKN / 131

Transmission Control Protocol
Reliability
+ve acknowledgement with Sender Receiver Pkt Recv Pkt Send Ack Recv Ack Send Pkt retransmission The sender keeps a record of each packet it sends and waits for an ack before sending the next pkt Sender also starts a timer and retransmits a packet if the timer expires before receiving the ack

• Disadvantages

• Duplication of data / Ack due to premature retransmission • To avoid confusion caused by delayed or duplicated Ack, seq. no. is sent back with Ack • Wasting of substantial amount of N/W bandwidth
Computer Networking / Module III/ AKN / 132

END-to-END vs Point-to-Point
1. TCP needs an explicit connection establishment s.t. two parties establish some shared state to enable the sliding window algorithm to begin 2. Variations in RTT are possible due to various reasons.(?) Therefore timeout mechanism that triggers retransmissions must be adaptive. 3. How late a packet can arrive at the destination? IP throws packets away after their TTL expires, TCP assumes that each packet has a max. segment life time(MSL).
TCP has to be prepared for very old packets to suddenly show up at the receiver, potentially confusing the sliding window algorithm.
Computer Networking / Module III/ AKN / 133

END-to-END issues
4. In case of point-to-point link
delay × bandwidth ≈ window size ≈ buffer space

The amount of resources dedicated to any one TCP connection highly variable, especially considering that any one host can potentially support hundreds of TCP connections at the same time
i.e TCP must include a mechanism that each side ‘learn’ what resources the other side is able to apply to the connection

5. TCP connection has no idea what links will be traversed to reach at the destination.
The sending machine might be connected directly to a relatively fast Ethernet and somewhere in the middle a slower link has to traversed, which leads to ‘congestion’

Computer Networking / Module III/ AKN / 134

TCP Segment
Appl process
Write bytes

Appl process
Read bytes

TCP is a byte oriented protocol. i.e. It describes the service provided to appl. process.

TCP Send buffer
segment

TCP Recv The pkts exchanged between buffer TCP peers are called segments
segment

TCP has three mechanisms to trigger the transmission of a segment
1. TCP maintains a variable, maximum segment Size (MSS), and it sends a segment as soon as it has collected MSS bytes from sending process 2. Sending process invokes push operation to effectively flush the buffer of unsent bytes 3. A timer that periodically fires; the resulting segment contains as many bytes as are currently in buffer
Computer Networking / Module III/ AKN / 135

TCP Segment Header Format
0 4 10 16 19 24 31

Src Port

Dst Port

Sequence Number Acknowledgement Flags HLen unused Advertised window Urgent pointer Checksum Options (variable length) Data ...
Computer Networking / Module III/ AKN / 136

Padding

TCP Header Format Explanation
SrcPort and DstPort, identify the source and destination application programs respectively
A TCP connection is identified by a 4-tuple {SrcPort, SrcIPAddr, DstPort, DstIPAddr}

Because TCP is a byte oriented protocol, each byte of data has a sequence number
SeqNum field contains the sequence number for the first octet of data carried in that segment Ack field defines the octet number that is expected next AdvertisedWindow contains the buffer space available at seqNum receiver Receiver Sender
Ack+advWin
Computer Networking / Module III/ AKN / 137

TCP Header Format Explanation
Flags: 6 bits, when set it is understood as follows
5. 6. 4. 3. 1. 2. SYN: Synchronize seq. nos during connection FIN: Terminate the connection RESET: reset the connection PUSH: request for push URG: urgent pointer is valid ACK:

Urgent pointer specifies the position, where the urgent data ends. Options: TCP header can have 40 bytes of optional information
Computer Networking / Module III/ AKN / 138

TCP Header Options
Max Seg Size(MSS): 4bytes determined at the time of connection establishment Window Scale factor:3bytes
Used to increase the window size New window size=window size × 2scaleFactor Largest value possible for scale factor is 16 i.e. 216 × 216 = 232 max size of seq. number

Time Stamp: 10 bytes
Used to calculate round trip time
Computer Networking / Module III/ AKN / 139

Connection Establishment
Client SYN, seq Num=x
SYN+

Server The algorithm used is called three-way-handshaking

=y eqNum ACK s

ACK=

y+1

The client sends a segment to the server stating (flags=SYN, seqNum=x ) Then server responds with a single segment that both acknowledges (Flags=ACK, Ack=x+1) and states it own beginning seqNum (Flags=SYN, seqNum=y) Finally client responds with a third segment that acknowledges the server’s sequence number (flags=ACK, Ack= y+1)
Computer Networking / Module III/ AKN / 140

Connection Termination, four-way-handshaking
Client FIN, seqN um=x
x+ AC K = 1

Server

The client sends a segment to the server stating (flags=FIN, seqNum=x ) Then server responds with a single segment that acknowledges (Flags=ACK, Ack=x+1) now the connection is in half close mode. i.e. server can send data (remaining) but client can’t

F ACK=

y qNum= IN, se

y+1

Finally server sends a segment to the client stating (flags=FIN, seqNum=y ) The client responds with a segment that acknowledges the server’s sequence number (flags=ACK, Ack= y+1)
Computer Networking / Module III/ AKN / 141

Connection Resetting
TCP may request for resetting a connection. i.e. the current connection is destroyed. Resetting is done in one of the following three cases
The TCP of one side has requested a connection to a non-existent port. TCP of other side sends a segment with RST bit set One TCP may want to abort the connection due to an abnormal situation The TCP on one side may discover that the TCP on the other side has been idle for a long time
Computer Networking / Module III/ AKN / 142

TCP State Transition
To keep track of all the different events during connection establishment to connection termination The TCP of both sides are implemented as a finite state machine and is represented in a state transition diagram Notations
The states are shown using ovals Transition from one state to another is shown using directed lines Each line is contains two strings separated by slash. First string is input to TCP and second is output Dotted lines represent server and solid lines represent client
Computer Networking / Module III/ AKN / 143

State transition diagram
Client Diagram
Starts in CLOSED state When receives an Active open request from client application, it sends a SYN segment to server and goes to SYN-SENT state Client TCP receives a SYN+ACK segment from server TCP. It sends an ACK to server TCP and goes to ESTABLISHED state This is the data transfer state. Client remains in this state till data transmission continues

Computer Networking / Module III/ AKN / 144

State transition diagram contd.
Client Diagram
Client TCP receives a close request from its application program. It sends a FIN segment to the other TCP and goes to FINWAIT-1 state When the ACK is received from server TCP, it goes to FINWAIT-2 state. The connection is closed in one direction Client receives a FIN segment from server TCP and sends an ACK and goes to TIME-WAIT state When client TCP is in this state it starts a timer and waits till the timer goes off. The value of this timer is set to double the MSL The client TCP remains in this state to let all duplicate packets, if any arrive to be discarded. After the time-out the client goes to CLOSED state again
Computer Networking / Module III/ AKN / 145

State transition diagram contd.
Server Diagram
Server TCP starts with CLOSED state It receives a passive open request from the server application and goes to LISTEN state IT now receives a SYN segment from the client TCP and sends a SYN+ACK segment to client TCP and goes to SYN-Rcvd state It then receives ACK from client TCP and goes to ESTABLISHED state. Data transfer occurs between client and server applications After data transmission it receives a FIN segment from client TCP, it now sends an ACK and goes to CLOSE-WAIT state Server TCP receives a close request from server application program and sends a FIN segment to client TCP and goes to LAST-ACK state When it receives the last ACK from client it goes to CLOSED state again Computer Networking / Module III/ AKN / 146

TCP’s Sliding Window
1. It guarantees the reliable delivery of data, 2. It ensures data is delivered in order and 3. It enforces flow control between sender and receiver The algorithm places a small, fixed size virtual window on the stream sequence and transmits all octets that lie inside the window without receiving an Ack. Three pointers are maintained into the send buffer
Sending Application TCP LastByteWritten Receiving Application TCP LastByteRead

LastByteAckd

LastByteSent

NextByteExpected

LastByteRecvd

Direction of transmission

Computer Networking / Module III/ AKN / 147

Reliable and Ordered Delivery
TCP on sending side maintains a send buffer, this buffer is used to store data that has been sent but not yet acknowledged, as well as data that has been written by the sending application, but not transmitted On other side, TCP maintains a receive buffer that holds data that arrives out of order, as well as the data that is in correct order but that application process has not yet read it The relations among send buffer pointers can be as follows
LastByteAckd ≤ LastByteSent and LastByteSent ≤ LastByteWritten

bytes to the left of LastByteAcked and bytes to the right of LastByteWritten need not be saved / Module III/ AKN / 148 Computer Networking

Reliable and Ordered Delivery
Similarly at the receive buffer is true As a byte cannot be read by the application until it is received

LastByteRead < NextByteExpected

NextByteExpected ≤ LastByteRecvd + 1
i.e. if data has arrived in order, NextByteExpected points to the byte after LastByteRecvd if data has arrived out of order, NextByteExpected points to the start of the first gap in data The bytes to the left of LastByteRead need not be buffered because they have already been read by the local process bytes to the right of LastByteRecvd need not be buffered because they have not yet arrived.
Computer Networking / Module III/ AKN / 149

TCP Flow Control
Both buffers are of finite size defined by MaxSendBuffer and MaxRcvBuffer. Receiver sends a window advertisement that it can buffer. At receiving side, it maintains as
overflowing its buffer, it therefore advertises a window size of
AdvertisedWindow = MaxrecvBuffer- ((NextByteExpected-1) LastByteRead) i.e. the free space remaining in receive buffer

LastByteRecvd – LastByteRead ≤ MaxRcvBuffer to avoid

NextByteExpected-1 is same as LastByteExpected in case of inorder receive, it will be different if out of order receive

If the receiving process is reading data just as fast as it arrives, then the advertised window stays open.
Computer Networking / Module III/ AKN / 150

TCP Flow Control
If the receiving process falls behind, then advertise window shrinks and eventually goes to zero On the other hand sender end TCP ensures that
LastByteSent – LastByteAcked ≤ AdvertisedWindow
i.e. it calculates How much data it can send as

EffectiveWindow = AdvertisedWindow – (LastByteSent – LastByteAcked) i.e. how much extra
bytes it can send

Also sending side should ensure that the local process doesn’t overflow the send buffer, that is

LastByteAcked ≤ MaxSendBuffer tries to LastByteAcked) + y > Computer Networking LastByteAcked) + itten – write y bytes and (LastByteWritten – / ModuleTCP blocks MaxSendBuffer then III/ AKN / 151

TCP Flow Control
How does the sending side know that the advertised window is no longer zero?
i.e. once the receiver side has advertised a window size of 0, the sender is not permitted to send any more data, which mince it has no way to discover that the advertised window is no longer zero at some time in the future.

Solution: the sending side persists in sending a segment with one byte of data every so often. The data may not be accepted but eventually it gets a response whenever send buffer becomes free. The size of MSS is set to MTU of the directly connected network minus the size of TCP and IP header s.t. can be sent without fragmentation
Computer Networking / Module III/ AKN / 152

Adaptive Retransmission
TCP retransmits each segment if an Ack is not received in a certain period of time(RTT) But choosing an appropriate timeout value is very difficult and TCP uses adaptive retransmission mechanism Original Algorithm:
TCP sends a data segment, records the time. When Ack for that segment arrives, it reads the time again. Difference between two times gives a SampleRTT. TCP then computes a weighted average between the previous estimate and this new sample as EstimatedRTT = α × EstimatedRTT + (1 - α) × SampleRTT α between 0.8 and 0.9 used to smooth the EstimatedRTT
Computer Networking / Module III/ AKN / 153

Adaptive Retransmission
Then TimeOut = 2 × EstimatedRTT Problems
Ack does not acknowledges a transmission but receipt of data. i.e. it is difficult to associate an ACK with an transmission or retransmission Associating the ACK with original transmission may be an over estimate and associating with retransmission may be an under estimate as shown in two figures Solution?

Sender
SampleRTT

Receiver Sender
SampleRTT

Receiver
ansmissio n
AC K

Original Tran smission Retransmiss ion

Original T r

A CK

Retran smissio n

Original transmission

Retransmission
Computer Networking / Module III/ AKN / 154

Congestion Control
Congestion is a situation which may occurs when the load on the network is greater than the capacity of the network i.e. The number of packets sent to the router is much more then the Number of packets the router can handle. Router has so many packets queued that it runs out of buffer space and has to start dropping packets, which is a worst condition Therefore to control the congestion we try to avoid heavy data traffic that may cause congestion
If the rate of packet arrival rate is higher than processing rate then input queues becomes longer

If the rate of packet departure rate is higher than processing rate then output queues becomes longer

Computer Networking / Module III/ AKN / 155

Traffic descriptors
Average data rate = amount of data/total time Peak datarate= max datarate of the traffic Max. burst size= max length of time the traffic is generated at the peak rate Effective bandwidth= is a function of average datarate, peak data rate, and max. burst size

Computer Networking / Module III/ AKN / 156

Traffic Profiles
Constant-bit-rate traffic: Datarate is constant throughout

Variable bit rate: The rate of data flow changes in time

Bursty: The datarate changes suddenly in a very short period of time. This type of traffic creates congestion in a network.
Computer Networking / Module III/ AKN / 157

Network performance
Delay vs Load
When load is much less than the capacity of the network, the delay is at a minimum Delay composed of propagation delay and processing delay, which is negligible! When load reaches the network capacity, the delay increases sharply because waiting time is added to the delay

Throughput vs Load
Throughput is the number of packets passing through the network in unit time when the load is below capacity, the throughput increases proportionally with load When load reaches the network capacity, throughput declines sharply due to discarding of packets followed by retransmissions further makes things worse
Computer Networking / Module III/ AKN / 158

Congestion Control

Two categories of mechanisms for congestion control
Open Loop: congestion prevention Closed Loop: congestion removal

Open Loop: preventing congestion
Retransmission policy
The retransmission policy and retransmission timers must be designed to optimize the efficiency and to prevent congestion

Window Policy
The selective repeat is better than Go-Back-N policy for congestion control?

ACK Policy
If ACK is not received, sender slows down, help prevent congestion

Discarding Policy
Selective discarding of less sensitive packets when likelihood of congestion increases

Admission Policy

Before admitting for a flow it checks the resources

Computer Networking / Module III/ AKN / 159

Congestion Control: closed Loop
Closed Loop: removal of congestion, if occurs
Back Pressure
Router informs previous routers to slow down (recursive)

Choke Point
Router informs source to slow down by sending a special packet

Implicit Signaling
Source predicts about congestion and slows down (like delay in getting ACK)

Explicit Signaling
Router sends an explicit signal by setting a bit in the packet Backward signaling:The bit can be set in a packet moving in the opposite direction. This bit warns the sender to slow down Forward signaling:The bit can be set in a packet moving in the direction of congestion. This bit warns the destination to slow down. Receiver slows down sending ACK
Computer Networking / Module III/ AKN / 160

Congestion Control TCP

When congestion occurs in a router and some packets might be dropped, then sender retransmits those packets. This may create more congestion and more dropping of packets. The condition become so worse that the system can pass no more data. This situation is called congestion collapse i.e. If the cause of the lost segment is congestion, retransmission of the segment does not remove the cause—it aggravates it. To avoid this situation, TCP assumes that the cause of a lost segment is due to congestion in the network and takes necessary action to remove Networking / Module III/ AKN / 161 Computer congestion.

Congestion Control TCP contd.
The window size is decided not only by the receiver’s advertisement but also by congestion in the network Actual Window = Min(receiver’s window, Congestion window) Congestion avoidance
To avoid congestion we have two strategies
Slow start and additive increase till there is no congestion Multiplicative Decrease, if congestion occurs
Computer Networking / Module III/ AKN / 162

Congestion avoidance
Slow start
At the beginning of a connection TCP sets the congestion window size = 1MSS For each segment ACK it receives the congestion window size is increased by 1 MSS till it reaches a threshold value = ½ of allowable window size i.e. ACK for 1 seg –> congestion window size = 2 MSS ACK for 2 segs -> congestion window size = 4 MSS ACK for 4 segs -> congestion window size = 8 MSS . . . -> congestion window size = ½ advt. Window

Additive Increase
After the size reaches the threshold, it increases the size by one for each received ACK. i.e. ACK may be received for several segments but increase is only by 1 MSS
Computer Networking / Module III/ AKN / 163

Congestion avoidance
This strategy continues till it receives ACK before time-out or congestion window size = advt. Window size.

Multiplicative Decrease
The only way to guess that a congestion has occurred is through a lost segment. i.e. if the sender does not receive ACK before time-out If congestion occurs than threshold value is set to ½ of congestion window and congestion window is set to 1MSS again

Computer Networking / Module III/ AKN / 164

Congestion control in frame relay
Frame relay is designed for high throughput and low delay but congestion decreases throughput and increases delay Frame relay does not have flow control, but allows user to transmit bursty data that can cause congestion For congestion avoidance, Frame relay protocol uses 2 bits the frame to warn the source and destination about the congestion.
Backward Explicit congestion Notification (BECN) bit Forward Explicit congestion Notification (FECN) bit
Computer Networking / Module III/ AKN / 165

BECN bit
It warns the sender about congestion in the network using two methods
Method 1: the switch uses response frames from the receiver Method 2: the switch can use a predefined connection, DLCI=1023 to send special frames for this specific purpose Sender responds by reducing data rate

Computer Networking / Module III/ AKN / 166

FECN bit
Used to warn the receiver about the congestion If there is an ACK mechanism at the higher level the receiver can delay the ACK, thus forcing the source to slow down

Four cases of congestion in Frame Relay

Computer Networking / Module III/ AKN / 167

Quality of Service (QoS)
Is an assurance from the network for a particular kind of service e.g. network uses retransmission strategy to make sure that data arrives correctly. This service is ok for non-real time application. But may not be ok for real-time applications as it does-not guarantee timeliness i.e. we need a new service model in which, application that need higher assurances can ask the network for that A network that can provide these different level of services is said to support QoS.
Computer Networking / Module III/ AKN / 168

Flow characteristics
Reliability
Lack of reliability means losing a packet or ACK, which may or may not needs retransmission Example: Email, file transfer needs retransmission Audio and video may not need retransmission

Delay (Source-to-destination delay)
Application can tolerate delay in different degrees Example: multimedia application need minimum delay, but in case of file transfer or email it is less important

Jitter
Is a variation in delay for packets belonging to same flow. Audio and Video cannot tolerate high jitter No effect for file or mail transfer

Bandwdth
Different application needs different BW In video transmission we need million of bits to refresh a color screen While total no of bits in an email may not reach even a million

Computer Networking / Module III/ AKN / 169

Techniques to Improve QoS
Common methods are scheduling, traffic shaping, admission control,and resource reservation Scheduling (FIFO, priority and weighted fair queuing)
When packets from different flows arrive at a router, It is needed to treat the different flows in a fair and appropriate manner. Some techniques are as follows FIFO Queuing with tail drop
In this queuing, packets wait in a buffer until the node is ready to process them If average arrival rate is higher than the average processing rate, the queue will fill up and new packets will be discarded without regard to which flow the packet belongs to or how important the packets is? It is simplest and most widely used in Internet routers

Computer Networking / Module III/ AKN / 170

Scheduling Techniques contd.
Priority Queuing
Each packet is marked with a priority class The router implements multiple FIFO queues, one for each priority class It processes packets of higher priority first and moves on to the next priority if the higher priority one is empty If there is a continuous flow in a high priority queue, then this will create a starvation problem in others Therefore this should be optimized to put hard limits on how much high priority traffic can be inserted in the queue These scheduling is used in Internet to protect most important packets like routing updates

Computer Networking / Module III/ AKN / 171

Scheduling Techniques
Weighted fair Queuing
The packets are still assigned to different classes before inserting to the queues The router than serves queues in around-robin fashion according to the weight of the queue i.e. for above example: 3 pkts from first, two from 2nd and one from 3rd queue

Computer Networking / Module III/ AKN / 172

Traffic shaping
Is a mechanism to control the amount and the rate of the traffic sent to the network. Two techniques used 1. Leaky Bucket, 2. Token Bucket

Leaky Bucket
The idea is to have a constant bit rate traffic in the network in spite of bursty data coming from source.
i.e. if a bucket has a small hole at the bottom, the water leaks from the bucket at a constant rate and is independent of the rate of input to the bucket
Computer Networking / Module III/ AKN / 173

Leaky bucket implementation
When the packets are of same fixed size then one packet can be pushed to network per clock tick If packets are of variable size than more packets per tick may be allowed. i.e. if rule is 1024 bytes per tick then one 1024 byte packet is allowed per tick, two 512 bytes per tick and four 256 byte packets per tick and so on

Algorithm
for each clock tick{ 1. Initialize a byte counter to n 2. while n ≥ size of the packet 3. send the packet and decrement the counter by the packet size. 4. Stop the transmission till next tick } Where n is max number of bytes allowed per tick

Computer Networking / Module III/ AKN / 174

Leaky bucket contd.
A leaky bucket algorithms shapes bursty traffic into fixed-rate traffic by averaging the data rate. The packets will be dropped if the buffer is full This algorithm prevents congestion by avoiding instantaneous heavy traffic at the output line The buffer capacity should be carefully designed s.t. it should be able to store the bursty data for short period of time, otherwise packets will be dropped Example:
data comes at a rate 25 Mbps, one 40ms burst every second. Design the leaky bucket

Solution:
total data per sec = 25Mbps * 40 *10-3 = 1Mb Thus capacity of buffer can be chosen as 1Mb Uniform output rate may be chosen as 2Mbps, s.t. it will take 500ms to drain the complete data
Computer Networking / Module III/ AKN / 175

Token Bucket
The leaky bucket is restrictive. i.e. if a host is idle then bucket becomes empty, if the host has bursty data then bucket allows only an average rate. But the token bucket algorithm allows idle hosts to accumulate credit for the future in form of tokens Algorithm:
token bucket holds tokens generated by a clock at the rate of one token per ∆T sec or n tokens per sec It consumes one token per packet sent i.e. to send a packet there should be a token available in the bucket

Computer Networking / Module III/ AKN / 176

Token Bucket contd.
Leaky bucket and Token Bucket provides different kind of traffic shaping
The Leaky bucket algorithm does not allow idle hosts to save up permission to send large bursts later But the token bucket algorithm does allow saving, up to the max size of bucket. i.e. bursts upto the size of bucket can be sent at once The token bucket algorithm throws away tokens when the bucket fills up but never discard packets. But the Leaky bucket discards packets when bucket fills up

One variation to Token bucket
Each token represent the right to send k bytes in place of one packet. A packet can only be sent if enough tokens are available to cover length in bytes. Fractional tokens are kept for future use
Computer Networking / Module III/ AKN / 177

Quality of Service (QoS)
Two models have been proposed to provide Quality of Service in the Internet
Integrated Services (IntServ)
Is a flow based QoS model designed for IP. i.e. a user needs to create a flow, a kind of virtual circuit, from the source to destination and inform all routers about the resource requirement.

Differentiated Services (DiffServ)
Is a class based QoS model designed for IP. i.e. the applications, or hosts, define the type of service they need each time they send a packet.
Computer Networking / Module III/ AKN / 178

Integrated services features
Signals
IP is a connection less protocol To implement a flow based service a signaling protocol is used to run over IP that provides the signaling mechanism for making reservation The protocol is named as Resource Reservation Protocol

Flow Specification
has two parts: Rspec and Tspec Rspec(resource specification)
Defines the resource that the flow needs to reserve (buffer, bw etc.)

Tspec(Traffic specification)
Traffic characterization of the flow

Admission
After receiving flow specification the router decides to admit or deny the flow Computer Networking / Module III/ AKN / 179

Integrated services features
Guaranteed Service Class

Two service classes are defined
Designed for real time traffic that needs guaranteed minimum end-to-end delay. (multimedia) end-to-end delay = sum of delays in routers + propagation delay + setup mechanism Only delay in router can be guaranteed by router The amount of end-to-end delay and the data rate must be defined by the application

Controlled-Load Service Class
Designed for applications that can accept some delays, but are sensitive to an overload network and to the danger of losing packets Example application are file transfer, email etc.
Computer Networking / Module III/ AKN / 180

Resource ReserVation Protocol (RSVP)
The resource reservation protocol is a signaling protocol to help IP create a flow and consequently make a resource reservation The signaling system of RSVP is designed for multicasting to enable it to provide resource reservation for all kinds of traffic including multimedia, which often uses multicasting In this case the receivers (not the sender) makes the reservation It has several types of messages for above tasks. Two of them are used for resource reservation, i.e. Path message and Resv message
Computer Networking / Module III/ AKN / 181

RSVP Path message
A Path message travels from the sender and reaches all the receivers (downstream) in multicast path On the way path message stores the necessary information for the receivers. A new message is created when the path diverges.

Computer Networking / Module III/ AKN / 182

RSVP Recv message
Receiver sends a recv message, which travels towards sender (upstream) and makes a resource reservation on the routers that support RSVP If a router does not support RSVP on the path, it routes packet using traditional delivery methods Reservation merging
Resources are not reserved for each receiver in a flow. Reservation is merged to larger of the two (or more) requests As different qualities for multimedia is required by different receivers, thus different requirements for the same flow Computer Networking / Module III/ AKN / 183

Reservation Styles
When there are more than one flow, the router needs to make a reservation to accommodate all of them

RSVP defines three types of reservation styles
Wild card Filter: router creates a single reservation for all senders based on largest request. This is used when flow from different receivers do not occur at the same time Fixed Filter: router creates a distinct reservation for each flow. It is used when there is a high probability that from different receivers occurs at the same time Shared Explicit: creates a single reservation which can be shared by a set of flows
Computer Networking / Module III/ AKN / 184

Differtiated services
Problems with integrated services
Scalability
This model requires that each router keep information for each flow, which is impractical as load on routers will increase

Service type limitation
It provides two services 1. Guaranteed and control load

Solutions
The routers do not have to store information about flows. i.e. The applications, or hosts, define the type of service they need each time they send a packet The per-flow service is changed to per class service. The router routes the packet based on the class of service This is called Differentiated services Computer Networking / Module III/ AKN / 185

Differentiated service features
Each packet contains a field called DS field. The value of this field is set by the first router designated as the boundary router. contains two sub-fields:
Differentiated services code point: defines per hop behavior (PHB) and an unused field DE PHB(default PHB) same as TOS. EF PHB (expedited forwarding) provides following services like Low loss, Low latency, Ensured bandwidth. AF PHB (Assured forwarding) delivers the packet with a high assurance as long as the class traffic does not exceed the traffic profile of the node. The users of the network need to be aware that some packets may be discarded
Computer Networking / Module III/ AKN / 186

Traffic conditioner
The DS node uses traffic conditioners like
Meters: checks to see if the incoming flow matches the negotiated traffic profile Marker: can remark a packet that is using best-effort delivery or down-mark a packet based on information received from the meter. Shaper: reshapes the traffic if not compliant with negotiated traffic Dropper: discards a packet if flow severely violates the negotiated profile

Computer Networking / Module III/ AKN / 187

END of module III

Thank You
Computer Networking / Module III/ AKN / 188