B Fundamentals

8/3/2019 B Fundamentals

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1HPN Fundamentals

Fundamentals of Communication Systems

1. Layering and the Internet

2. Application Layer in a Nutshell

3. Transport Layer in a Nutshell

4. IP and Routing in a Nutshell

5. Link Layer in a Nutshell

6. Physical Layer in a Nutshell

This section summarizes and harmonizes material which basically should be

known from Bachelor courses (or similar).

Copyright 2011/2012 Computer Science 4, University of Bonn


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2HPN Fundamentals

1. Layering and the Internet


1.1 What is the Internet?

1.2 How does the Internet work?

1.4 Delays, Losses, etc.

1.3 Layers

1.5. Protocol Architecture in Real Life: Introducing WireShark


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3HPN Fundamentals

1.1 What is the Internet?



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4HPN Fundamentals

The first Internet

Router

Router

The first internetwork became reality in October 1977:

Packet Radio Network in San Francisco

ARPANET across the USA

SATNET via satellite across the Atlantic to London.

Packet Radio

Network

SATNET

ARPANET

Packet Switching may be used for forwarding packets across ... a specific network

a network of networks (internetwork, internet).



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5HPN Fundamentals

A nuts and bolts view


Source: Jim Kurose, Keith Ross: Computer Networking: A Top

Down Approach - 5th edition, Addison-Wesley, April 2009.

millions of connected computing devices:

hosts(end systems)

running network apps Web, VoIP, email, games, e-commerce,

file sharing

communication links

fiber, copper, radio, satellite

different transmission rates andproperties

routers

forward packets (chunks of data)


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6HPN Fundamentals

A closer look at network structure




network edge: applications and hosts

access networks

network core: interconnected routers

network of networks


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7HPN Fundamentals

Network edge and services




end systems (hosts): run application programs

e.g. Web, email

at edge of network

communication services provided toapps:

reliable data delivery from source to

destination

best effort (unreliable) data delivery

client/server model client host requests and receives

service from always-on server

e.g. Web browser/server; email

client/server

peer-peer model: minimal (or no) use of dedicated

servers

e.g. Skype, BitTorrent

client/server

peer-peer


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8HPN Fundamentals

Access networks and physical media




How to connect end systems to edge router?residential access nets

institutional access networks (school,

company)

mobile access networks


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9HPN Fundamentals

Dial-up Modem and Digital Subscriber Line (DSL)


telephone

network Internet

home

dial-up

modem

ISP

modemhome

PC

central

officeDial-up Modem

uses existing telephony infrastructure

home is connected to central office

up to 56Kbps direct access to router

cant surf and phone at same time:

not always on

telephonenetwork

DSL

modemhome

PC

home

phone

Internet

DSLAM

Existing phone line:0-4KHz phone; 4-50KHz

upstream data; 50KHz-

1MHz downstream data

splitter

central

office

Digital Subscriber Line (DSL)

also uses existing telephone infrastructure

up to 1 Mbps upstream

up to 8 Mbps downstream

Source: Jim Kurose, Keith Ross: Computer Networking: A TopDown Approach - 5th edition, Addison-Wesley, April 2009.




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10HPN Fundamentals

Internet Access via (Wireless) Local Area Networks


Local Area Networks

typically used in companies, universities,

etc

10 Mbs, 100Mbps, 1Gbps, 10Gbps

Ethernet

Wireless access networks

shared wireless access network connects

end system to router

via base station aka access point

100 Mbps

100 Mbps

100 Mbps1 Gbps

server

Ethernet

switch

Institutional

router

To Institutions

ISP

basestation

mobilehosts

router


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11HPN Fundamentals

Typical home network components


wirelessaccesspoint

wirelesslaptops

router/firewall

cablemodem

to/fromcable

headend

Ethernet




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12HPN Fundamentals

Element of a wireless network


network

infrastructure

wireless hosts

r laptop, PDA, IP phone

r run applications

r may be stationary (non-mobile) or mobile

m wireless does notalways mean mobility



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13HPN Fundamentals

Element of a wireless network (2)


network

infrastructure

base station

r typically connected to wirednetwork

r relay - responsible for

sending packets betweenwired network and wirelesshost(s) in its area

m e.g., cell towers, 802.11access points

m handoff: mobile changesbase station providingconnection into wirednetwork



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14HPN Fundamentals

Element of a wireless network (3)



network

infrastructure

wireless link

r typically used to connectmobile(s) to base station

r also used as backbone

linkr multiple access protocol

coordinates link access

r various data rates,transmission distance


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15HPN Fundamentals

Characteristics of selected wireless link standards



Indoor10-30m Outdoor50-200m Mid-rangeoutdoor200m 4 Km

Long-rangeoutdoor5Km 20 Km

.056

.384

1

4

5-11

54

IS-95, CDMA, GSM 2G

UMTS/WCDMA, CDMA2000 3G

802.15

802.11b

802.11a,g

UMTS/WCDMA-HSPDA, CDMA2000-1xEVDO 3G cellular

enhanced

802.16 (WiMAX)

802.11a,g point-to-point

200 802.11n

Datarate(Mbps)

data


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16HPN Fundamentals

Other Wireless Links Spectrum Map US


www.fas.org/spp/military/program/sigint/allochrt.pdf


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17HPN Fundamentals

Alternative Ad-hoc mode / Mesh networks


ad hoc mode

r no base stationsr nodes can only

transmit to other nodeswithin link coverage

r nodes organizethemselves into anetwork: route amongthemselves


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18HPN Fundamentals

Wireless network taxonomy


single hop multiple hops

infrastructure(e.g., APs)

no

infrastructure

host connects to

base station (WiFi,WiMAX, cellular)

which connects to

larger Internet

no base station, no

connection to larger

Internet (Bluetooth,

ad hoc nets)

host may have to

relay through severalwireless nodes to

connect to larger

Internet: mesh net

no base station, noconnection to larger

Internet. May have to

relay to reach other

a given wireless node

MANET, VANET


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19HPN Fundamentals

letter mail

network

abstract network cloud

1.2 How does the Internet work?

Analogy: letter mail

Originator:

- name

- street + number

- ZIP code + city

(several components)

Destination address:

- name

- street + number

- ZIP code + city

(several components)

Max MustermannRmerstr. 164

D-53117 Bonn



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20HPN Fundamentals

Internet

Internet cloud

How does the Internet work?

Originator:

- IP address

- Protocol ID

- Port Number(several components)

Destination:

- IP address

- Protocol ID- Port Number(several components)

Data

Protocol Control

Information

IP Datagram (IP = Internet Protocol)

Router



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21HPN Fundamentals

Packet Switching

In the Internet, the information units (packets, datagrams) travel across a

store-and-forward network of redundant connections:

Packets are stored until they have been forwarded to the next-hop station.



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22HPN FundamentalsCopyright 2011/2012 Computer Science 4, University of Bonn

1.3 Layers

different layer models are used for the specification of communicationprotocols.

layered reference model for discussion

modularization eases maintenance, updating of system change of implementation of layers service transparent to rest of

system

Why layering?


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layer n layer n

The Hierarchy Principle

According to the hierarchy principle

each layersolves specific problems. ( protocol of this layer) layer ndirectly communicates with layer n + 1 (offers service to layern + 1) layer ndirectly communicates with layer n - 1 (uses service provided by layern - 1)

A layer n indirectly communicates with the layer nof the peer instance exchanges data units with a well-defined format with the peer

(Protocol Data Units PDUs).

protocol

services provided by lower layers

station A station B

service access point(SAP)

(layer n)

service access point(SAP)

PDUs


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Important Properties of the Hierarchy Principle

Layer nonly knows

the service access point (SAP) of layer n - 1 the basic characteristics of the service provided by layer n - 1

Layer n is not aware of the internal structure of layer n 1 (at least should not be aware).

Advantage of the hierarchy principle:

Clear structure Flexibility resulting from modularity

Thus: Internal change of layer n - 1 has no impact on layern(in general),

layers may be divided into sublayers,

layers may be omitted.

Disadvantage of the hierarchy principle:

a lot of overhead (each layer adds control information)


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The ISO Reference Model for Open Systems Interconnection

The International Standards Organization (ISO)

has standardized the ISO reference model for open systems as a firm basis for the

standardization of protocols:

Goals of the OSI model:

standardized nomenclature,

structuring,

framework for standards (of protocols).

Remarks:

The OSI model does not standardize protocols. Instead, it is a framework for thestandardization of protocols.

The OSI - model does not specific any implementation.

Both, the OSI structure and the protocols specified within this structure are independent fromany specific implementation.

Reference Model for Open Systems Interconnection (OSI)


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26HPN Fundamentals

The Seven Layers of the ISO/OSI Reference Model

1 Physical Layer

2 Data Link Layer

3 Network Layer

4 Transport Layer

Session Layer5

Presentation Layer6

Application Layer7

application oriented: layers 5 to 7

transport oriented: layers 3 and 4

technology oriented: layers 1 and 2

physical representation of 0 and 1(in addition: synchronization)

error protection, flow control(ensures secure transmission)

addressing, routing(How to find the destination host?)

reliable end-to-end connection(error recognition and correction)

structuring of information exchange(e.g. reset to well defined state after error)

adaptation of the information presentation(interpreter, consistent language inside the network)

functionality for special applications(e.g. file access)



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27HPN Fundamentals

1 Physical Layer

2 Data Link Layer(Leitungs- und Sicherungsebene)

The Layer Model of the IEEE LMSC (IEEE 802)

2 Data Link Layer

3 Network Layer

4 Transport Layer

Session Layer5

Presentation Layer6

Application Layer7

OSI-Modell:

Upper Layers

LAN/MAN Standards Committee der IEEE

(IEEE 802)

MAC

(Media Access Control)

LLC

(Logical Link Control)

How to control the access to the media?



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28HPN Fundamentals

The Layer Model of the IEEE LMSC (IEEE 802) (2)

1 Physical Layer

2 Data Link Layer(Leitungs- und Sicherungsebene)

3

4

5

6

7

Upper Layers

MAC

(Media Access Control)

LLC

(Logical Link Control)

2a

2b

3

4

5

6

7

Upper Layers

The LAN/MAN Standards Committee of the

IEEE (better known as IEEE 802)

standardizes protocols for

layer 1,

layer 2a and

layer 2b

of the OSI model.

Well known protocols are

Ethernet (IEEE 802.3),

Token Ring (IEEE 802.5)

Wireless LAN (IEEE 802.11) Wireless PAN, Bluetooth (IEEE 802.15)


Th L M d l f h IETF


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29HPN Fundamentals

1

2

1

2

Physical Layer

Data Link Layer

3 Network Layer

OSI model:Internet Engineering Task Force

Network

Technology

3 Internetwork

4 Transport Layer4 Transport

5

6

7

Session Layer

Presentation Layer

Application Layer

Application

The IETF standardizes protocols for

layer 3 and

layer 4

of the OSI model.

Well known protocols are

Internet Protocol (IP), Transmission Control Protocol (TCP) and

User Data Protocol (UDP)

In addition, the IETF works on applications

like e-mail, file transfer and remote login.

5

6

7

router

router

The Layer Model of the IETF


T


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Terms

(N)-layer

(N)-servicespecifies the service provided by the (N)-layer to higher layers

(N)-service providerabstract machine offering the (N)-service

(N)-service usera (N+1)-protocol entity, communicating with one or more (N+1) peer entity/entities using the

(N)-service (N)-service access point (SAP)

exchange point where (N)-service primitives are used

(N)-service primitive (N)-SPimplementation independent representation of an interaction between (N)-service

provider and (N)-service user

(N)-service data unit (SDU)(N)-SDUs are parameters of (N)-SPs, they carry information

The OSI model played a tremendous role in the specification of wording.

Important terms include:

T (2)


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Terms (2)

(N)-protocolcontrols the (indirect) communication of the (N)-peer entity

(N)-protocol entityindependent entity of the (N)-layer, is able to communicate with (N)-peer entities

(N)-peer entityentity of layer (N)

(N)-protocol data unit (PDU)consists of (N)-SDU and (N)-PCI, is sent from (N)-entity to (N)-peer entity

(N)-protocol control information (N)-PCIcontrol information (e.g. for error recognition and -correction), added by an (N)-entity to

an (N)-SDU.

(N 1) Service and (N 1) Protocol


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(N+1)-Service and (N+1)-Protocol

(N)-service

(N)-SAP

(N)-SPs;

where required

with (N)-SDUs

(N+1)-protocol

exchange of (N+1)-PDUs (virtual)

(N+1)-service user

(N+1)-

Protokoll-

-anz

(N)-SAP

(N)-SPs;

where required

with (N)-SDUs

(N+1)-

layer

(N+1)-SPs

(N+1)-service user

(N+1)-SPs

(N+1)-service

(N+1)-

protocol-

entity

(N+1)-

protocol-

entity

Simplified Model of a Communication System


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layer (N)

layer (N-1)

layer (N+1)

Simplified Model of a Communication System

(N+1)-PDU

(N)-SDU(N)-PCI

(N)-PDU

(N+1)-PDU

(N)-SDU(N)-PCI

(N)-PDU

Important note:

An implementation with queues between the layers is neither mandatory nor desirable.

The OSI model does not specify this kind of implementation detail.

Sending a (N+1) PDU to layer (N)


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layer (N)

layer (N-1)

layer (N+1)

Sending a (N+1)-PDU to layer (N)

(N+1)-PDU

Wait for

processing!

A PDU becomes a SDU


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A PDU becomes a SDU

(N+1)-PDU

Enter now!The (N+1)-PDU

becomes a

(N)-SDU

(N)-SDU

layer (N)

layer (N-1)

layer (N+1)

(N+1)-PDU

Layer (N) forms a (N)-PDU from a (N)-SDU


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Layer (N) forms a (N)-PDU from a (N)-SDU

(N)-PCI

(N)-PDU

Adding the

(N)-PCI

results in the

(N)-PDU.(N)-SDU

layer (N)

layer (N-1)

layer (N+1)

Transfer to Layer (N-1) and Transmission


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Transfer to Layer (N 1) and Transmission

(N)-PDU

(N)-PDU

(N)-PDU

layer (N)

layer (N-1)

layer (N+1)

The PDU

is forwarded

to

layer N-1

Forward to Layer (N) and Process


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Forward to Layer (N) and Process

(N)-PDU

(N)-PDU

(N)-PCI (N)-SDU

layer (N)

layer (N-1)

layer (N+1)

layer N

receives

the message.

Processing by Layer (N)


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layer (N+1)

Processing by Layer (N)

(N)-PCI (N)-SDU

Lets see whatis written here...

layer (N)

layer (N-1)

Forward to Layer (N+1)


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layer (N+1)

Forward to Layer (N+1)

(N)-SDU

layer (N)

layer (N-1)

(N+1)-PDU

Wrapping of Data


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pp g

audio

compact

disc

jewel box

storage casedisplay

packageshipping box

Each layer wraps data with an additional envelope (header and/or trailer), before

transferring the data to the lower layer:

For the protocol of layer (N) only (N)-PCI is relevant.

The protocol just works on the wrapping.

But: The size of a (N+1)-PDU has to obey certain rules.

(The postal service does not accept packets of 30 tons)

The transfer of the PDU has to be done in a proper manner.

Basic Principle Encapsulation


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p psource

applicationtransportnetwork

link

physical

HtHn M

segment Htdatagram

destination

applicationtransportnetwork

linkphysical

HtHnHl M

HtHn M

Ht MM

network

linkphysical

linkphysical

Ht

Hn

Hl

M

HtHn M

HtHn M

HtHnHl M

router

switch

message MHt M

Hn

frame


From Ethernet to HTML


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DA SA Type Data ChecksumPreamble SF

Ethernet Frame

Data

IP Datagram

Data

TCP Segment

Data

HTTP Slice

HTTP/1.0 200 OK\r\nServer: PAWS ElB-1.42\r\nContent-Type: text/html\r\nDate: Tue, 3

Network

Technology

Internetwork

Transport

Application

The TCP/IP-Internet


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44HPN Fundamentals

TCP/IP allows stations to communicate across totally different networks !


The Transmission Control Protocol (TCP)


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45HPN Fundamentals

Networks are unreliable.

End systems take care of error detection / correction.

The Internet Philosophy

The Transmission Control Protocol (TCP)

makes sure that all damaged and lost packets are retransmitted and thatduplicates are removed,

re-orders the messages at the receiver(ordering preservation), splits large information units into flows of small, numbered packets,

decides how fast packets are sent to the network, subject to capabilities ofthe sender, the receiver and the network in-between.


The Internet Protocol (IP)


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46HPN Fundamentals

The Internet Protocol

defines ................................

tries ..................

does not guarantee anything.

un-ambiguous, standardized, network-independent addresses,

to take the messages to the receiversomehow,

Net 1

IP

TCP

Application

Login,

File Transfer,

e-mail, ...

Net 1 Net 2

IP

TCP

Application

Net 1 Router

Net 2

IP

Identicalfor all

applications

Net 2

The Internet Protocol (IP)

specifies what all packets must look like to allow for routing(based on hierarchic addresses: network-ID, subnetwork-ID, ..., host-ID).


The Slim Waist


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47HPN Fundamentals

FTP mail remote login WWW access...

TCP UDP TP4...

IP

Ethernet Token Ring ISDN DSL Satellite

banking

Wireless LAN ...


Summary: Names, T-, N- and MAC-Addresses


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(in case of networks within the Internet)

Transport Protocol: TCP, UDP

(Layer 4)

Network Protocol: IP(Layer 3)

Medium Access, Network Technology(Layers 2 and 1)

Applications

File Transfer (ftp) WWW browser (http)

Names

Port Number

IP Addresses

MAC Addresses

Statically assigned

to applications

+ dynamically

assigned by the OS

DNS

ARP/

RARP

1.4 Delays, Losses, etc.


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www.n24.dephotothek.net

Analogy: Road Traffic

Delay Loss

www.colber-forster.de www.bmvbs.de

How do loss and delay occur?


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A

B

packet being transmitted (delay)

packets queueing (delay)

free (available) buffers: arriving packetsdropped (loss) if no free buffers

packets queue in router buffers packet arrival rate to link exceeds output link capacity

packets queue, wait for turn


Four sources of packet delay


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1. nodal processing

check bit errors

determine output link

2. queueing

time waiting at output link for transmission

depends on congestion level of router

3. transmission delay R=link datarate (bps), L=packet length (bits)

time to send bits into link = L/R

4. propagation delay

d = length of physical link, s = propagation speed in medium propagation delay = d/s

A

B

propagation

transmission

nodal

processing queueing



Nodal delay


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dproc = processing delay

typically a few microsecs or less

dqueue = queuing delay

depends on congestion

dtrans = transmission delay

= L/R, significant for low-speed links

dprop = propagation delay

a few microsecs to hundreds of msecs

A

B

propagation

transmission

nodal

processing queueing



proptransqueueprocnodalddddd

Queueing delay (revisited)


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R=link datarate (bps)

L=packet length (bits)

a=average packet arrival rate

traffic intensity = La/R

La/R ~ 0: average queueing delay small

La/R 1: delays become large

La/R > 1: more work arriving than can be serviced, average delay infinite!

Packet Loss


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queue (aka buffer) preceding link in buffer has finite capacity packet arriving to full queue dropped (aka lost) lost packet may be retransmitted by previous node, by source

end system, or not at all

A

B

packet being transmitted

packet arriving tofull buffer is lost

buffer

(waiting area)


1.5. Protocol Architecture in Real Life: Introducing WireShark


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WireShark is not the only open source tool available for protocol analysis:

tcpdump / windump (command line tool with textual output),

ngrep (filters and displays network connections),

netstat (command line tool to view open connections),

ettercap

A nice starting point for more tools is http://www.insecure.org/tools.html

WireShark is an open source (GNU GPL) tool for

software and protocol development,

troubleshooting,

analysis,

education,

(from http://www.wireshark.org/, accessed October 2007)

WireShark basically is of a successor of Ethereal!

Getting Connected WireShark Perspective


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ARP

TCP/HTTP

DNS

Packet List

Packet Details

Data on the wire

Traffic in a wireless campus network


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T.Henderson,D.Kotz,I.Ab

yzovThechangingusageofamature

cam

pus-widewirelessnetwork-ProceedingsMobiCom2004

Measurements in WiFi-network of Dartmouth Campus Hanover, USA (121 Access Points measured).

Inbound: Traffic sent by the AP to the card. Outbound: Traffic sent by the card to the AP.

Ports & TCP-Header information were analyzed

2. Application Layer in a Nutshell


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2.1 Conventional data communication

2.2 Multimedia communication

2.4. Skype

2.3 What is SIP?

2.1. Conventional data communication

"Cl i l" d t i ti lt i b t t ffi


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59HPN Fundamentals

"Classical" data communication results in bursty traffic.

In general, we find the following requirements:

Error free transmission (or error detection and correction),

As much bandwidth (throughput) as possible,

Delays as small as possible,

(maybe) support ofBroadcasting (One-to-all),

(maybe) support ofMulticasting (One-to-many),

(maybe) characteristics similar to LANs, e.g. connectionless communication(many applications were originally developed for LANs).


The Domain Name System (DNS)Instead of using numerical addresses in "Dotted Decimal Notation" a human user would


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prefer to have alphanumerical Internet names, e.g. "www.uni-bonn.de".

Therefore, we need a "DNS server" implemented as a distributed database which isable to map Internet names to Internet addresses.

DNS uses a hierarchical structure of names. This allows for a decentralized assignment of

names within the specific "Domain".

( nameless root)

jpus de

uni-bonn rwth-aachen dtag

informatik

va

reston

National

mil edu govcom org

sun

eng

yale

cs eng

IEEEACM

Generic

net int

The complete name of the domain results from the path up towards the root of the name tree.

Names can be mapped to IP addresses by asking the DNS server of the domain responsible.

Caching of name/address mappings increases the efficiency considerably.

Top-Level-Domains

But how long should those name/address mappings reside within the cache?

Example: Using Names when Surfing the Internet


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Name Server


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Name Space

The whole DNS name space is subdivided into so-called "zones". Each zone has one or

several name servers.

mil edu govcom org jpus de

uni-bonn rwth-aachen dtagva

reston

sun

eng

yale

cs eng

IEEEACM

net int

ai linda

robot

cnri

pharmazieinformatik

1 42 3 5 6

The decision whether (and how) to structure zones into sub zones is responsibility of the

"owner" of a specific zone.

This allows for high flexibility in the naming hierarchy.

Implementation of Name Resolution iterative


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A client's name resolvercontacts a name server.

This name serverpasses back a reference to the next responsible name server. The resolver contacts this server, ...

Source:A.S.Tanenbau

m,M.vanSteen:

DistributedSystems-PrinciplesandParadigms.

2ndEd.,Prentice-Hall,

2006

Implementation of Name Resolution recursive


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A client's name resolveronly contacts the next name server. Finding the responsible name server and thus the address now

is made by the involved name servers.

Recursive compared to iterative name resolution:Con: higher performance demands for name serverPro: caching is easier

may reduce communication costs at edge

Source:A.S.Tanenbau

m,M.vanSteen:

DistributedSystems-PrinciplesandParadigms.

2ndEd.,Prentice-Hall,2006

DNS Query


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Source:

BeckyGranger:Slides-DNSSECforthe.eduDomain,2010

IllustrationofNiranjanKunwar/Nirlog.com

2.2. Multimedia communication


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66HPN Fundamentals

In case of multimedia communication we often find

predictable load profiles,

elastic applications (e.g. coding subject to the current condition inside the network),

minimum throughput,

maximum delay,

maximum jitter ("Schwankungen der Verzgerung"),

maximum message loss rate(In case of appropriate coding a certain loss rate is acceptable).


Specification of requirementsThe application can specify the acceptable delay in different ways. In general, the

specification refers to


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67HPN Fundamentals

specification refers to

information units at application layer (pictures, samples, ...),

the local clock.

i,DD maxi

minmaxi ZDDProb

i,JD-D=J maxii

minmaxi UJJProb

Deterministic delay limits:

Di the delay of message number i,Dmax the upper limit selected by the application

Statistical delay limit:

Di, Dmax as above. Zmin is the lower probability bound

for successful and in-time reception of message number i.

Deterministic jitter limit:

Di as above. D is the "perfect" delay.

Ji is the jitter of message number i,

Jmax

the upper jitter limit selected by the application.

Statistical jitter limit:

Ji, Jmax as above. Umin is the lower probability bound

for successful reception within the specified

jitter limits.


Total delay in case of "multimedia"


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68HPN Fundamentals

Digitization

Pixels

Coding

Macro blocks

Packetization

Transmission buffer

Network access

Network

receive packet, de-packetize

check delay

buffer or drop

Playback bufferDe-coding

Pixels

Macro blocks

Packets

Sender

Receive

r

Ddig

Dencode

Dpack

Dnetwork

Dplayback

Ddecode

Ddisplay

Receive buffer Packets


VoIP QoS

7

ITU G.114

Utility

Packet loss (%) VoIP Issues Solutions


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69HPN Fundamentals

1 2 3 4 5 6 1 3 5 4 6

Packet Loss

Inversion

Jitter

IPNe

twork

Network Delay

SenderNetwork

Receiver

Sender Delay:

Coding delay

Packeting delay

Transmission delay

Receiver Delay:

Decoding delay

DePacketing delay

Receiver delay

100 200 300 400 500

0

4

3

5

2

1

7

6

OperationalTarget for

Voice

Possibly

Tolerable forVoice

Unacceptable for

Voice or Fax

yRecommendation

Delay (ms)

Packetizing delay Small packets for VoIP

Serial delay Priorities and jitter buffers

High bit-rate video Video compression

Constant-bit-rate voiceSilence suppression and comfort

noise

Resend due to errors Real-time Transport Protocol


2.3. What is SIP?

SIP* is an application layer control (signaling) protocol for creating


70/213

70HPN Fundamentals

SIP is an application layer control (signaling) protocol for creating,

modifying and terminating multimedia sessions with one or moreparticipants.

* SIP is the Session Initiation Protocol, specified in RFC 3261

What Can You Do With SIP?

SIP is a peer-to-peer protocol where end-devices initiate sessions

SIP sessions include Internet Multimedia conferences, Internet telephone calls, and multimediadistribution

SIP is suitable for applications having a notion of session, e.g. network games, video conferences

SIP is designed for scalability, simplicity, mobility, and service creation

SIP is text-based for easyimplementation and debugging

SIP is a simple, extensible protocol

SIP is text-based for easyimplementation and debugging

SIP is a simple, extensible protocol


Setting up a call to a known IP address

Ali SIP i itAliceBob


71/213

71HPN Fundamentals

Alices SIP invite message

indicates her port number, IPaddress, encoding she prefers to

receive (PCM ulaw)

Bobs 200 OK message indicates

his port number, IP address,preferred encoding (GSM)

SIP messages can be sent over

TCP or UDP; here sent over

RTP/UDP.

default SIP port number is 5060.

time time

Bob's

terminal rings

Alice

167.180.112.24 193.64.210.89

port5060

port 38060

Law audio

GSMport 48753

[email protected]=INIP4167.180.112.24m=audio38060RTP/AVP0 port5060

200OK

c=INIP4193.64.210.89

m=audio48753RTP/AVP

3

ACKport5060




2.4. Skype


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Skype Details ?Research?


73/213


Skype Details

uses a proprietary solution (no SIP, H323, )

difficult to reverse engineer due to extensive use of both cryptography and


74/213


Source:TrackingdownSkypetraffic

byDarioBonfiglio,MarcoMellia,

MichelaMeo,NicoloRitaccaand

DarioRossi[INF

OCOM'08]

difficult to reverse engineer due to extensive use of both cryptography and

obfuscation techniques

may rely on eitherTCP or UDP at the transport layer

both signaling and communication data are preferentially carried over UDP

a single random port is selected during application installation

never changed (unless forced by the user)

when a UDP communication is impossible, Skype falls back to TCP

listening to the same random port whenever possible, or using port 80 and 443

can select between different Codecs according to an unknown algorithm

Skype Adaptive Voice Coding

Mellia,


75/213


Source:TrackingdownSkypetrafficb

yDarioBonfiglio,MarcoM

MichelaMeo,N

icoloRitaccaandDario

Rossi[INFOCOM'08]

Average Bitrate (B): the average amount of bits generated at application layer in atime interval of 1 second.

Inter-Packet-Gap (IPG): the time elapsed between two consecutive packetsbelonging to the same flow.

Payload length (L): the number of bytes carried by TCP or UDP.

Skype Features

http://www.blackhat.com/presentations/bh-europe-06/bh-eu-06-biondi/bh-eu-06-biondi-up.pdf


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3. Transport Layer in a Nutshell


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3.1 Transport services and protocols

3.2 TCP - Error Control

3.3 TCP - Connection Management

3.4 TCP - Retransmission Timer

3.5 TCP - Flow Control and Congestion Control

3.1 Transport services and protocols

g:ATop

009.


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78HPN Fundamentals

provide logical communicationbetween app processes running

on different hosts

transport protocols run in end

systems sender side: breaks app

messages into segments,passes to network layer

receiver side: reassemblessegments into messages,

passes to app layer

more than one transport protocol

available to apps Internet: TCP and UDP

application

transport

network

data link

physical

application

transport

network

data link

physical

logicalend-endtransport

Source:JimKurose,Keith

Ross:ComputerNetworking

DownApproach-5thedition,Addison-Wesley,April

20


Internet transport-layer protocols

reliable in-order delivery

Top

.


79/213

79HPN Fundamentals

reliable, in order delivery

(TCP)

congestion control

flow control

connection setup unreliable, unordered delivery:

UDP

no-frills extension of best-

effort IP services not available:

delay guarantees

bandwidth guarantees

application

transport

network

data link

physicalnetwork

data link

physical

network

data link

physical

network

data link

physical

network

data link

physical

network

data link

physical

networkdata link

physical

application

transport

network

data link

physical

logicalend-endtrans

port Sou

rce:JimKurose,KeithRoss:ComputerNetworking:

A

Dow

nApproach-5thedition,Addison-Wesley,April200

9.


Internet transport protocols services

TCP service:

connection-oriented:setup required between

UDP service:

unreliabledata transfer between


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80HPN Fundamentals

co ect o o e ted p q

client and server processes

reliable transportbetween sending andreceiving process

flow control:sender wont overwhelm receiver

congestion control:throttle sender whennetwork overloaded

does not provide:timing, minimumthroughput guarantees, security

u e ab e

sending and receiving process

does not provide: connection

setup, reliability, flow control,

congestion control, timing,

throughput guarantee, or security

Application Application layerprotocol Underlying transport protocol

e-mail SMTP [RFC 2821] TCP

remote terminal access Telnet [RFC 854] TCP

Web HTTP [RFC 2616] TCP

file transfer FTP [RFC 959] TCP

streaming multimediaHTTP (eg Youtube),

RTP [RFC 1889]TCP or UDP

Internet telephonySIP, RTP,

proprietary (e.g., Skype)typically UDP


Transport service requirements of common apps

Data loss

some apps (e.g., audio) can tolerate some loss


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Application Data loss Throughput Time Sensitive

file transfer no loss elastic no

e-mail no loss elastic no

Web documents no loss elastic no

real-time audio/video loss-tolerant audio: 5kbps-1Mbps;video:10kbps-5Mbps

yes, 100s msec

stored audio/video loss-tolerant same as above yes, few secs

interactive games loss-tolerant few kbps up yes, 100s msec

instant messaging no loss elastic yes and no

other apps (e.g., file transfer, telnet) require 100% reliable data transfer

Throughput

some apps (e.g., multimedia) require minimum amount of throughput to be effective

other apps (elastic apps) make use of whatever throughput they get

Timing some apps (e.g., Internet telephony, interactive games) require low delay to be effective

Security

Encryption, data integrity,

UDP

Why is there a UDP?UDP: User Datagram Protocol [RFC 768]

0 8 16 24 31


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82HPN Fundamentals

no frills, bare bones Internet

transport protocol

best effort service, UDP segmentsmay be:

lost

delivered out of order to app

connectionless:

no handshaking between UDPsender, receiver

each UDP segment handled

independently of others

no connection establishment(which can add delay)

simple: no connection state at

sender, receiver

small segment header

no congestion control: UDP canblast away as fast as desired

ChecksumDatagram Length

Data

....

Source Port Destination Port

often used for streaming multimedia

apps

loss tolerant rate sensitive

other UDP uses

DNS

SNMP

reliable transfer over UDP: add

reliability at application layer

application-specific error

recovery!


UDP PDUs

The service provided by the User Data Protocol (UDP) is a best effort service. UDP is used forconnectionless data transmission. The PDUs have the following structure:


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ChecksumDatagram Length

Data....


0 8 16 24 31

Source Port (16 bit)(Optional) Identification of sender process for mapping of replies.

Destination Port (16 bit)Identification of destination process.

Datagram Length (16 bit)Total TPDU length (in byte) incl. UDP overhead.

Checksum (16 bit)(Optional) Error detection; mandatory for UDP with IPv6.

Note:Without UDP checksum, there is no error detection for the data field: IPv4 error control

is limited to the IP header.

TCP PDUs


0 8 16 24 31


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Sequence Number

Acknowledgement Number

HLEN Reserved Code bits Window

Urgent PointerChecksum

Options (if any) Padding

Data

....

Source Port, Destination Port (16 bit each)Addresses of specific processes; also used for connection identification.

Sequence Number (32 bit)Position of a data segment within the byte stream (for window mechanism).

Acknowledgement Number (32 bit)Number of the next expected byte in the opposite direction.Cumulative acknowledgement; for this reason: robust against loss of ACKs.

HLEN (4 bit)Header length (in multiples of 32 bit); also: Offset for the data field in the TPDU.

The Transmission Control Protocol(TCP) makes communication reliable.

TCP adds to the IP address a

16 bit TSAP address.With TCP, a TSAP is called Port.

TCP PDUs (2)

Reserved (6 bit)Reserved for future use.


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Code Bits (6 bit)Purpose and contents of the TPDU. From left to right:

URG Urgent pointer field is valid

ACK Acknowledgement field is valid; flag reset in the connection setup packet

PSH This segment requests a push (Immediate delivery to the receiver)

RST Reset the connection

SYN Synchronize sequence numbers (Used during connection establishment)

FIN End of data stream

Window (16 bit)

The receiver controls the transmission window size, cf. chapter on flow control.

Checksum (16 bit)Checksum for the whole TPDU and a pseudo header which includes the IPaddresses of sender and receiver.

Urgent Pointer (16 bit)Marks the end of urgent data included in the data stream.

OptionsTCP specifies additional options, e.g. window scaling (multiply the window value by 2n; n

between 0 and 14).

TCP Message Format

TCP provides a connection-oriented, reliable, byte-stream service


86/213

86HPN Fundamentals

TCP Packet

which cares for network resources

Sequence Number


HLEN Reserved Code bits Window

Urgent PointerChecksum

Options (if any) Padding

Data

....


0 8 16 24 31


TCP Connections


hi h f k


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87HPN Fundamentals


TCP connects applications across networks

Multiple applications on the same host are distinguished by theirports Unlike UDP, a TCP port is not simply a queue!

TCP connections are full-duplex

TCP uses the connection, not the protocol port, as its fundamental abstraction.Connections are identified by a pair of endpoint identifiers.


TCP Connections (2)


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88HPN Fundamentals

TCP Packet

0 8 16 24 31

A TCP endpoint is a pair of

integers

(host id, port number)


10.1.5.3 10.5.2.3

A given TCP port number can be shared by multiple connections.


TCP Byte Stream


hich cares for net ork reso rces


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89HPN Fundamentals


TCP is byte-oriented, not packet oriented

TCP streams are unstructured The application has (nearly) no means to control the TCP flow

TCP transmits data in units called segments

a segment may be as small as 1 byte!

typical segment size is 1024 (plus header overhead)


Reliable Stream Transport Service




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90HPN Fundamentals


TCP is reliable:

TCP delivers data correctly or not at all

TCP delivers data completely

TCP eliminates duplicates

TCP delivers data in the correct order


3.2. TCP Error Control

Potential errors:

Corruption of bits

Potential errors:

Corruption of bits

Detection:

Checksum

Detection:

Checksum


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91HPN Fundamentals

TCP Packet


Corruption of bits

Loss of entire messages

Duplicates

Packet misordering

Huge delays

Corruption of bits

Loss of entire messages

Duplicates

Packet misordering

Huge delays

Checksum

missing acknowledgement

sequence number

sequence number

congestion control

Checksum

missing acknowledgement

sequence number

sequence number

congestion control

Sequence Number


Checksum


0 8 16 24 31

A

Automatic Repeat Request


92/213


Idea: After sending a message, the sender waits for a positive acknowledgement.

If the acknowledgement does not arrive before a timer expires,

the message is repeated and

the senderwaits for a positive acknowledgement again.

This strategy is called stop-and-wait.

The most important strategy for the detection (and correction) of the loss of entire

messages is called ARQ (Automatic Repeat reQuest).

A new message is only transmitted after the acknowledgement for the previous

message has been received by the sender.

The Alternating Bit Protocol

Sender Receiver Sender Receiver

The alternating bit protocol is a straightforward way of realizing stop-and-wait:


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DT.1

ACK.1 successfultransmission

successfultransmission

data lost

DT.0

ACK.0

DT.1

DT.1ACK.1

DT.0

retransmissionafter timeout; correctacknowledgement

Business as

usual

Sender Receiver

DT.0

ACK.0acknowledgementlost

retransmissionafter timeout; receiverignores duplicate butsends ACK

DT.0

ACK.0

DT.1

ACK.1

Sender Receiver

successfultransmission

Obviously, message numbering is based on 1 bit only.

In real life, this approach is used in short-range networks such as Bluetooth.

When sending data in both directions, the acknowledgement may be carriedtogether with the data in the opposite direction (piggybacking).

Sliding Windows

Stop-and-Wait is inefficient with short messages and/or large signal

propagation delay:

S d R i


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DT.1

ACK.1Receive + check message 1,send ACK

DT.0

ACK.0

Sender Receiver

Receive + check message 2,send ACK

Send message 1

Send message 2

DT.1

ACK.1

Receive + check message 3,

send ACK

Send message 3

Wait

Wait

Wait

Efficiency may be improved by allowing the sender to transmit several PDUs

before stopping and waiting for ACKs.The mechanism used in this case is called

Sliding Window Protocol

Sender Window and Receiver Window

Sender and receiver negotiate a window size W before starting transmission:

1


95/213


1 W m, with m 2 modulus

The sender window:

tells which PDUs may currently be sent,

changes (slides upwards) when receiving correct ACKs.

0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7

161 2 3 4 5 6 7 8 9 101112131415Overall number of the message:

Sequence number of the PDU: 0 1 2

191718

The receiver window:

tells which PDUs are currently accepted at the destination,

changes (slides upwards) when receiving correct (new) PDUs

The PDUs are numbered modulo m (using a n-bit field).

Example:

Example: sender window size = 3; receiver window size = 1

Start:

0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7Sequence no of PDU: 0 1 2

0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7Sender window: 0 1 2


96/213


0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7Sender window: 0 1 2

0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7Receiver window: 0 1 2


Sender window :

Receiver window :

0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2

0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2


0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7Sender window : 0 1 2

0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7Receiver window : 0 1 21


0 2 3 4 5 6 7 0 1 2 3 4 5 6 7Sender window : 0 1 2

0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7Receiver window : 0 1 21

1

Transmission of PDUs 0, 1 and 2: Both windows remain unchanged

PDU 0 received: Change at receiver window

ACK for PDU 0 received: Change at sender window

2 31

0 0 30 0

Strategies at the Receiver

With sliding window protocols, the receiver has several options of how to react to PDU

loss:

Ask for a retransmission of those PDUs which were lost (specific selection)


97/213


et a s ss o Us c e e ost ( p )

Ask forretransmission ofall PDUs beginning with the first one which was lost

Hybrid solutions of both options

For the retransmission of specific PDUs, negative acknowledgements (NACK) may

be used: These carry the sequence number(s) of PDUs to be retransmitted.

Go back n

All PDUs beginning with the first one lost are retransmitted.(Receiver window with size 1)

Pro: - Simple implementation of the protocol

Con: - Channel capacity is wasted by retransmitting PDUs correctly deliveredto the destination.

Remark: The receiver only accepts PDUs in the correct order.

Selective Repeat and Selective Reject

Selective Repeat

The receiver

Selective Reject

Retransmit lost PDUs only


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The receiver

buffers all PDUs received correctly

acknowledges the sequence ofpackets received without a gap

Pro: - Improved efficiency

Con: - More complex implementation

(out-of-order packets bufferedat destination)

Retransmit lost PDUs only(requested by NACKs).

The retransmission is initiated by some

kind of NACK.

Pro: - Excellent efficiency

Con: - No gain when loosing sequence

of PDUs (several or even manyPDUs have to be retransmitted anyway)

Question: What is the impact of the receiver strategy on the size of the sender buffer?

Comparison of Receiver Strategies

1

32

1

23al

1

23al

D t

Sender Receiver Sender Receiver Sender Receiver


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34567

89

101123

456789

10

1

--

------

-23456

78910

34567

892

101112

131415161718

19

1

34

567892

101112131415

161718

19

Tim

eoutInterva 3

4567

89

101123

45678

12

13

1

34

5678910

112----

--1213

Tim

eoutInterva

TimeoutInterval

Data:

ACK:

NACK:ACK

1

ACK

2

ACK

3 ACK

11

ACK

1

ACK

1

NACK2

ACK

9

Go back n Selective Repeat Selective Reject

Sliding Window with TCP

The sliding window scheme of TCP works byte by byte.Therefore, 3 pointers are in use:

Left border of the sending window


100/213


Left border of the sending window(border between ACKnowledged and not yet ACKnowledged data)

Right border of the sending window(all bytes up to this border may be sent without waiting for any ACKs)

Marker of current sending position(border between data already sent and data waiting to be sent)

1 2 3 4 5 6 7 8 9 10 11 12 13 ...

current sender window

All bytes waiting to be sent are being sent as soon as possible. Therefore, the

marker of the sending position usually moves pretty fast towards the right border.

TCP ACKs indicate which byte is expected next by the receiver:All bytes up to the one given in the ACK are acknowledged.

3.3. TCP Connection Management

For connection establishment, TCP uses the concept of a Three Way Handshake.The following bit flags of the TCP header are important:

SYN bit (synchronize sequence numbers):


101/213


SYN bit (synchronize sequence numbers):Is set to 1, if the sequence number of the TCP PDU carries its initial value. The SYNPDU is the very first PDU of a TCP connection.

ACK bit (acknowledgement):Is set to 1, if the PDU carries a valid acknowledgement number.

A sends SYN seq=xB receives SYN

B sends SYN seq=y, ACK x+1

Host A Host B

A receives SYN+ACK

A sends ACK y+1B receives ACK

(Application needs transport connection)

Server allocates resourcesfor connection management.

SYN-Flooding - an Denial-of-Service-Attack

Client Server


102/213


send SYNServer receives SYN

Server sends SYN+ACK

send SYN Server receives SYNServer sends SYN+ACK

send SYNServer receives SYN

Server sends SYN+ACKsend SYN

Server receives SYN


Server receives SYN


Server receives SYN

Server sends SYN+ACK

Connection Release with TCP

TCP establishes full duplex connections. Therefore, both end points have to agree

on the release.

Connection release is based on a modified Three Way Handshake.


103/213


Anotherbit flag of the TCP header is used for connection release:

FIN bit:

Is set to 1, if the sender has no more data to transmit.

The first FIN-PDU will be issued when the senderhas send all of its data andadditionally has received all outstanding ACKnowledgements.

A sends FIN seq=xB receives FIN

B sends ACK x+1

Host A Host B

A receives ACK

A sends ACK y+1B receives ACK

(Application wants to close the connection)

(B informs its application)

(Application closes connection)B sends FIN seq = y; ACK x+1A receives FIN+ACK

Simplified State Diagram of TCP Connection Management

closed

begin

anything / reset

Terminology:

Input/ Output


104/213


SYN

sent

estab-lished close

wait

last

ACK

FIN

wait-1closing

FIN

wait-2

timed

wait

passive open /close

active open /SYN

close / timeout /reset

send /SYN

SYN / SYN+ACK

reset

SYN / SYN+ACK

close/FIN

ACK /

close/ FINFIN / ACK

SYN+ACK /ACK

close/ FIN

ACK

/ACK/

FIN /ACK

ACK/

FIN /ACK

FIN+ACK/ACK Timeout after 2 lifetimes of a segment

SYN

receivd

listen

3.4. TCP Retransmission TimerIn the Internet, we observe

heavy variations of theRound Trip Time

(= time from sending a PDU toreceiving the corresponding

190

200

210

220

s]


105/213

105HPN FundamentalsCopyright 2011/2012 Computer Science 4, University of Bonn 17.10.2011

receiving the corresponding

ACK).

The figure shows an example.

In 1988, Van Jacobson proposed an RTT estimation algorithm for TCP which uses anaging function:

1. For each PDU, determine the point in time of sending the PDU

2. For each PDU determine the point in time of receiving the corresponding ACK

3. Calculate the difference of both

4. Update a weighted average value

The choice of (0<


106/213


ACK for original PDU or

ACK for retransmission/duplicate ?

4567

8910112

1

--

------

-2

Tim

eoutInterv

ACK

1

ACK

2

Go back n

ACK2

? RTT ?

RTT ?

Karns Algorithm and Timer Backoff

ti tti t

a) Ignore RTT measurements in case of timeout and PDU retransmission.

b) Instead, use a timer backoff strategy:


107/213


timeouttimeoutnew

A typical value is= 2.

This algorithm originally proposed by Phil Karn (included in almost all implementations ofTCP)

eliminates the ambiguity of ACKs, enlarges the timeout interval in case of high network load,

in critical situations decouples the calculation of timeout intervals fromRTT measurements.

The timeout interval grows until a successful transmission (without retransmissions)happens again.

After successful transmission, the timeout interval estimation is based on RTT

measurements as discussed before.

Timer Control according to Van Jacobson

The Timer control algorithm originally proposed by Van Jacobson is presented by

William Stallings** as follows:

aging functionSRTT(k+1) = (1-g) x SRTT (k) + g x RTT (k+1)


108/213


SERR (k+1) = RTT(k+1) SRTT(k)

SDEV (k+1) = (1-h) x SDEV (k) + h x SERR (k+1)

RTO (k+1) = SRTT (k+1) + f x SDEV (k+1)

*Van Jacobson, Michael J. Karels, Congestion Avoidance and Control, Proceedings of SIGCOM

Almost identically available at ftp.ee.lbl.gov/papers/congavoid.ps.Z

**W. Stallings, Data & Computer Communications, 6th Ed., Prentice Hall, 2000

aging function

RTT round trip time SRTT smoothed round trip time

SERR smoothed error SDEV smoothed mean deviation

RTO retransmission timeout

Van Jacobson proposed for the constants: g = 1/8; h =1/4; f = 2

Later (1990) he changed his recommendation to: g = 1/8; h =1/4; f = 4

3.5. TCP Flow Control and Congestion Control

(end-to-end) Flow control

protects the receiver from being overloaded by the sender.

Obviously the receiver cannot be overloaded in case of stop and wait: A new


109/213


Congestion control(Internet flow control)

protects the network(s) from being congested.

The senders reduce their load in critical situations.

Obviously, the receiver cannot be overloaded in case of stop-and-wait: A newmessage may only be transmitted after receiving the (positive) ACK.

ACK based window mechanisms provide flow control in a similar way: The

transmission window may only be moved after receiving ACKs.This effect is not sufficient because ofbuffer overflow and retransmissionmechanisms.

Flow control by acknowledgements

Wi h i i k l d

Acknowledgement based error detection and correction with appropriately

chosen window sizes results in flow control.


110/213


With increasing network load,

data arrive later at the receiver.

With increasing receiver load,

data received wait longer for processing (by the receiver).

In both cases,

acknowledgements are sent later,

acknowledgements arrive later at the sender,

finally, the sender must wait for acknowledgements before resuming

transmission, [ there is a risk of timeouts and (unnecessary) duplicate transmissions ].

A situation where the sender is completely controlled by the incoming ACK stream

is called ACK Clocking.

TCP Window Advertisement

The TCP header includes a window field (Window Advertisement) telling thebuffer space available at the sender of this packet.


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TCP Packet


0 8 16 24 31

SA F

Sequence Number


Checksum

Window

TCP Window Advertisement

Sender Receiver1000

The TCP header includes a window field (Window Advertisement) telling thebuffer space available at the sender of this packet.

Note:

The receiver must


112/213


Ack1001,Window200

Send

Receive

Send

Wait

Wait

ACK and

new window

Ack1,Window1000

Ack1201,Window0

Receive

ACK and

new window

The receiver must

not revoketransmission

permissionsalready granted:

The window may

only be reduced

after receiving

(new) data.

Remark:

The strategy shown here

may also be used at OSIlayer 2.

1000bytes

200bytes

TCP window control mechanisms (simplified)

TCP sender windowT itt d

The following figure illustrates the TCP window control mechanisms.


113/213

113HPN Fundamentals

Data to transmit

Not yet transmitted.May only be transmitted

after the window has

opened again.


Not yet transmitted,but ready for immediate

transmission.

Transmittedbut not yet

acknowledged.

Retransmission may be

necessary.

TCP sender window

Minimum of

sender buffer size and

receiver capabilities (advertised window)

Transmittedand

already

acknowledged

Border moves right

when receiving

ACKs

Border moves right

when transmitting

data.

Border moves right

if and only if the

receiver tells thesender to do so.

TCP in case of high load and overload

In the Internet,

more than 80 % of the total load is TCP traffic,

network collapse is avoided by the cooperative behaviour of TCP.


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TCP tries to reach both maximum efficiency and maximum predictability: a trade-off!

Maximumnetwork utilization

(max. efficiency)

Predictablenetwork behaviour

(in particular

for real time traffic)

TCP supports both interactive application and bulk transfer:

Interactive applications: usually small amount of data

typical examples: telnet, ssh

Bulk transfer of data: usually large amount of data

typical examples: ftp, email, http

TCP in case of bulk data transfer

For bulk data transfer, TCP tries to achieve

maximum throughput with minimum packet loss.


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Slow Start

Fast Retransmit and Fast Recovery

Congestion Avoidance

p

With adaptive load control TCP achieves efficient resource utilization in todays Internet.

The throughput achievable per data stream usually cannot be predicted.

Inventions

by Van Jacobson

TCP applies mainly three techniques to achieve these goals:

Slow Start

The slow start and congestion avoidance algorithms MUST be used by a TCP

sender to control the amount of outstanding data being injected into the network.

RFC 2581, TCP Congestion Control, April 1999, p.3

The slow start and congestion avoidance algorithms MUST be used by a TCPsender to control the amount of outstanding data being injected into the network.

RFC 2581 TCP Congestion Control April 1999 p 3


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RFC 2581, TCP Congestion Control , April 1999, p.3RFC 2581, TCP Congestion Control , April 1999, p.3

Slow Start makes TCP discover the load situation on the path from the source

to the destination when starting data transfer and

after packet loss with Retransmission Timer Timeout.

Idea: Limit the transmission window by a Congestion WindowIdea: Limit the transmission window by a Congestion Window

CONGESTION WINDOW (cwnd):A TCP state variable that limits the amount of data a TCP can send.

At any given time, a TCP MUST NOT send data with a sequence number higher than

the sum of the highest acknowledged sequence number and the minimum of cwnd andrwnd*.


*RECEIVER WINDOW (rwnd): The most recently advertised receiver window.

TCP window control mechanisms

TCP sender window

Minimum of sender buffer size and

i biliti ( d ti d i d )

Transmittedand

already


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117HPN Fundamentals

Data to transmit

Not yet transmitted.May only be transmitted

after the window has

opened again.


Not yet transmitted,but ready for immediate

transmission.

Transmittedbut not yet acknowledged.

Retransmission may be

necessary.

receiver capabilities (advertised window)

current congestion window

acknowledged

Border moves right

when receiving ACKs

Border moves

right when

transmitting data.

Border moves right

if and only if the

receiver tells the

sender to do so*.

*if the congestion window is not exceeded

Initial cwnd size: maximum message size (new: 2 x max. message size)

When to increment cwnd: cf. next slide

How slow is slow start ?

The congestion window size is influenced by

the size of transmitted messages and by

the Round Trip Time:


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Initial cwnd size: max. one segment (max. SMSS); cf. RFC 2001, 1997max. 2 segments (max. 2xSMSS); cf. RFC 2581, 1999

How to increment cwnd: One segment per correctly received ACK.

SENDER MAXIMUM SEGMENT SIZE (SMSS):

The SMSS is the size of the largest segment that the sender can transmit.

This value can be based on the maximum transmission unit of the network, the path MTU discovery algorithm, Receiver Maximum Segment Size, or other factors.

The size does not include the TCP/IP headers and options.


Example: Congestion window size

The graph shows how the congestion window is incremented for each ACK

received.

Finally, the tx_window and the transmission data rate become too large:


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0

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Segments transmitted

Congestion

wi

ndows

ize

The resulting timeout yields

packet retransmission (duplicate),

change to Timer Backoff and

congestion window reduction to 1 segment.

Packet loss due to buffer overflow.

Linear or exponential growth?

The transmission window only seems to grow slowly.

The larger the tx_window, the larger the number of ACKs.


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After log2 N Round Trip Times, TCP may already transmit N segments

RoundTrip

Time

With maximum speed into the traffic jam

Slow Start obviously makes TCP

very quickly reach the available bandwidth and

very quickly go beyond this to overload.

... upon a timeout cwnd MUST be set to no more than ... 1 full-sized segment


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Slow Start


CongestionAvoidance

cwnd < ssthresh ?Yes No

When a TCP sender detects segment loss using the retransmission timer, the value of ssthreshMUST be set to no more than the value ...

ssthresh = max (FlightSize / 2, 2*SMSS) .. [ssthresh = slow start threshold]

..., FlightSize is the amount of outstanding data in the network.

RFC 2581, TCP Congestion Control, April 1999

After this, the value ssthresh (any value for connection setup) determines the upperlimit for staying with slow start:

RFC 2581, TCP Congestion Control, April 1999

Introduction to Time Sequence Plots

Time Sequence plots are a common means of logging (and observing) TCP behavior.

Each mark resembles a packet receive or a packet transmission event. The X-axis denotes the time at which an event occurs

The Y-axis denoted the sequence number of the packet


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0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2

S

egmentnumber

Time [s]

Segment transmittedAck received

In the following

a diamond ( ) denotes a data segment transmission event

a crosshair ( ) denotes an acknowledgement segment reception event

At second 1.0, the sender

The following plot derives from the observation of a TCP sender:

receives an acknowledgement for

segment 10

transmits segment 20

Introduction to Time Sequence Plots (2)

Time Sequence plots convey more information than that.

In the following we assume that the sender is saturated (i.e. the senders application hasalways sufficient data to transmit)


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4 4.5 5 5.5 6

Segmentnumber

Time s


Observable facts from the plot:

The maximum transmissionwindow size Wis 10 segments. Itcorresponds to the y-offset of the

dot trails.

The round trip time is one

second. It corresponds to the x-offset of the dot trails.

The transmission rate W/ RTT is10 segments per second. It

corresponds to the slope of the dottrails.

W

RTT

W / RTT

TCP Slow Start (Time Sequence Plot)

The TCP Slow-start algorithm is applied on connection establishment, or

on reestablishment after a retransmission timeout

Fundamental question on connection startup:

Which congestion window size to begin with?


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The slow-start algorithm results in

exponential growth of the congestionwindow over time.

Q: How long can this continue?

g g

TCP solution: start with minimum congestion window size and increase cwnd by one

segment for each acknowledgement received.

Slow-start example:

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0 0.2 0.4 0.6 0.8 1 1.2

Sequencenumber

Time [s]


TCP Slow Start (2)

Q: How long can slow-starts exponential growth continue?

A: Until packet loss is detected, or the congestion window grows too large.

TCP maintains a state variable, the slow-start threshold (ssthresh) that determineswhether the TCP sender is in slow-start or congestion avoidance.


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For

congestion window size < ssthresh: TCP is in slow-start

congestion window size >= ssthresh: TCP is in congestion avoidance

The task ofslow-start is to quickly ramp up thecongestion window to ssthresh. The ssthresh is

regarded as an estimate for the proper magnitude ofthe congestion window.

The task ofcongestion avoidance is tocontinuously probe if the network can handle a

higher congestion window.

RTT

cwnd

ssthresh

Congestion

avoidance

Slow-start

Congestion Avoidance

Idea:

The congestion only grows linearly over time (instead of exponentially)

Current versions of TCP carefully discover the bottleneck capacity.


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The congestion only grows linearly over time (instead of exponentially).

One formula commonly used to update cwnd during congestion avoidance is ...:

cwnd += SMSS*SMSS/cwnd (2)

This adjustment is executed on every incoming non-duplicate ACK.

Equation (2) provides an acceptable approximation to the underlying principle of increasing

cwnd by 1 full-sized segment per RTT.

During congestion avoidance, cwnd is incremented by 1 full-sized segment perround-trip time (RTT). Congestion avoidance continues until congestion is detected.

RFC 2581, TCP Congestion Control, April 1999, p. 4

Congestion window during Congestion Avoidance

100

Over time, the congestion window size

initially grows exponentially (Slow Start)

then grows close to linearly (Congestion Avoidance)


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0

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Time [Round Trip Times]

cwdsize

Note:

TCP believes all kinds ofpacket loss to be due to network congestion.

Consequently, TCP considerably reduces the load.

TCPs fundamental assumption is not necessarily true for wireless networks.

TCP Congestion Avoidance (Time Sequence Plot)

In congestion avoidance, the congestion window grows, but at a slower rate than in slow

start.

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