Computer Networks – A Top Down Approach: Notes

(Chapters 1, 2, 3, 9)

Chapter 1 – Computer Networks and the Internet

1.1 – What is the Internet?

•
• 1.1.1 A Nuts-And-Bolts Description
 o End systems/hosts connected by a network of communication links
and packet switches
Hardware and software components
o End systems/hosts access the Internet through lower-tier Internet
Service Providers (ISPs), which are interconnected through national
and international upper-tier ISPs
o Protocols are run to control the sending and receiving of information
TCP/IP
o Internet standards developed by the Internet Engineering Task Force
(IETF), whose standard documents are requests for comments (RFCs)
• 1.1.2 A Services Description
o Infrastructure that provides services to distributed applications
Distributed since they run on multiple end systems
o End systems and the Internet provide a socket interface for
communication
• 1.1.3 What Is a Protocol?
o Defines the format and order of messages exchanged between two
or more communicating entities, as well as the actions taken on the
transmission and/or receipt of a message or other event

1.2 – The Network Edge

• Internet of Things (IoT) – concept of loads of things (besides traditional
computers) being connected to the Internet (watches, glasses, thermostats)
• Hosts
o Host application programs
o Clients or servers
• 1.2.1 Access Networks
o Digital Subscriber Line (DSL)

Often from same local telephone company (telco) that

provides local phone access
Exchange data with a digital subscriber line access multiplexer
(DSLAM) located in the telco’s local central office (CO)
DSL modem translates digital data to high-frequency tones
DSLAM translates back analogue signals from other home PCs
to digital data, separates data and phone signals, and sends
each into the respective networks
Residential telephone line carries data and traditional
telephone signals simultaneously, encoded at different
frequencies
• Upstream and downstream data rates are different
(asymmetric)
o Cable

Uses cable television company’s existing infrastructure

Uses fiber and coaxial cables, referred to as hybrid fiber coax
(HFC)
Cable modems connect to home PCs (similar to DSL modems),
convert digital signals to analogue, and divide the HFC network
into typically asymmetric channels, defined by Data Over Cable
Service Interface Specification (DOCSIS) to be 42.8 Mbps
upstream and 30.7 Mbps downstream
Cable modem termination system (CMTS) converts analogue
signal from cable modems back into digital
Shared broadcast medium majorly hinders downstream rate,
and creates the need for a distributed multiple access protocol
to coordinate transmissions and avoid collisions
o Fiber to the home (FTTH)

Active optical networks (AONs) direct signals only to the

relevant customer, passive optical networks (PONs) do not
Also referred to as fiber optic service (FIOS)
Optical network terminators (ONTs) are present in each home
and connect to a neighborhood splitter via a dedicated optical
fiber
Optical line terminator (OLT) convert between optical and
electrical signals
Gigabit per second potential
o Dial-Up
Over traditional phone lines, based on same model as DSL,
slow (56 kbps)
o Satellite
> 1 Mbps
o Local Area Networks (LAN)
Access in the enterprise and home typically work using
Ethernet switches and WiFi base stations
• Both Mbps – Gbps
o Wide-area wireless access
Third-generation (3G) wireless, provides packet-switched
Internet access at > 1 Mbps
Fourth-generation (4G)
Long-Term Evolution (LTE) is similar to 3G, downstream > 10
Mbps
• 1.2.2 Physical Media
o Guided (waves along a solid medium) and unguided (waves
propagate in the atmosphere and in outer space)
o Twisted-pair copper wire
Least expensive, most common guided transmission medium
Two insulated copper wires ~1 mm thick in a spiral pattern
• Reduces electrical interference from similar close by
pairs
Bundled in a cable
Single communication link
Unshielded twisted pair (UTP) is common for LANs, 10 Mbps –
10 Gbps
o Coaxial cable
Guided and can be a shared medium
Two concentric copper conductors
Insulation and shielding
Common in cable television systems
High data transmission rates
o Fiber Optics
Thin, flexible, guided medium which conducts pulses of light
10 Gbps – 100 Gbps
Immune to electromagnetic interference
Low signal attenuation up to 100 km
Very hard to tap
High cost
o Terrestrial Radio Channels
Carry signals in the electromagnetic spectrum, potentially for
long distances
No physical wire, can penetrate walls, provide connectivity to a
mobile user
Potential packet loss and decrease in signal strength over
distance and around/through obstructing objects (shadow
fading)
o Satellite Radio Channels
Geostationary satellites
• Covers same area over Earth over time
• 36,000 km above Earth’s surface
 • Signal propagation of 280 ms
• >100 Mbps
Low-earth orbiting (LEO) satellites
• Independent orbit
• Require many satellites to continuously cover an area

1.3 – The Network Core

• 1.3.1 Packet Switching

o Long messages are divided into packets

o Packets travel through communication links and packet switches
(routers and link-layer switches)
o Store-and-forward transmission: packet switch must receive entire
packet before beginning to transmit the first bit of the packet onto
the outbound link
o Packet switches have output buffers for each attached link, which
store packets before transmission
Resulting in queueing delays
Packet loss if the output buffer a packet needs to be placed
into is full
 o Forwarding tables in each packet switch map portions of destination
addresses to outbound links, and the entries of forwarding tables are
set by routing protocols
o Advantages
Sharing of transmission capacity
• Allows for faster service (consider file transfer with TDM
and little user activity)
Simpler to implement
Less costly
Provides essentially the same performance as circuit switching
and can support significantly more users (because user activity
is probabilistic)
o Disadvantages
Not great for real-time services due to variable end-to-end
delays
• 1.3.2 Circuit Switching

o Resources (buffers, link transmission rate) needed along a path are

reserved for the duration of the communication session between end
systems
o Traditional telephone networks have this structure
Circuit: the connection between the sender and receiver
o Multiplexing
 Frequency-division multiplexing (FDM)

• Frequency band of a link is divided among the

connections established across the link
o Bandwidth: width of the frequency band (4 KHz
above for both bands)
• FM (frequency modulation) radio stations use FDM
Time-division multiplexing (TDM)

• Time is divided into frames of fixed duration, frames are

divided into a fixed number of slots
o Advantages
Guarantees smooth service
o Disadvantages
Dedicated circuits waste resources during silent periods
Difficult to establish end-to-end circuits and reserve end-to-
end transmission
• 1.3.3 A Network of Networks

o ISPs are interconnected in a tiered structure

o Access ISPs (customers) pay the regional ISP (provider) it connects to
o Regional ISPs (customers) pay the tier-1 ISP (provider) it connects to
Regional ISP may be split into provincial ISPs and national ISPs
o Points of presence (PoPs)
Exist on all tiers above access level
Are groups of one or more routers at the same location in the
providers network where customer ISPs can connect into the
provider ISP
o Multi-home: customer ISP connected to multiple provider ISPs
o Peering: pairs of ISPs at the same level can directly connect their
networks together
All traffic between them should only be for direct connections
between hosts connected to the ISPs
Allows them to avoid the cost of routing through provider ISPs
o Internet Exchange Points (IXPs)
Third parties which create a meeting point for multiple ISPs to
peer
o Content-provider networks
Private networks which try to bypass the upper tiers of the
Internet by peering with lower-tier ISPs (either directly or at
IXPs)
 Ultimately still have to be customers to tier-1 ISPs, but reduce
their costs by exchanging less traffic with them through
peering
o Around a dozen tier-1 ISPs
o Hundreds of thousands of lower-tier ISPs

1.4 – Delay, Loss, and Throughput in Packet-Switched Networks

• 1.4.1 Overview of Delay in Packet-Switched Networks
o Total nodal delay
Nodal processing delay (microseconds)
• Examine packet header
o Check for bit errors
• Determine output link
Queueing delay (microseconds – milliseconds)
Transmission delay: transmit packets onto link (microseconds –
milliseconds)
Propagation delay (milliseconds)
• 1.4.2 Queueing Delay and Packet Loss
o Varies from packet to packet, as a result of
Fluctuations in traffic intensity (La/R)

• L = bit length of packets

• a = average rate packets arrive at a queue
• R = transmission rate
Burstiness of packet arrival
 o Packet loss
Full, finite capacity queues will have to drop packets upon new
arrivals
• 1.4.3 End-to-End Delay
o Depends on physical medium
End systems using shared mediums (such as WiFi) may
purposefully delay its transmission as part of its protocol for
sharing the medium with other end systems
o Traceroute
• 1.4.4 Throughput in Computer Networks
o Instantaneous and average throughput
o Transmission rate along a path is equal to that of the bottleneck link
Fluid analogy
Typically the access network

1.5 – Protocol Layers and Their Service Models

• 1.5.1 Layered Architecture
o Each layer provides its service via
Certain actions
Using the services of the layer directly below it
o Advantages to layering
Easier to update system components
o Disadvantages
Potential duplicate lower-layer functionality
Isolation of information
o Internet protocol stack
Application layer
• Distributed over multiple end systems
• Messages
Transport layer
• Connection-oriented service to its applications
• Segments
Network layer
• Host to host
• Datagrams
Link layer
• Node to node
 • Frames
Physical layer
• Move individual bits within frames from node to node
o Open Systems Interconnection (OSI) protocol stack
Presentation layer and session layer below application layer
Presentation layer
• Deals with message interpretation
• Data compression, encryption, and description
Session layer
• Delimiting and synchronizing data exchange
• Facilitate checkpointing and a recovery scheme
• 1.5.2 Encapsulation
o Packets from above layers become the payloads of packets from
below layers, with a new packet header prepended
o Fragmentation
o Tunneling

1.6 – Networks Under Attack

• Malware: malicious software
• Botnet: group of similarly compromised devices
• Malware is often self-replicating: seeks entry into other hosts over the
Internet from newly infected hosts
• Viruses: Malware requiring user interaction to infect the user’s device
• Worm: Malware that can enter a device without explicit user interaction
• Denial-of-service (DoS) attacks render a piece of infrastructure unusable by
legitimate users
o Vulnerability attack to stop services or crash systems
o Bandwidth flooding on the target’s access link
o Connection flooding to consume all threads/processes
o Distributed DoS (DDoS) are DoS attacks from multiple sources
• Packet sniffing
o Wireshark
• IP spoofing: injecting packets into the Internet with a false source address

1.7 – History of Computer Networking and the Internet

• 1.7.1 The Development of Packet Switching: 1961-1972
 o Packet switching as an efficient and robust alternative to circuit
switching
o ARPAnet: the first packet-switched computer network
By the end of the 1969 had 4 nodes
By the end of the 1969 had 15 nodes
• 1.7.2 Proprietary Networks and Internetworking: 1972-1980
o Network control protocol (NCP): first host-to-host protocol
o New packet-switching networks developed: ALOHAnet’s microwave
network and DARPA’s packet-satellite and packet-radio networks
o ALOHA protocol, the first multiple-access protocol (for a radio
frequency), was developed
o Internetting: mission to network all created networks
o TCP split into modern day TCP and IP
o Ethernet protocol was built on the ALOHA protocol
o By the end of the decade, there were approximately 200 hosts
connected to the ARPAnet
• 1.7.3 A Proliferation of Networks: 1980-1990
o Universities linked
o 1983 – transition from NCP to TCP/IP
o DNS developed
o French launched the Minitel project, which aimed to bring data
networking into everyone’s home
o 1986 – NSFNET (National Science Foundation) created to provide
access to NSF-sponsored supercomputing centers
By the end of the decade, served as a primary backbone linking
regional networks
o By the end of the decade, there were approximately 100,000 hosts
connected to the ARPAnet
• 1.7.4 The Internet Explosion: The 1990s
o ARPAnet ceased to exist
o 1991 – NSFNET removed its restrictions its the commercial use
o 1995 – NSFNET was decommissioned. Internet backbone traffic was
carried by commercial ISPs
o Emergence of World Wide Web application, which was invented at
CERN
o E-mail
o Instant messaging
 o Peer-to-peer file sharing of MP3s
o Internet startups flooded the stock market, and many collapsed
• 1.7.5 The New Millennium
o Aggressive deployment of broadband Internet access to homes
o Ubiquity of high-speed (> 54 Mbps) public WiFi networks and
medium-speed (tens of Mbps) Internet access via 4G cellular
telephony networks
o Online social networks
o Private networks by online service providers
o Cloud applications

Chapter 2 – Application Layer

2.1 – Principles of Network Applications

• 2.1.1 Network Application Architectures
o Application architecture dictates how the application is structured
over various end systems
o Client-server architecture
 Server
• Always-on host
• Fixed, well-known IP address
Client
• Issues requests
Clients communicate indirectly via a server
Data centers are used to create a powerful virtual server, to
better deal with all of the client requests
o Peer-to-Peer (P2P)

Minimal (or no) reliance on dedicated servers in data centers

Direct communication between pairs of intermittently
connected hosts (peers)
Self-scalability
• Peers generate workload by requesting files, but also
add service capacity to the system by distributing files to
other peers
Disadvantages due to highly decentralized structure
• Security
 • Performance
• Reliability
o Hybrids of client-server and P2P
For example, many instant messaging applications use servers
to track the IP addresses of users, but user-to-user messages
are sent directly between hosts
• 2.1.2 Processes Communicating
o Process – program running within an end system by rules governed
by the system’s operating system
Want process communication which works independently
from the end systems’ operating systems
o Processes exchange messages
o Broadly, in each communication session between a pair of processes,
there is always a client and server process
Client process initiates communication, and downloads from a
network
Server process waits to be contacted, and uploads to a
network
o Socket: software interface which allows a process to send and
receive messages from a network
Application Programming Interface (API) between the
application and the network
o Application developer’s control on the transport-layer
Choice of transport protocol
Ability to configure transport-layer parameters
• Maximum buffer size
• Maximum segment size
o Host is identified by IP address
o Process is identified by port number
• 2.1.3 Transport Services Available to Applications
o Reliable data transfer
Sending process can pass its data into the socket and know
with complete confidence that the data will (eventually) arrive
without errors (likely) at the receiving process
Not necessary for loss-tolerant applications
o Throughput
Guarantees are great for bandwidth-sensitive applications
 Not necessary for elastic applications
o Timing
Guarantees are great for interactive real-time applications
Not necessary for non-real-time applications
o Security
Confidentiality
Data integrity
End-point authentication
• 2.1.4 Transport Services Provided by the Internet
o TCP
Connection-oriented service
• Client and server exchange transport-layer control
information with each other before the application-level
messages begin to flow (handshaking)
• Full-duplex TCP connection forms between the sockets
of the two processes
Reliable data transfer service
Congestion-control mechanism
o UDP
Connectionless service
Unreliable data transfer service
No congestion-control mechanism
o Security
Provided by neither TCP or UDP
Secure Sockets Layer (SSL) enhancement for TCP providing
confidentiality, data integrity, end-point authentication
• Implemented in the application layer
 • 2.1.5 Application-Layer Protocols
o Define
The types of messages exchanged between application
processes
The syntax of various message types (fields and delineation
between fields)
Semantics of the fields
Rules for determining when and how processes send and
respond to messages
• 2.1.6 Network Applications Covered in This Book
o The Web
o Electronic mail
o Directory services
o Video streaming
o P2P applications

2.2 – The Web and HyperText Transfer Protocol (HTTP)

• 2.2.1 Overview of HTTP

o Web’s application-layer protocol

o Implemented in client and server programs
o Web page
Consists of objects (files)
Base HTML file with referenced objects
 Each object is addressable by a Universal Resource Locator
(URL)
o Uses TCP
o Client-server application architecture
o Stateless
Maintains no information about the clients
o Persistent connection by default
• 2.2.2 Non-Persistent and Persistent Connections
o Non-Persistent: each request/response pair is sent over a separate
TCP connection
o Persistent: each request/response pair is sent over the same TCP
connection
o Round-trip time (RTT) – time for a small packet to travel from client
to server and then back to the client

o Persistent connections avoid the extraneous

Two RTTs per TCP handshake and object request
Allocation of TCP buffers and variables
Delay when not using pipelining (HTTP/2)
• 2.2.3 HTTP Message Format
o ASCII
o Request

o
Methods
• GET
• POST
• HEAD
• PUT
• DELETE
Browser generates header lines as a function of
• The browser type and version
• User configuration of the browser
• Whether the browser currently has a cached, but
possibly out-of-date version of the object
o Response
 Web servers generate headers in a similar way to browsers
• Different products, versions, configurations
• 2.2.4 User-Server Interaction: Cookies
o An HTTP server is stateless
Simplifies server design
However, often desirable to identify users
o Cookies
• 2.2.5 Web Caching

o Functions as both a server and a client

o Typically purchased and installed by an ISP
Relatively cheap – public-domain software that runs on
inexpensive PCs
o Can substantially reduce the
Response time for a client request
Traffic on an institution’s access link to the Internet
Web traffic in the Internet as a whole
o Want to maximize cache hit rates
o Content Distribution Networks (CDNs) install many distributed caches
throughout the Internet, localizing much of the traffic
o A cache can verify that its objects are up to date using conditional
GET messages
GET method
If-Modified-Since header
2.3 – Electronic Mail in the Internet

• User agent – allow users to

o Read
o Reply to
o Forward
o Save
o Compose messages
• Mail server
o Each recipient has a mailbox in a mail server, which
Manages and maintains the messages sent to the recipient
o Authenticates recipients
o Deals with failure
Holds messages in a message queue and attempts
(re)transmission until success
• 2.3.1 SMTP
o Transfer messages from senders’ mail servers to the recipients’ mail
servers
o Everything encoded in 7-bit ASCII (old protocol, designed before
people emailed large attachments)
o Common commands
HELO
MAIL FROM
RCPT TO
DATA
QUIT
o Each command followed by CRLF
o Terminate DATA with CRLF.CRLF
• 2.3.2 Comparison with HTTP
o Pull vs push protocol
HTTP is mainly a pull protocol (pulling uploaded information to
a server)
SMTP is primarily a push protocol
o Encoding
SMTP requires each message to be encoded in 7-bit ASCII
format, including non 7-bit ASCII (which is encoded and
decoded appropriately)
HTTP data does not impose this restriction
o Handling other media types
HTTP encapsulates each object in its own HTTP response
message
SMTP places all of the message’s objects into one message
• 2.3.3 Mail Message Formats
o Message header
From:
To:
Subject:
o Blank line after message header, then body
• 2.3.4 Mail Access Protocols
o Could have recipient’s local PC run a mail server, to create more
direct communication, but recipient’s PC would have to be always on
and connected to the Internet
o Easier to have the recipient’s local PC run a user agent program
which pulls from the appropriate mailbox in the appropriate mail
server (via a mail access protocol)
o Post Office Protocol – Version 3 (POP3)
Simplistic – does not carry state information across POP3
sessions
Does not provide any means for a user to maintain a folder
hierarchy on a remote server that can be accessed from any
computer
o Internet Mail Access Protocol (IMAP)
Each message associated with a folder
Maintains user state information across IMAP sessions
o HTTP
2.4 – DNS – The Internet’s Directory Service
• 2.4.1 Services Provided by DNS
o Identify a host via hostname and IP address
Hostnames preferred by people, fixed-length IP addresses
preferred by routers
o Domain name system (DNS) translates memorable hostnames to IP
addresses
Distributed database implemented in a hierarchy of DNS
servers
• Often UNIX machines running Berkeley Internet Name
Domain (BIND) software
o Application-layer protocol that allows hosts to query the distributed
database
o Commonly employed by other application-layer protocols (HTTP,
SMTP) to translate user-supplied hostnames to IP addresses
o Host aliasing – hosts with complicated hostnames can have alias
names
o Mail server hostname aliasing
o Load distribution
Servers can be replicated over multiple end systems with
distinct IP addresses
DNS can associate a set of IP addresses with one canonical
hostname
The order of the sequence of IP addresses in the set is rotated
within each reply
• Load balances since the first IP address in the reply is
commonly used
• 2.4.2 Overview of How DNS Works

o Distributed to avoid
A single point of failure
High, growing traffic volume
Having a large distance between querying clients
Constant maintenance to account for constant new hosts
o Classes of DNS servers
Root DNS servers
• Provide IP addresses of top-level domain (TLD) servers
• Over 400 scattered globally
TLD servers
• com, org, net, uk, fr
• Provide IP addresses for authoritative DNS servers
Authoritative DNS servers
 • Houses an organization’s publicly accessible DNS records
that map the names of hosts within the organization to
IP addresses
o Local DNS (LDNS) server
Provided by ISPs to allow connected hosts to query the DNS
database
o In general, TLD servers do not always know the IP addresses of
authoritative DNS servers for the queried hostname
May only know of an intermediate DNS server (often have
hostname dns.organization), which knows the authoritative
DNS server for the hostname
o Recursive and iterative queries
o DNS servers cache responses to queries for some time (often set to
two days)
• 2.4.3 DNS Records and Messages
o DNS servers store resource records (RRs)
Four-tuple (Name, Value, Type, time to live (TTL))
The meaning of Name and Value depend on Type
• Type=A
o Name is a hostname
o Value is the IP address of the hostname
o (relay1.bar.foo.com, 145.37.93.126, A, TTL)
• Type=NS
o Name is a domain
o Value is the hostname of an authoritative DNS
server that knows how to obtain the IP addresses
for hosts in the domain
o (foo.com, dns.foo.com, NS, TTL)
• Type=CNAME
o Name is an alias hostname
o Value is a canonical hostname for the alias
hostname
o (foo.com, relay1.bar.foo.com, CNAME, TTL)
• Type=MX
o Name is an alias hostname
o Value is a canonical name of a mail server for the
alias hostname
 o (foo.com, mail.bar.foo.com, MX, TTL)
o MX record allows a company’s mail server and
Web server to have identical (aliased) hostnames

o
o nslookup program to send DNS query messages
o Insert records into the DNS database via a registrar
Commercial entity that verifies the uniqueness of a domain
name and enters it into the DNS database for a small fee
o Need to provide names and IP addresses of your primary and
secondary authoritative DNS servers
o DNS vulnerabilities
DDoS to TLD servers
• Partial damage control from caching at LDNS servers
Man-in-the-middle
• Intercept queries from hosts and return bogus replies
DNS poisoning attack
• Attacker sends bogus replies to a LDNS server, tricking
the server into accepting bogus records into its cache

2.5 – Peer-to-Peer File Distribution

•
• BitTorrent
o Torrent: collection of all peers participating in the distribution of a
particular file
o Peers may selfishly or altruistically leave or stay in the torrent after
acquiring the entire file
o Tracker: infrastructure node
When a peer joins a torrent, it registers itself with the tracker
The tracker randomly selects a subset of the peers in the
torrent and gives their IPs to the new peer
Peers periodically inform the tracker that they are still in the
torrent
o Attempt to form TCP connections with each peer, to form a set of
neighboring peers
o Periodically, ask neighboring peers which chunks they have
o Rarest first: request chunks that are the rarest among her neighbors
o Give priority to neighbors that are supplying data at the highest rate
(tit-for-tat incentive mechanism)
Top four highest supply rate neighboring peers are sent chunks
back (unchoked)
 • Overtime, peers capable of uploading at compatible
rates tend to be in each other’s neighborhoods
o Every 30 seconds, one additional, randomly selected neighbor is also
sent chunks (optimistically unchoked)
o All other neighboring peers not in these five are choked
• Distributed Hash Tables (DHT)
o Database entries distributed over peers in a P2P system
o Partition keyspace
o Ring structure
Peers know some entries, the IP addresses of their two
neighboring peers, and one distant peer
If peer is queried and doesn’t have the corresponding value,
asks neighboring/distant peer (whichever is closest to key with
respect to keyspace partition)
o Trust neighbors to give true values
o More secure since no single point of failure

2.6 – Video Streaming and Content Delivery Networks

• 2.6.1 Internet Video
o Prerecorded video placed on servers
o Users send requests to servers to view videos on demand
o Video
Sequence of images being displayed at a constant rate (24-30
images/second)
• Uncompressed image: array of pixels encoded into a
number of bits to represent luminance and color
High bit rate
Compressible
• To different rates
For continuous playout, the network must provide an average
throughput to the streaming application that is at least as large
as the bit rate of the compressed video
• 2.6.2 HTTP Streaming and Dynamic Adaptive Streaming over HTTP (DASH)
o Video encoded into several different versions, each with a different
bit rate and, correspondingly, a different quality level
o Allows clients to select different quality chunks based on perceived
bandwidth
 Exponentially weighted moving average accounting for
variance
o Manifest file: provides a URL for each video version along with its bit
rate
Client selects one chunk at a time by specifying a URL and a
byte range in an HTTP GET request message
• 2.6.3 Content Distribution Networks (CDNs)
o Manage servers in multiple geographically distributed locations
o Stores copies of the videos in its servers
o Attempts to direct each user request to a CDN location that will
provide the best user experience

• (Note (6) should be a double-headed arrow)

Intercept requests for videos using DNS
Determine a suitable CDN server cluster for that client
• Geographically closest
o Tends to work well for majority of clients
 o Can be bad for some, since geographically closest
may not minimize number of hops of the network
path
o Some end-users are configured to use remotely
located LDNSs, in which case this is a bad heuristic
for distance
o Ignores variance in delay and available bandwidth
over time of Internet paths
• Based on current traffic conditions
o Real-time measurements of delay and loss
between clusters and clients
o Make all clusters within a CDN periodically send
probes to all LDNSs globally
Unfortunately, many LDNSs are configured
to ignore such probes
Redirect the client’s request to a server in that cluster
o Can be private (Google’s CDN) or third-party (Akamai)
o Placement philosophies
Enter deep: deploy server clusters in access ISPs, globally
• Reduce user-perceived delay
• Highly distributed, difficult to maintain and manage the
clusters
Bring home: build large clusters at a smaller number of sites
(typically IXPs)
• Greater user-perceived delay
• Lower maintenance and management overhead
o CDNs may not place copies of every video in every cluster
Resolved by clusters retrieving the videos from other units and
storing a local copy while streaming the video
Remove videos that are not frequently requested
• 2.6.4 Case Studies: Netflix, YouTube, and KanKan
o Netflix
Combines Amazon cloud and its own private CDN
infrastructure
Amazon cloud
• Content ingestion: studio master versions of movies are
uploaded to hosts in the Amazon cloud
 • Content processing: the machines in the cloud create
many different formats for each movie (for all client
video players), each at multiple bit rates, allowing for
DASH
• Uploading versions to its CDN

Initially employed third-party CDN companies to distribute

video content, but now has its own private CDN
• Server racks in both IXPs and residential ISPs, combining
bring home and enter deep philosophies
o Each server rack has several 10 Gbps Ethernet
ports and over 100 terabytes of storage
• Does not use pull-caching – instead uses push caching –
caching videos to its CDN servers during off-peak hours
o YouTube
Private CDN to distribute videos
Server clusters in many hundreds of different IXP and ISP
locations
Uses pull caching
 Does not use adaptive streaming – instead requires users to
manually select a version
Limits prefetching of video to reduce resource waste via
repositioning and early termination
Processing of uploaded content occurs within Google data
centers
o Kankan
P2P, along with client-server delivery
Similar to BitTorrent file downloading
Pushes video content to hundreds of servers in China
Recently migrated to a hybrid CDN-P2P streaming system
• Client requests beginning of content from CDN servers,
and in parallel requests content from peers
• When total P2P traffic is sufficient for video playback,
the client ceases streaming from the CDN, only using
peers
• If P2P streaming traffic becomes insufficient, the client
restarts CDN connections and returns to the mode of
hybrid CDN-P2P streaming

2.7 – Socket Programming

• Send raw byte representation of data
• 2.7.1 Socket Programming with UDP

o Specify (server IP, server port) when sending

 • 2.7.2 Socket Programming with TCP

Chapter 3 – Transport Layer

3.1 – Introduction and Transport-Layer Services
• 3.1.1 Relationship Between Transport and Network Layers
o Transport-layer protocol provides logical communication between
processes running on different hosts
o Network-layer protocol provides logical communication between
hosts
Logical communication: appearance of direct connection
• 3.1.2 Overview of the Transport Layer in the Internet
o Internet provides User Datagram Protocol (UDP) and Transmission
Control Protocol (TCP)
o Transport layer segments
o Internet’s network-layer protocol (IP) is a best-effort (unreliable)
delivery service

3.2 – Multiplexing and Demultiplexing

• Demultiplexing: delivering data in a transport-layer segment to the correct
socket
• Multiplexing: gathering data chunks at the source host from different
sockets, encapsulating each data chunk with header information (for later
demultiplexing), creating segments, and passing segments to the network
layer
• Uses source and destination port numbers
o 16-bit number ranging from 0 to 65535
o 0 to 1023 are well-known and reserved for use by well-known
application protocols
• UDP sockets are fully identified by the two-tuple (destination IP address,
destination port number)
• TCP sockets are fully identified by a four-tuple (source IP address, source
port number, destination IP address, destination port number)
o Need this additional information due to the protocol being
connection-oriented
• Port scanning: determining which applications are listening on which ports
o nmap
• Not always a one-to-one correspondence between connection sockets and
processes – today’s high-performing Web servers often use one process,
 and create a new thread with a new connection socket for each new client
connection

3.3 – Connectionless Transport: UDP

• Advantages for applications using UDP
o Finer application-level control over what data is sent, and when
o No connection establishment
No handshaking between sending and receiving transport-
layer entities before sending a segment
o No connection state
o Small packet header overhead

• 3.3.1 – UDP Segment Structure

 o
• 3.3.2 – UDP Checksum
o 1s complement of the sum of all the 16-bit words in the segment
o Example of system design’s end-end principle

3.4 – Principles of Reliable Data Transfer

• 3.4.1 Building a Reliable Data Transfer Protocol
o Use finite-state machine (FSM) definitions for the sender and
receiver
o rdt1.0 – a protocol for a completely reliable channel
Sending side waits for procedure call from upper-layer
protocol and sends packet
Receiving side waits for procedure call from lower-layer
protocol and extracts packet
o rdt2.0 – reliable data transfer over a channel with bit errors
Need a form of error detection, attach checksum field to all
packets
Need receiver feedback
• Positive acknowledgements (ACK) and negative
acknowledgements (NAK)
• NAKs trigger retransmission of packets
• Automatic Repeat reQuest (ARQ) protocols: reliable data
transfer protocols based on retransmissions
Stop-and-wait protocols: sender does not send a new piece of
data until it is sure that the receiver has correctly received the
current packet (via an ACK)
 o rdt2.1 – accounting for potentially corrupted receiver feedback
Add sequence numbers to packets. That way if the receiver
sent an ACK which became corrupted into a NAK, causing the
sender to retransmit the packet the receiver already received,
the receiver would be able to know that the new packet is a
duplicate packet
o rdt2.2 – equivalent to rdt2.1, but is NAK free
Instead of sending a NAK when receiving a corrupted packet,
send an ACK + sequence number for the last correctly received
packet
o rdt3.0 – reliable data transfer over a lossy channel with bit errors
Implement a countdown timer that can interrupt the sender
after a given amount of time, which prompts retransmission of
the corresponding packet
rdt3.0 is sometimes known as the alternating-bit protocol,
since sequence numbers alternate between 0 and 1
• 3.4.2 Pipelined Reliable Data Transfer Protocols
o Stop-and-wait protocols have very low link utilization

o
o To accommodate pipelining
Range of sequence numbers must be increased
Sender and receiver sides of the protocols may have to buffer
more than one packet
o Two approaches: Go-Back-N (GBN) and selective repeat (SR)
protocols
• 3.4.3 Go-Back-N

o Sender transmits multiple packets

o Can have no more than N unacknowledged packets in the pipeline
Must track this number, and decline the procedure call to send
if there are already N unacknowledged packets in the pipeline
Limit creates flow control
o Sliding-window protocol: window slides over once base packet is
ACKed
o Receiver feedback is a cumulative acknowledgement, indicating that
all packets within a sequence number up to and including n have
been correctly received
o Sender starts timers for each transmitted packet. If any of them
timeout, the sender retransmits all previously sent, unacknowledged
packets (usually equal to window size N, hence the name Go-Back-N)
Due to this retransmission of everything, the receiver discards
out-of-order packets – no buffering is done
Bad when window-size and bandwidth-delay product are both
large, as many packets can be unnecessarily retransmitted
 • 3.4.4 Selective Repeat

o Same as GBN, except

Individually acknowledge correctly received packets
Receiver buffers out-of-order packets
Potential for receiver to send ACKs for packets with a sequence
number below that of rcv_base (if previously transmitted ACK
is lost)
• Thus, window size must be at most half of the size of the
sequence number space
• Note: have assumed in order delivery
o If not, channel can be thought of as buffering packets and
spontaneously emitting these packets at any point in the future
o Due to reusing of sequence numbers, care must be taken to guard
against duplicate sequence number, yet distinct packets
o In reality, assume that packets cannot “live” in the network for
longer than some amount of time (3 minutes assumed in TCP
extensions for high-speed networks)

3.5 – Connection-Oriented Transport: TCP

• 3.5.1 The TCP Connection
o Logical connection, with common state residing only in the TCP
processes in the two communicating end systems
o Full-duplex service
o Point-to-point – between a single sender and a single receiver

o
o Application data passed into TCP send buffer
o TCP intermittently passes chunks of data to the network layer
o Maximum segment size (MSS): maximum amount of data that can be
grabbed from the TCP send buffer and placed in a segment
(accounting for TCP/IP header length, typically 40 bytes)
Determined by largest link-layer frame that can be sent by the
local sending host (maximum transmission unit (MTU), usually
1500 bytes)
Thus typical values are MTU = 1500 bytes, MSS = 1460 bytes
• 3.5.2 TCP Segment Structure

o Flags
CWR and ECE
• Used for explicit congestion notification
URG
• Indicates whether data in the segment that the sending-
side upper-layer entity has been marked as “urgent”
(whose location is pointed to by the urgent data pointer,
rare usage in practice)
ACK
• Indicates whether the value in the acknowledgment
number field is valid or not
PSH
• Indicates whether the receiver should pass data to the
upper layer protocol process immediately (rare usage in
practice)
RST, SYN, FIN
 • Used for connection setup and teardown
o Sequence number: byte-stream number of the first byte in the
segment
Randomly chose initial sequence number, minimize the
possibility that a segment still present in the network from an
earlier, already-terminated connection between the same two
hosts using the same port numbers is mistaken for a valid
segment
o Acknowledgment number: next byte that the sending host expects
from the receiving host (cumulative acknowledgments)
TCP RFC does not impose rules on buffering out-of-order
bytes, but in practice this is often done to increase the
efficiency of network bandwidth
Piggybacking: TCP segment with ACK and data
• 3.5.3 Round-Trip Time Estimation and Timeout
o Want timeout to be greater than RTT to avoid unnecessary
retransmissions, but not so much larger that link utilization is low
o SampleRTT: amount of time between when a segment is sent (passed
to IP process) and received
Most TCP implementations only take one SampleRTT
measurement (track SampleRTT for a single transmitted but
currently unacknowledged segment) at a time (roughly once
per RTT)
Never tracked for retransmitted packets
Fluctuates from segment to segment due to congestion in
routers and varying load on the end systems
o EstimatedRTT: (1-a)*EstimatedRTT + a*SampleRTT
a is recommended to be 0.125
Exponentially weighted moving average (EWMA)
o DevRTT: (1-b)*DevRTT + b*|SampleRTT - EstimatedRTT|
b is recommended to be 0.25
Also EWMA
o Finally, TimeoutInterval: EstimatedRTT + 4*DevRTT
• 3.5.4 Reliable Data Transfer
o TCP ensures that the byte stream that a process reads out of its TCP
receive buffer is exactly the same byte stream as that sent by the end
system on the other side of the connection
 o Specific TCP details and mechanisms
SendBase variable in TCP sender: sequence number of the
oldest unacknowledged byte
Doubling the timeout interval for a segment after a timeout
• Provides a limited form of congestion control
Some versions of TCP have an implicit NAK mechanism – TCP
fast retransmit
• Three duplicate ACKs for a given segment serve as an
implicit NAK for the following segment, triggering
retransmission before timeout
Delayed ACK: when in-order segment arrives with expected
sequence number, wait up to 500 msec for the arrival of
another in-order segment. If next in-order segment does not
arrive in this interval, send an ACK
• 3.5.5 Flow Control

o Eliminate possibility of sender overflowing the receiver’s buffer

o Receiver maintains
LastByteRcvd – LastByteRead <= RcvBuffer
• Since overflowing buffers is not permitted
Receive window rwnd: RcvBuffer – (LastByteRcvd -
LastByteRead)
o Sender maintains
rwnd from receiver: how much free buffer space was recently
available at the receiver
 LastByteSent – LastByteAcked <= rwnd
o Problem: rwnd = 0 being advertised, and sender currently has
nothing to send
Sender will never be told that rwnd has increased (after
application reads from full TCP receive buffer) since this
happens only if the receiver needs to send data, or has an
acknowledgment to send
Solution: sender can send segments with one data byte when
rwnd = 0
• Note: UDP does not provide flow control, hence segment loss can occur at
the receiver due to buffer overflow
• 3.5.6 TCP Connection Management
o Connection establishment (three way handshake)
Client-side sends TCP segment (SYN) with
• SYN bit set to 1
• Randomly chosen initial sequence number
Server-side receives TCP segment and
• Allocates TCP buffers and variables to the connection
• Sends TCP segment (SYNACK) with
o SYN bit set to 1
o Acknowledgement field set to received sequence
number + 1
o Randomly chosen initial sequence number
Client-side receives TCP segment and
• Allocates TCP buffers and variables to the connection
• Sends TCP segment (ACK) with
o SYN bit set to 0
o Acknowledgment field set to received sequence
number + 1
o Potential first payload of data
o Connection closing

Close means resources (buffers and variables) are deallocated

FIN is a TCP segment with FIN bit set to 1
Timed wait is usually either 30 seconds, 1 minute or 2 minutes
Afterwards, the connection formally closes and all resources
(including port numbers) are released
o SYN flood attack
DoS
Send large number of TCP SYN segments, without completing
the third handshake step, exhausting the server’s connection
resources
Defend with SYN cookies
• In response to TCP SYN, server does not create a half-
open TCP connection
 • Instead, creates an initial TCP sequence number that is a
hash function of source and destination IP, port, and a
secret number, and responds with SYNACK segment to
the sender of the TCP SYN message
• A legitimate client responds with an ACK segment,
whose acknowledgment field = output of hash function
+1
o Server verifies this value using the fields present
in the ACK segment, and the secret number stored
in memory. If valid, resources are allocated and a
fully open connection is created
• Otherwise, no harm is done to the server
o nmap
Send TCP SYN segment, and either receive
• TCP SYNACK segment
o Application running TCP on corresponding port,
nmap returns “open”
• TCP RST segment
o SYN reached target host, but the host is not
running an application with TCP on corresponding
port
o Not blocked by firewall
• Nothing
o Likely blocked by firewall

3.6 – Principles of Congestion Control

• 3.6.1 The Causes and the Costs of Congestion
o Offered load: sending + retransmission rate of a host onto a link
o Retransmissions, especially unnecessary retransmissions (for
example due to a premature timeout) cause link utilization to
plummet
o If a packet is dropped along a multihop path, the transmission
capacity that was used at each of the upstream links to forward that
packet to the point at which it is dropped ends up having been
wasted
 • 3.6.2 Approaches to Congestion Control

o End-to-end congestion control

Network congestion must be inferred by the end systems
based only on observed network behavior (packet loss and
delay)
o Network-assisted congestion control
Routers provide explicit feedback to the sender and/or
receiver regarding the congestion state of the network,
typically using a choke packet

3.7 – TCP Congestion Control

• End-to-end congestion control, since the IP provides no explicit feedback to
the end systems regarding network congestion
• TCP congestion-control mechanism operating at the sender tracks the
variable congestion window (cwnd)
o LastByteSent – LastByteAcked <= min(cwnd, rwnd)
min(congestion control limit, flow control limit)
o Bandwidth probing: uses ACKs to increase congestion window
o Uses timeouts or three duplicate ACKs to decrease congestion
window
o Too small of a value and link utilization is unnecessarily low
 o Too high of a value potentially creates lots of congestion, also
lowering link utilization (and wasting transmission capacity up to that
point)
• Slow start

o When a TCP connection begins, the value of cwnd is typically 1 MSS

o cwnd increases by 1 MSS every time a transmitted segment is first
acknowledged
o If there is a loss event
Slow start threshold (ssthresh) = cwnd/2
cwnd is reset to 1 MSS
Slow start is re-entered, until cwnd = ssthresh, in which case
congestion avoidance mode is entered
• Congestion avoidance
o Increase cwnd by 1 MSS every RTT
o If there is a timeout
Same as loss in slow start mode
o If triple duplicate ACKs are received
ssthresh = cwnd/2
cwnd = cwnd/2 + 3 MSS
Fast recovery mode can be entered
 • TCP Tahoe only uses slow start and congestion avoidance modes
• Fast recovery
o Recommended but not required
o cwnd increases by 1 MSS for every duplicate ACK received for the
missing segment that caused the entry of fast recovery mode
o When the ACK for the missing segment is received, congestion
avoidance mode is entered after setting cwnd = ssthresh
o If there is a timeout
Same as loss in slow start mode
o Implemented by TCP Reno

•
o TCP Tahoe and Reno behave the same until triple duplicate ACK at
transmission round 8
o Loss at cwnd = 12, so ssthresh = 6
o TCP Reno sets cwnd = cwnd/2 + 3 (= 9 MSS) and enters fast recovery
mode
o TCP Tahoe sets cwnd = 1 MSS and enters slow start mode
• TCP splitting
o Desirable for cloud services to provide a high-level of responsiveness
o If end system is far from a data center, RTT will be large, potentially
leading to poor response time performance due to TCP slow start
o Solution: clients establish TCP connection to nearby front-end server.
Front-end server maintains a persistent TCP connection to the data
center with a large TCP congestion window
• TCP’s congestion control is often referred to as an additive-increase,
multiplicative-decrease (AIMD) form of congestion control

o Ignore slow start

o Assume loss is indicated by duplicate ACKs rather than timeouts
o Assume fast recovery mode is entered
o “Saw tooth” behavior
• TCP Vegas
o Tries to detect congestion in the routers between source and
destination before packet loss occurs by observing RTT
o If congestion is detected, lower the rate of transmission linearly
• TCP’s congestion control causes it to use high speed links inefficiently
o Only takes one loss event for rate of transmission to completely reset
o Researchers are investigating new versions of TCP for high-speed
environments
• 3.7.1 Fairness
o Link capacity split equally among end systems connected to link
o TCP connection 1 and 2 sharing a link of transmission rate R

Start at point A
• Total throughput into link < R, so no loss
Moves to point B
• Total throughput into link > R, so loss
• Both connections set cwnd = cwnd / 2
Moves to point C
• Connection 1, whose cwnd was greater than that of
connection 2, experiences a greater reduction than
connection 2
o Closer to equal bandwidth sharing
Converges to fairness
• Though in reality, RTT varies
o Hosts with lower RTT are able to grab the available
bandwidth (increase their cwnd) at the link more
quickly as it becomes free
o TCP’s congestion control mechanism incentivizes high bandwidth
applications to run over UDP
o Can use multiple TCP connections in parallel to grab more bandwidth
• 3.7.2 Explicit Congestion Notification (ECN): Network-assisted Congestion
Control

o ECN Echo bit = 1 should cause TCP sender to half cwnd

Chapter 9 – Multimedia Networking

9.1 – Multimedia Networking Applications

• 9.1.1 Properties of Video
o High bit rate (100 kbps – 3 Mbps)
o Compressible
Can produce multiple versions of the same video for flexibility
o Redundancy
Spatial
Temporal
• 9.1.2 Properties of Audio
o Analogue to digital
Pulse Code Modulation
Compressible
• MPEG 1 layer 3 (MP3)
• Advanced Audio Coding (AAC)
• 9.1.3 Types of Multimedia Network Applications
o Streaming stored audio/video
Streaming: begin media playout after receiving only some of
media
Interactive
Continuous playout
Video
• Average throughput is the most important performance
measure
• Streamed from CDN or P2P applications
o Conversational voice/video-over-IP
Voice-over-IP (internet telephony, VoIP)
Delay-sensitive
Loss-tolerant
o Streaming live audio/video
Typically via CDNs

9.2 – Streaming Stored Video

• Extensive client-side application buffering for smooth playback

o Absorb variations in server-to-client delay and server-to-client

bandwidth
• UDP streaming
o Server transmits video at a rate that matches the client’s video
consumption
o Typically consists of a small client-side buffer (holds less than a
second of video)
o Typically uses the upper-level Real-Time Transport Protocol (RTP)
o Client and server also maintain a separate control connection where
the client communicates changes in session state (pause, resume,
reposition)
o Disadvantages
Often fails to provide continuous playout due to fluctuations in
bandwidth between server and client
Requires a media control server, such as a Real-Time
Streaming Protocol (RTSP) server to process client-to-server
interactivity requests and to track client state for each ongoing
client session
Many firewalls block UDP traffic
• Consequently, rarely used
• HTTP streaming

o Server stores video as an ordinary file, addressed by a URL

o Client uses HTTP GET requests over TCP for that URL
o Server sends video as quickly as TCP allows with flow and congestion
control
o Client application begins playback once enough bytes are present in
the client application buffer (decompresses and displays frames)
o Disadvantages
Transmission rate often exhibits “saw-tooth” shape as a result
of TCP congestion control
Potential significant delay as a result of TCP’s retransmission
mechanism
• Fixable with client buffering and prefetching, so typically
used today (YouTube, Netflix)
o Prefetching video
Stored in client application buffer
“Back pressure” from full TCP buffers will force the server to
reduce its transmission rate
• A full client application buffer indirectly imposes a limit
on the rate that video can be sent from server to client
o Early termination and repositioning the video
Client application uses the HTTP byte-range header in the HTTP
GET request message to specify the desired frames
 Early termination and repositioning the video waste bandwidth
and server resources
• Smaller client application buffers reduce waste
• Adaptive HTTP streaming (Dynamic Adaptive Streaming over HTTP, DASH)
o See 2.6.2

9.3 – Voice-over-IP (internet telephony, VoIP)

• 9.3.1 Limitations of the Best-Effort IP Service
o Receiver must take care in determining
When to play back a chunk
What to do with a missing chunk
o Packet loss
Buffer overflow in routers along path between sender and
receiver
Could be eliminated with TCP, but
• Retransmission mechanisms are often considered
unacceptable for conversational real-time audio
applications
• Loss would result in reduction of the sender’s
transmission rate, likely to a rate lower than the
receiver’s drain rate, possibly leading to buffer
starvation and severely impact voice intelligibility at the
receiver
o Thus, most VoIP applications run over UDP (Skype,
unless a user is behind a NAT/firewall which
blocks UDP segments)
o End-to-end delay
Accumulation of transmission, processing, queuing delays in
routers; propagation delays in links; and end-system
processing delays
Perception
• < 150 msecs not perceived by a human listener
• 150 – 400 msecs are acceptable but unideal
• > 400 msecs seriously hinder interactivity
o Packet jitter
Varying queuing delays at routers
• 9.3.2 Removing Jitter at the Receiver for Audio
 o Prepend each chunk with a timestamp
o Delay playout of chunks at the receiver
o Fixed playout delay

Receiver attempts to play out each chunk exactly q msecs after

the chunk is generated
o Adaptive playout delay
Playout time of ith packet is calculated via estimations of
network delay and the variance of the network delay
• Exponentially weighted moving averages for both
• 9.3.3 Recovering from Packet Loss
o Loss recovery schemes
Forward Error Correction (FEC)
• Add redundant information to the original packet
stream
• Send a redundant encoded chunk every n chunks = XOR
of the original n chunks.
o If any single packet in the n+1 packets is lost, the
receiver can recover the packet by XORing all of
the other packets
o If more than one packet is lost, there is no
recovery
 • Create lower-resolution audio stream, and prepend
already sent packets to new packets

Interleaving

Error concealment
• Produce a replacement for a lost packet that is similar to
the original
• Possible via short-term self-similarity
 • Packet repetition
• Interpolation
• 9.3.4 Case Study: VoIP with Skype
o Proprietary
o Clients can use many different codecs
o Audio and video packets via UDP (by default, TCP otherwise)
o Control packets via TCP
o FEC for loss recovery
o Adapts streams to current network conditions
o P2P
o Skype super peers relay data between two callers behind UDP-
blocking NATs
o For video calls with N > 2 participants, each participant’s video
stream is routed to a server cluster, which relays to each participant
the streams of the N-1 streams of the N-1 other participants,
avoiding the likely low bandwidth upstream links of each participant

9.4 Protocols for Real-Time Conversational Applications

• 9.4.1 Real-Time Transfer Protocol (RTP)
o Runs on top of UDP

o
Payload type = audio or video encoding
Synchronization source identifier (SSRC) uniquely identifies the
source of the RTP stream
o No insurance of timely delivery of data, or other quality-of-service
(QoS) guarantees
• 9.4.2 Session Initiation Protocol (SIP)
 o Establishes calls
o Can allow the caller to determine the current IP address of the callee

1: SIP INVITE message to the umass SIP proxy

2: SIP proxy does DNS lookup on the SIP registrar (not shown),
and forwards the INVITE message to the SIP registrar
3: Redirect response since host is registered with the NYU SIP
registrar
o Call management (adding new media streams, changing the encoding
of a media stream, inviting new participants to during the call)
o Ends calls

9.5 Network Support for Multimedia

•
• 9.5.1 Dimensioning Best-Effort Networks
o Bandwidth provisioning: how much capacity to provide at network
links in a given topology to achieve a given level of performance
o Network dimensioning: how to design a network topology (where to
place routers, how to interconnect routers with links, what capacity
to assign links) to achieve a given level of end-to-end performance
o Need
Models of traffic demand between network end points
Well-defined performance requirements
Models to predict end-to-end performance for a given
workload model, and techniques to find a minimal cost
bandwidth allocation that will result in all user requirements
being met
o Internet could support multimedia traffic at an appropriate
performance level if dimensioned correctly, but does not, primarily
due to economic and organizational reasons
 Users may not be willing to pay their ISPs enough for sufficient
bandwidth to support multimedia applications over a best-
effort Internet
Different ISPs would have to be willing cooperate to ensure
that end-to-end paths are properly dimensioned to support
multimedia applications
• 9.5.2 Providing Multiple Classes of Service
o Type-of-service (ToS) field in the IPv4 header could be used for
packet marking
o Ideally provide traffic isolation among classes, so one class is not
adversely affected by another class of traffic that misbehaves
Traffic policing (drop/delay packets that violate a criteria)
Link-level packet-scheduling to provide logically distinct links of
different capacities within the same physical link
o Desirable to use resources (link bandwidth, buffers) as efficiently as
possible
Better than logically distinct solution since this can waste
resources
o Criteria for policing
Long-term average rate (6000 packets/minute)
Peak rate (1500 packets/second)
Burst size (750 packets/instant)
o Leaky buckets
 Packets can only be sent if a token can be removed from the
bucket
Burst rate b
Maximum long term average of rt+b, for any length of time t
Can use two leaky buckets in series to police a flow’s peak rate
• First: flow limited by link limited to r1t+b1
• Second: limits flow after initial limit, limiting peak rate (if
r2 < r1, peak = r2(r1t+b1) + b2)
o Leaky bucket + Weighted Fair Queuing = Provable Maximum Delay in
Queue

dmax = (b1 R w1) / sum(wj) for delay of class 1

• R is the transmission rate of the link
• If there are b1 packets in the queue and each packet is
removed at a rate of at least (R w1) / sum(wj), the last
packet cannot be removed later than (b1 R w1) / sum(wj)
• 9.5.3 Diffserv
o Differentiated service for different classes of traffic
o Network edge router functions: packet classification and traffic
conditioning
o Network core router function: forwarding
 Each packet is forwarded onto its next hops according to per-
hop behavior (PHB) associated with the packet’s class
o Very scalable, no need to maintain connection-specific information
o End users may have to agree to limit its packet-sending rate to
conform to a declared traffic profile
o Metering functions compare incoming packet flow with the
negotiated traffic profile, determine whether a packet is behaving as
expected, and act accordingly (forward if behaving, delay/drop if not)
o Expedited forwarding PHB specifies that the departure rate of a class
of traffic must equal or exceed a configured rate
o Assured forwarding PHB divides traffic into four classes, where each
class is guaranteed with some minimum amount of bandwidth and
buffering
• 9.5.4 Per-Connection Quality-of-Service (QoS) Guarantees: Resource
Reservation and Call Admission
o For a network to make guarantees, it must integrate call admission
A flow must declare its QoS requirement, and have the
network either accept the flow (at the required QoS) or block
the flow
This requires the reservation of sufficient resources at each
and every network router on its source-to-destination path
(call setup)