Networks Week 1

Comp 343/443 Spring 2010, Corboy Law 522, Thursday 4:15-6:45

Class 1

History,etc, Packet switching. Layering.
Basic services: stream, request/reply
loss v delay tolerance
bandwidth v propagation delay
headers
Sketch of Ethernet, IP, routing, UDP, TCP, ports
Readings: 1.1, 1.2, 1.3, 1.5

Class 2

Basics of network programming: sockets
Ch 2: links
Readings: 1.4, 2.1, 2.2, 2.3, 2.4, 2.5

Class 3

Ch 2: encodings, checksums
Abstract sliding windows

Class 4

Ethernet physical level
Switched Ethernet; switch learning; datagram forwarding
//Demo of fixing port # to receive in wclient.java
Discussion of duplicated REQ, dallying, old-late-duplicates

Class 5

Datagram v Virtual Circuit
IP. fragmentation & routing..
subnets, including variable-sized masks
Distance-Vector intro

Class 6

Slow convergence for DV routing
ARP
TCP connection management; state diagram
connection creation for TCP
Building stateful connections on top of stateless layers
TCP: timewait, sequence numbering

Class 7

Virtual Circuits
Source Routing
Framing; error detection/correction
ICMP, ping &traceroute
DHCP

Class 8

TCP timeouts
IP addressing models; CIDR
IP large-scale routing: geographical v provider
Midterm exam (~ 1 hour)

Class 9

TCP almost-to-Tahoe
TCP congestion avoidance
More routing issues: SPF, BGP, uses of BGP

Class 10

Lingering routing issues, NAT
TCP extensions, RPC, T/TCP, NFS
TCP Tahoe, Reno
ns simulation
Greediness of TCP
Fairness of TCP
cwnd = k / sqrt(P); TCP-friendliness

Class 11

multiple IP addrs is norm: web, localhost, lan, ...
HighSpeed TCP & the problems necessitating it
TFRC
Realtime protocols: RTP, VOIP

Timing:
PacketPairs
DecBIT
ECN - set one bit to indicate ECN-awareness, other to indicate congestion
RED - Random Early Drop
TCP Vegas

RSVP, IntegratedServices
Fair Queuing
RED, RIO

RPC: Sun, BLAST/CHAN
Error-CORRECTING codes

Class 12:

Class 13:

Class 14:

Week 1

READ: 1.1, 1.2, 1.3, and 1.5
READ: "A brief overview of networks", web page
READ: 2.1-2.3 (skim; this material is tricky)
Chapter 1
Network requirements:

www/http: file transfers
VOIP (Voice (telephone) over IP)
network video, 5 250x200 frames/sec, 4 bytes/pixel = 1mB/sec! (Well, we can probably manage with 1 byte/pixel. Still, with video compression at 10-30-fold, compression becomes part of the network infrastructure.)

1.2.1: connectivity v links

links: point-to-point v multiple-access
you need links to get connectivity, though they alone don't guarantee connectivity; you may also need switches. Connectivity and switching work at different levels of abstraction; example: the cloud model. Connectivity requires some mechanism (global?) of addressing. Connectivity also requires switching/routing, usually via a store&forward mechanism.
Addressing and Routing together
unicast v multicast/broadcast

1.2.2: sharing

circuit switching: a separate wire for each connection. Not very practical, but it is still used for POTS (Plain Old Telephone Service) for many residences.
packets & packet switching: packets carry a destination address.
maximum packet size as a sharing requirement (max size also plays role in error handling; ok to lose one packet)
packet switching also accomodates bursty traffic rather well: an idle station doesn't consume any link resources.

Packet switching automatically implements multiplexing (sharing) on links; the demultiplexor is the destination address.
Port numbers allow demultiplexing of multiple streams to same destination host.

other sharing strategies:

STDM: time slicing (popular with voice, where per-connection bandwidth is fixed). Phone network digital DS lines and Sonet optical lines are built around STDM, with each connection getting 1 byte every 1/8000 sec.
FDM: like cable-tv channels; not as efficient; sort of obsolete but note that cable modems may use this

Some of these still appear in cellular telephony (where they still make sense); GSM is a form of STDM; the old AMPS was a form of FDM.

switch issues

store-and-forward!
queuing, congestion, drop, FIFO, droptail, random-drop, round-robin
Sidebar: LANs, MANs, SANs, WANs

1.2.3: common services

Reliable bi-directional channel: this is what the TCP protocol provides.
just read and write; network fixes packet loss and reordering

FTP v NFS: file transfer v filesystem sharing

FTP & Telnet are obsolete, but are still the canonical examples.
New replacements: SFTP/scp, ssh

Some comparisons:

bursty v steady, reliable v loss-tolerant
request/reply v message stream
delay-indifferent v delay-sensitive

delay-sensitive, loss-intolerant is very bad!

voice/video requirements

Voice requires 64,000 kbps (1 byte every 1/8000 sec) (at least in the US). However, we also need uniform propagation delay. The statistical variation in delay is known as jitter. Also, the propagation delay must be relatively small; if the delay is, say, 100ms, then the round-trip "turnover time" is 200ms, which means that whenever one person stops talking and the other starts, a delay of 200ms will be perceived. That value is large enough to be objectionable.

Video simply requires greater bandwidth, in addition to the delay issues. However, note that for one-way video (broadcast), the propagation delay doesn't really matter at all.

Reliability

Four things that can go wrong:

bit errors: typical rate 1 in 10⁷
bit-burst errors, eg due to a refrigerator turning on
packet drops (eg in queues)
backhoes, switch failure and other catastrophic damage

1.3: Layering & Protocols

headers
HHP - host-to-host protocol (eg IP)
RRP - request-reply protocol

Layering and architecture (P&D 1.3) (a method of abstraction)
Levels: 5-level TCP/IP model, 7-level ISO model, summary of levels

Physical (e.g. physical ethernet)
Network interface (eg logical ethernet) (small)
Network (machine endpoints; routing & delivery issues) (BIG)
Transport (reliable streams; process endpoints)
Application (OSI divides this into Application/Presentation/Session, where Presentation includes compression, byte conversions, encryption, and Session includes local echo, buffering, access rights, billing, portmap)

HHP/RRP layers; protocol graphs, encapsulation (1.3)
See Figure 1.14 for an IP graph

Multiplexing / Demultiplexing

Network Issues

These are a few of the things to consider.

Robustness & connectedness - can everyone reach everyone else?
performance - # of users, demand for bandwidth
Internetworking between different protocols (TCP/IP, NetBios, SNA, ...)
Scalability -- can a network grow?
handling growth of the Internet
Authentication and privacy
How can one endpoint tell if someone is who they claim to be?
trusted hosts v. passwords v. public-key systems
interception, modification, spoofing
Gateways and routing - how to handle this efficiently
management - How can managers detect & fix problems; plan for growth
continuous v. bursty traffic
options: quality-of-service, real-time performance, multicast, acctg.

Section 1.5 (parts may be deferred as necessary)

bandwidth v delay (esp propagation delay)
delay: propagation + bandwidth (transmit) + queue
basic diagrams:

one packet, no switching, with propagation delay and bandwidth delay
one packet through a switch
three smaller packets through a switch
prop delay >> bandwidth delay

implications for protocols:
bandwidth-limited: easy to design for; extra RTTs don't cost much
delay-limited, eg to moon (0.3 sec RTT), jupiter (1 hour RTT): Here the protocol designer must focus on minimizing extra RTTs.

cross-continental US roundtrip delay: ~100ms (speed 200 km/ms in cable, ~10,000 km cable route (6000 mi). The cable path is probably a lot shorter, but there's also likely to be queuing delay.)
At 1.0 Mbit, 100ms is 10K or so.
At 1.0 Gbit, 100ms is 10,000K.

From my house, I use satellite internet, which has a propagation delay of ~600 ms.

100 RTTs v 101 RTTs: not significant
1 RTT v 2 RTTs: very significant

Latency = propagation + transmit (bandwidth) + queue
Propagation = Distance / speed_of_light
Transmit = Size / Bandwidth

Usually, to good approximation most of the delay is propagation, and so latency and bandwidth are effectively independent. Note that when propagation delay is small, though, the two are interrelated.

Delay × Bandwidth (usually RTT delay), and pipe size: this is how much we can send before we hear anything back, or how much is "pending" in the network at any one time. Note that, if we use RTT instead of one-way time, then half the "pending" packets will be returning acknowledgements.

1 ms x 10 mbps = 1200 bytes ~ 1 packet
100 ms x 1.5mbps = 20 K
100 ms x 600 mbps = 8 MB!

EXAMPLE:
prop delay 40 µsec
bandwidth 1 byte/µsec (1mB/sec, 8mbit/sec)
Send 200-byte packet (200 µsec bandwidth delay

Case 1: A------------B
Total time: 240 µsec = 200 µsec + 40 µsec

Case 2: A---------------------------------------B, prop delay 4ms
Total time: 4200 µsec = 200 µsec + 4000 µsec

Case 3: A------------R----------B, each link prop delay 40 microsec
Total time: 480 µsec = 240 + 240

Case 4: same as 3, but send two 100-byte packets
Total time: 3*100 + 2*40 = 380 µsec

Diagram for Case 1 and Case 3

Sketch of each layer in TCP/IP

Ethernet: basic LAN.

multiple-access coax versus hubs (not switches, not yet)
everything-is-really-broadcast
Network Interfaces & physical addrs; address uniqueness; addresses assigned at the factory
privacy implications; bandwidth implications
packet address format: 6 bytes
b'cast address: ff:ff:ff:ff:ff:ff
uses of b'cast
multicast

Switched Ethernet: changes lots of the sharing rules; two packets share a link only if their paths overlap.

Other links

DSL
Cable modem
Satellite
DS1 = T1 = 1.544 Mbps = 1544 Kbps = 24 * 64 Kbps + 8Kbps framing
basic voice: 8KBps. T1 = 24 voice bytes + 1 framing bit
DS3 = T3 = 44.736Mbps

Optical
STS-1: 51.840 Mbps, aka OC-1
STS-3: 155.520 Mbps = 3*STS-1, aka OC-3
STS-12: 622.080 = 12*STS-1
STS-48
STS-192, 9.9 Gbps
...
STS-3072, 159 Gbps

STS = synchronous Transport Signal

ATM: Asynchronous trannsport mode

Ethernet switching

forwarding tables
one entry for every node

IP

Net/host portions of addresses
Administratively assigned addresses, so that all hosts on the same network can be given the same net portion
IP delivers to destination network; that network must correspond to a physical LAN that can complete the delivery
CLASSIC IP: classes A, B, C (deprecated)
IP header v Ethernet header (ignore fields of IP header for now)
What an IP ROUTER does
Extraction of Dest_net
Routing tables
Routing table size

No b'cast (well, there is a limited form of IP broadcast, basically intended for your LAN only)

best-effort delivery: if the packet is lost (eg through router queue overflow), the IP layer is not responsible for replacing it. Nor does the IP layer handle acknowledgements.

Example of delivery through a router

IP routing algorithm:
Extract destination_network from packet, and decide if this router connects to that network

YES: local delivery; use LAN to deliver
NO: look up destination_network in forwarding table and send to the next_hop

Significance of administrative assignment : the destination_network part of each IP address represents something about the host's location.

Compare to Ethernet switch, with per-host forwarding table

A    B    C                         D    E    F
+----+----+--S1-------S2------S3----+----+----+
200.0.0                                                      200.0.1

Ethernet forwarding: S2 has to maintain entries for A,B,C,D,E,F
IP forwarding: S2 has two networks to maintain: 200.0.0/24 and 200.0.1/24

Goal of smaller routing tables

Two benefits to IP:

allows interconnection ("internetworking") of incompatible LANs
routing strategy is SCALABLE: works well with very large Internet

UDP

User Datagram Protocol: simple header, no retransmission or acknowledgement features
UDP is a very simple extension to IP; it basically just adds port numbers for application-level multiplexing, and a checksum over the data.
Socket = <host,port>; this acts like a mailbox address in that anything sent to it will be delivered to the receiving application (typically through some sort of socket handle in the underlying OS)

demo of stalkc, stalks (defer)
demo with two clients

TCP

Adds:
stream-oriented connection: no application packetization
reliability
port-number mulitplexing like UDP
Also: uses sliding-windows to keep max traffic en route at any one time
Adapts sliding-windows to manage congestion

You connect TO a socket = <host,port>
Everything sent on that connection goes to same place.
<host,port> doesn't received directly;
only "connected_sockets" receive data; you can only receive data from the endpoint to which you are connected.

TCP semantics are more like "telephone" semantics, for a room with dozens of phones all answering to the same phone number (eg for telephone sales). Each remote phone's data goes to, say, 1-800-LOYOLA1, but it is routed by the phone switch to the correct phone based on what phone it came from. That way, each remote phone has its own connection.

Uses:
TCP: telnet, http/ftp
UDP: voice/video. Why?