Comp 343/443     Fall 2011, LT 412, Tuesday 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 7IP address space exhaustion
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

General rules:
    midterm, project(s), final
    grad students will do an additional programming project
    some students (those who have not completed Comp 170) will get an alternative assignment

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:
  1. www/http: file transfers (bulk data)
  2. ssh/telnet, some database transactions: interactive exchanges
  3. VOIP (Voice (telephone) over IP)
  4. 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


1.2.2: sharing


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:
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

Example of store-and-forward delay: one packet sent along a path A-----S-----B. The bandwidth of each link is 1mbps; the packet side is 1kbit, so a packet takes 1ms to traverse a link.

Store-and-forward delay: the packet takes a total of 2 ms to traverse the two links. That is, S cannot begin retransmitting the packet to B until it has completely arrived. We are ignoring propagation time here. (Some switches did at one time implement cut-through, meaning that retransmission on the outbound link began before the packet had fully arrived. In practice, this turns out not to be as useful as one might think.)

There is another form of packetization delay related specifically to voice (and to some other real-time applications where bytes are generated at a specific rate). Voice is generated at 8 bytes/ms (64kbps). The time to fill a 1000 byte packet is thus 125 ms; if you start speaking, your voice cannot even leave the phone for 125 ms. I sometimes call this fill delay. The source is the only point where this delay occurs; packetization delay at internal switches is much smaller. However, there is a second 125 ms delay at the other end. Generally speaking, VOIP protocols use much smaller packet sizes (eg 160 bytes / 20 ms).

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:

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 at the hardware level.
  1. bit errors: typical rate 1 in 107
  2. bit-burst errors, eg due to a refrigerator turning on
  3. packet drops (eg in queues)
  4. backhoes, switch failure and other catastrophic damage
What are some things that can go wrong at the software level?

1.3: Layering & Protocols

headers
HHP - host-to-host protocol (eg IP)
RRP - request-reply protocol (this could be TCP or UDP)
MSP - message-stream 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
  1. Physical (e.g. physical ethernet)
  2. Network interface (eg logical ethernet) (small)
  3. Network (machine endpoints; routing & delivery issues) (BIG)
  4. Transport (reliable streams; process endpoints)
  5. 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/MSP layers; protocol graphs, encapsulation (1.3)
See Figure 1.14 for an IP graph

What would RRP be? Possibly UDP, but also possibly HTTP (on top of TCP): this supports GET requests and responses.
What would MSP be? TCP is the "canonical" answer, but it might also be RTP (Real-Time Protocol), on top of UDP. The reason for not using TCP is that TCP waits for lost packets to be retransmitted; real-time applications like interative voice/video simply need to keep moving.

Multiplexing / Demultiplexing

This is handled above the HTH level.

Network Issues

These are a few of the things to consider.


Section 1.5 (parts may be deferred as necessary)

bandwidth v delay (esp propagation delay)
delay: propagation + bandwidth (transmit) + queue
In this view, switching delay is being counted as bandwidth delay.

basic diagrams:
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 used to 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

When dealing with "large-scale" networks (eg the internet), 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: Consider a simple network with the following properties:

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

Note that sending two smaller packets is faster than one large packet. This is a very real effect, and has sort of put an end to interest in IP "jumbograms". As an extreme case, the (now-seldom-used) ATM protocol (intended for both voice and data) uses packets with 48 bytes of data and 5 bytes of header.


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
collisions
CSMA/CD
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; once the switches have acquired all the destinations, each packet takes the precise path to its destination. (Paths are unique because Ethernet does not allow alternative routes, that is, cycles.)

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

Wi-fi
Other wireless links (Wi-Max, proprietary)

Optical (SONET)
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 end node (eventually)
    fewer collisions (what replaces them?)

The general strategy here is called datagram forwarding, though the particular point of Ethernet switching is that we need a table entry for every host address.

An important issue with switched Ethernet is that each switch must acquire the next-hop information (output-port information) for every destination. If a switch is handling traffic for 10,000 hosts, it must have 10,000 entries. IP, on the other hand, allows some substantial consolidation of the forwarding table.

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

This is the year (2011) that APNIC (Asia-Pacific NIC) runs out of IPv4 addresses.

Actually, it looks like IANA (Internet Assigned Numbers Authority) has also run out of blocks (at least class-A blocks):
http://www.iana.org/assignments/ipv4-address-space/ipv4-address-space.xml

IP has no broadcast (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
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:
  1. allows interconnection ("internetworking") of incompatible LANs
  2. routing strategy is SCALABLE: works well with very large Internet
For the past ~15 years, IP addresses have not used the old Class A/B/C mechanism. Separation into network and host portion is still done, but is now "floating". When you buy a block of IP addresses, you get the network portion, or network prefix. The length of that portion in bits is your prefix length, and is usually appended to the address prefix itself following a slash, eg 
    10.0.0.0/8
    147.126.0.0/16
    19.192.0.0/10   (192 = 1100 0000)

If we don't care about the prefix, we omit it:
    "Class A's are /8 address blocks; class B's are /16"
    "It cost me $2,000/year to get a /18!"

Note that the IP protocol has not been changed, so there is no place in the IP header to specify this prefix length.


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:
Also: uses sliding-windows to establish a maximum number of packets en route at any one time, per connection.
Later, TCP sliding-windows was adapted to manage congestion.

You connect TO a socket = ⟨host,port⟩ (this is actually a "listening" socket, not a "connected" socket.)
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?