Internet Computing

Last updated Jan 17, 2022 Edit Source

• Sapling
• CPSC317

Notes for CPSC 317 (see all notes)

The Internet is a network of networks. The main goals was to integrate a number of separately
administrated entities into a common entity

See also: peer-to-peer, security

Enabling a Permanent Revolution in Internet Architecture by McCauley, Harchol, Panda, Raghavan, and Shenker

The current Internet architecture is both inherently flawed (so we should explore radically different alternative designs) and deeply entrenched (so we should restrict ourselves to backwards-compatible and therefore incrementally deployable improvements).

For example, the decades-long migration effort from IPv4 to IPv6

TLDR;

1. Internet architecture is, and will remain, difficult to change – clean-slate research and projects seem increasingly impossible
2. Current focus is on backwards-compatible designs that have been tested in large-scale operational networks
3. OSI model of the internet sees levels below them as ’logical pipes’ to get something from place A to B

Necessary conditions

• A communication medium
• Source(s) and destination(s)
• Protocol (language)
• Message

Dedicated path between source and destination. Path taken is determined when connection is established. Single stream of info per path

Works well when

• Guaranteed service is valuable
• Demand is steady (unchanging rate)
• Starting a new conversation is rare

Data is divided in packets that are sent individually where each packet is self contained (contains source, destination, and data) and independently routed.

Works well when

• Statistically good performance is good enough
• Demand is bursty (rapidly changing rate)
• Starting a new conversation is frequent

Multiple input streams must share the medium. It must be possible to “demultiplex” at the destination

Types

1. Time division multiplexing: each person gets a certain amount of time on the channel
2. Frequency division multiplexing: each person gets a single frequency band on the channel
3. Code division multiplexing: combines all messages using a specific code that can be decoded if code is known
4. Orthogonal multiplexing: a combination of techniques

Network Protocol Stack (from most abstract to least)

Each layer takes data from above adds header information to create new data unit passes new data unit to layer below

1. Application Layer ( HTTP)
2. Transport Layer ( TCP, UDP)
3. Network Layer ( IP Addresses) – this is the ’thinnest’ part of the network stack!
4. Link Layer
5. Physical Layer

1. Hub - broadcasting through cloning bits ( physical layer)
   1. Simplest and cheapest way to create a network
   2. Lots of unnecessary traffic
   3. Other people can see your traffic
2. Switch - hub but it knows where other hosts are for direct addressing ( link layer)
   1. Keeps a switch table mapping interface number to MAC address
   2. If table is initially empty, will behave like a hub and broadcast
   3. Can start populating switch table based off of sender field from frames
   4. Quicker than a router for internal communication (though some routers have an Ethernet switch built in)
   5. If engineered right, can be full-duplex
3. Router - glue that ties networks together ( network layer)
   1. Does NOT support broadcast
   2. Serves as a bridge between private home network and the network of the internet provider (which can reach the rest of the internet)
   3. Modern routers can also perform
      1. Network address translation ( NAT)
      2. Assigning IP addresses to hosts using DHCP
      3. Broadcast WiFi signal

Methods for fault tolerance in data transmission

1. Parity Bit
   1. Even parity is 1 if number of 1s is odd
   2. Can detect odd number of bit flips
2. 2D Parity Bit
   1. Additional parity bits for each row
   2. Additional parity bits for each column
   3. One last additional bit in last row of parity bits
3. Checksum
   1. Assume data is a sequence of 16-bit integers
   2. Addition, 1’s complement sum, carry out added back in
   3. Checksum is the 1’s complement of the computed value
   4. Compare with the received data (if same, it is ok)
      1. Alternatively, compute the same function over the data and checksum
4. Cyclic Redundancy Check (CRC)
   1. Uses only XOR and shift
   2. Parameterized by constants G and r
   3. r + 1 is the length of G (some power of 2)
   4. G is the generator (arbitrary bit pattern)
   5. Sender wants to send D
      1. Chooses r CRC bits, R such that <D, R> is exactly divisible by G (mod 2)
   6. Receiver knows G, divides <D, R> by G. If the remainder is non-zero an error is detected!
   7. Can detect all burst errors less than $r+1$ bits, and burst errors greater than $r+1$ with probability $1-0.5^r$

1. Half-duplex - both sides can transmit, but only one at a time
   1. Carrier Sense Multiple Access
      1. Listen before sending, only send if no one else is
   2. Collision Detection
      1. While sending, listen to see if what you are sending is garbled
      2. If so, give up
   3. “Try again later” uses binary exponential backoff
      1. Random backoff between 0 and power of 2 (n increases each time)
   4. Turn-based access control
      1. Controlled by centralized party - polls everyone
      2. Controlled in a decentralized manner - passes a token between senders
2. Full-duplex - both sides can transmit at the same time without interference/ NAT

• Bandwidth: max rate at which data can be sent over a link, usually in bits per second
  • Measured in base 10, kilobit is $10^3$ bits, megabit is $10^6$ bits
• Sizes: usually in bytes
  • Measured in base 2, kilobyte is $2^{10}$ bytes, megabyte is $2^{20}$ bytes
  • Except for disk managers, who use bits to make sizes look larger
• Latency: how long is it from when something is sent to when it is received
  • Packet and bit/byte latency: measure start of sending packet/bit/byte to receiving it
  • Round trip time: time to send packet and receiving a response
• Jitter: variation in latency – interpacket variance
• Throughput: amount of data moved from one place to another in a given time
  • Usually measured in bytes not bits
• Goodput: rate at which useful data arrives
  • Does not include headers, encoding costs, etc.

Delay

• Average Service Time: time taken to put the average packet on the wire
  • $S = \textrm{Average packet size} \times \frac{8 bits/byte}{\textrm{Bandwidth}}$
• Processing Delay: figuring out where packet should go
  • Pretty much fixed (almost always variable due to cache hits, network queues, etc. but because of how negligible the times are, we can treat as fixed.)
• Queueing Delay: waiting time to get access to a link
  • Variable
  • Increase over service time when idle
  • $\textrm{Delay}_{\textrm{Queueing}} = \frac{S}{1-U} - S$ where $S$ is the average service time when no other requests and $U$ is the server utilization (usually traffic intensity)
• Transmission Delay: time to write packet to medium
  • Fixed for bits, variable for packets (dependent on size)
  • $\textrm{Delay}_{\textrm{Transmission}} = \frac{\textrm{Message size} \times 8 bits/byte}{\textrm{Bandwidth}}$ for each segment (as each router needs to receive the entire packet before adding it to the queue)
• Propagation Delay: time to move each bit over transmission medium
  • Fixed for meter, variable depending on actual length traveled
  • $\textrm{Delay}_{\textrm{Propagation}} = \frac{\textrm{Total distance}}{\textrm{Link speed}}$
• End-to-end Delay: sum of all sources of delay

Traffic Intensity

• Determined by
  • Number of packets arriving per second: $a$
  • Average packet size: $L$ in bits
  • Transmission rate: rate at which bits are disposed of per second: $R$
• Traffic Intensity: $\frac{La}{R}$

• When we allow multiple packets in flight, modelling this with a finite state machine is less than ideal

• We can use a sliding window to represent the ‘window’ of all bits that are in flight

  • Although, better version of this is to just use a proper protocol like TCP

• Timeout: how much longer should I wait for the ack for the first packet in my window

• Big window costs us memory

• Smaller window is easier to reason about

• When packet is received, send ACK for last packet received in order
• Discard arriving packet if out of order (receiver window is 1)

• Can have a specific number of outstanding (unacknowledged) packets in memory: sender’s window
• Start timer on first packet sent
• On timeout go to last unack’ed packet and resend everything (restart timer)
• Received ACKs may be cumulative (restart timer on receipt)

Simulation Link

• Each packet is ack’ed individually
• Out of order packet is stored for later: receiver’s window
• If a packet arrives whose sequence number is too small to fit in the window it will be ACKed and dropped.
• Only when a packet whose sequence number is too large to fit in the window will it not be ACKed; such packets will be silently dropped.

• Can have a specific number of outstanding (unacknowledged) packets in memory: sender’s window
• Each packet has its own timer and is individually resent if timeout is reached
• ACKs received in order move the sender’s window

Simulation Link

Congestion/Flow Control

• As long as sender’s window size is smaller than the receivers window size, it’s fine
• If the network can’t handle a full window’s worth of data (packets + ACKs are dropped by routers) then sender can reduce sending window
• If all packets are ACKed, we can increase the window again (too slow! we can up the pace)
• Receiver will notify the sender about how much data it can handle (usually included in the ACK)
• If receiver runs out of space, sender will send a packet every once in a while even though its not supposed to just to check if it now has space