Notes for Monday, October 7 - second half of lecture

End of section on internetworking...
 

  Illustration of a router, incoming buffer, incoming and outgoing wires.

-  If incoming bandwidth is greater than outgoing bandwidth, we could have a buffer overflow (and drop packets).


  Illustration of a routing table that maps addresses to "Next Hops".



Section on reliable transmission
 

  We can use the automatic repeat request (ARQ) strategy to implement a reliable link.  ARQ uses ACKs and timeouts.
  Illustration of two ARQ scenarios: one working and one problematic.  The problematic ARQ scenario: A sends packet to B.  B receives packet and sends ACK to A, but B's ACK is lost.  Not having received an ACK, A times out and resends its packet.  B receives A's packet again and does not know that it just accepted the same packet twice.

  The problematic scenario suggests the need for sequence numbers or some kind of unique ID for packets.


  Illustration of sequence numbers





  For increased efficiency, we could use the sliding window algorithm which allows the sender to send multiple packets before receiving an ACK.  We try to select a window size so that we will "fill the pipe"
 ( e.g. 2 x bandwidth x delay ).

  Illustration of sliding window.



- Problem: what if we send a packet with sequence number x and this packet is delayed so that it arrives much later - possibly messing up the sequence number scheme?

Suggested fixes:
   - prevent late delivery
   - timestamp packets
   - in order deliveries

=> Time-to-live (TTL) - set TTL to 256, decrement at each router hop.  If 0, drop it and send an error packet to the sender.
  Illustration of delayed packet delivery.

 

  An illustration of send and receive buffers in a sliding window implementation:

  The sliding window is the span of packets from the first packet that has been sent but not ACK'd to the first packet that has not yet been sent.

Some variations on sliding window: "selective acknowledgements" and "go back N"

 
Estimating round trip time (RTT):

- For the Internet, estimated RTT for the first packet sent is 3-6 seconds.
- After the first packet, estimated RTT is based on measurement
- A problem with RTT measurement is retransmission ambiguity which makes
  the RTT appear too short - we try to throw out these measurements.

  Illustration of retransmission ambiguity.