CS558a Syllabus & Progress

RAP: Rate Adaption Protocol

As mentioned, every end-to-end protocol developed nowadays has to be TCP friendly - it's the "politically correct" thing to do in the present day Internet....
We will study a stream based transfer protocol called RAP by Reza Rajaie, Mark Hendley and Deborah Estrin.
- RAP: An End-to-End Rate-based Congestion Control Mechanism for Realtime Streams in the Internet click here
  This paper appeared in IEEE Infocom network conference in 1999.
- A more extensive tech report is available here: click here
  I suggest you read the tech report instead. The Infocom paper contains very little technical details about the method, and contains mainly description of the results.
Stream based transmission
- A stream is an unreliable data flow
- Think of a stream as a UDP connection if you will....
- Receiver(s) can still return feedback information to the sender, but these feedback information are not used to solicit retransmission - rather, they are used to adjust transmission rate to improve performance
  (Recall that you can adjust the transmission rate of multimedia streams in various ways - increase compression, reduce frame rate or reduce frame size....)
Typicall behavior of a stream transmission protocol:
- A stream transmission protocol will transmit packets at regular intervals
- The interval depends on:
  - Target data rate
  - Maximum packet size
- Example:
  - Target data rate = 100 K bps
  - Maximum packet size = 10 K bits (~ 1.125 K Byte)
  - Transmission interval = 10 packets per sec = 1 packet in 0.1 sec = 0.1 sec interval
Difference between (TCP) packet based and (Multimedia) Stream based transmissions:
- The ACK mechanism of TCP doubles as a built-in pacing mechanism that automatically slows TCP's transmission data rate down when the network shows some sign of congestion:
- Stream based transmission does not use ACK to pace the transmission of packets:
  
  The pace of transmission is controlled by the MPEG encoder (frame rate, frame size, compression parameters, etc).
Transmission Rate Adjustment:
- RAP adjust the transmission rate in a similar way as TCP (to make it "TCP-friendly")
- When RAP determines that there is network congestion, it reduces the transmission rate multiplicatively.
- When RAP determines that there is no network congestion, it "periodically" increases the transmission rate additively.
Congestion Signal used in RAP:
- Also similar to TCP: packet loss
Packet loss determination:
RAP uses 2 mechanisms to determine packet loss...
- If the receiver has not returned an ACK for some packet after a "long time", the sender assumes it is lost.
- The sender can also receives an ACK packet (from the receiver) that explicitly indicates that some packet is lost.
  NOTE: ACK packets are not used to request retransmission. They are used to signal congestion condition.
Data Structure used to implement RAP:
- Transmission history: a list of transmission records containing information of outstanding packets
- SRTT: the RTT estimate maintained by RAP. SRTT is used to determined is a packet is lossed
Estimating the RTT:
- Uses the "Jacobson/Karels formula" to estimate RTT (this method is used in TCP to estimate RTT)
- This method maintains a "running" estimate of the mean and variation (standard deviation) of the measured RTT of the packets.
- The timeout used is computed using a formula of the mean and variation of the RTT.
- Maintaining the running estimate of the mean (SRTT) and variation (varRTT) of SRTT:
  When a new sample of RTT is obtained, the new RTT estimate is computed as follows:
  - SRTT_i+1 = (1 - DELTA) * SRTT_i + DELTA * sampleRTT
  - DELTA is a value between 0.0 and 1.0 (recommended: DELTA = 1/8)
  - Difference = NewSampleRTT - SRTT_i
  - varRTT_i+1 = varRTT_i + DELTA*Difference
- Timeout value used:
  - Timeout = MU * SRTT + PHI * varRTT
  - Recommended: MU = 1 and PHI = 4
Transmission Algorithm:
1. Update SRRT (using the Jacobson/Karels algorithm)
2. Check for potential timeout among the packets in the transmission history.
3. If packet loss then update data rate
4. Otherwise, transmit next packet

Packet Loss Detection Mechanisms:

Timeout-based
ACK-based

If a packet loss is detected, immediately reduce transmission rate

Timeout-based loss detection algorithm:

   for each transmission record i do:
   {
      if ( DepartureTime_i >= CurrentTime + TimeOut )
      {  if ( Status_i != ACKED )
            Assume packet_i is lost
      }
   }

ACK-based loss detection algorithm:

ACK packet format used:
- ACK_last = sequence number of the last packet before N that was received.
  ACK_last = 0 if ACK_curr is the first packet
- N = sequence number of the last packet before sequence number "curr" that is missing
  N = 0 if there is no such packet
- ACK_curr = sequence number of the ACK packet (ACKs the packet with sequence number "curr")
Example 1:
Example 2:
Example 3:

ACK processing:

ACK packets are processed by the following algorithm:

   for each packet i in the Transmision History do
   {
      if ( ( (N < Seq_i) && (Seq_i <= A_curr) )
	   || (Seq_i = A_last) )
      then
	 Seq_i received;
      else
	 Seq_i lost;
   }

OK, so now have seen how RAP detects congestion (congestion = packet drop through timeout or ACK notification).

Next we need to determined how to react when there is congestion and also when there is no congestion.

Increase and Decrease Algorithm:
No surprise:
- Additive increase when no congestion (packet loss) is detected
- Multiplicative decrease when congestion is detected
Frequency of Increase/Decrease:
- Additive increase applied once per SRTT time (see above on how RAP estimates SRTT)
- Multiplicative decrease applied immediately when congestion is detected
But... wait.... the sender does not use window size to control transmission rate....
So how do you update the transmission rate ?
Transmission Data Rate:
- Sender has a send rate variable S
- S_t = transmission rate at time t
- How to change transmission data rate:
  - Packet size is unchanged
  - (The most efficient way to send data is use the max. packet size)

Increase Algorithm:

Suppose the current transmission rate is S_i.
The current Inter Packet Gap is IPG_i.
Thus: S_i = D / IPG_i.
An additive increase change the transmission rate to: S_i+1 = D / IPG_i+1, by using a (smaller) IPG_i+1
Suppose we want to increase IPG_i by some constant value X.
How do we determine the new IPG_i+1 that achieve this ???

The following derivation shows how the new Inter Packet gap can be computed:

			   D
 We know that:      S_i = ------ bps
			  IPG_i

			   D
 and that:        S_i+1 = ------ bps
			  IPG_i+1

 We want to increase transmission rate by X bps.

 I.e.:  :         S_i+1 = S_i + X bps

		   D        D 
 So:             ------ = ------ + X
		 IPG_i+1    IPG_i

 Which result in:

			  C * IPG_i
	         IPG_i+1 = --------- 
			  C + IPG_i

		     D
 where:         C = ---
		     X

Decrease Algorithm:
- Multiplicative decrease by changing IPG is very straight forward...
- If IPG_i+1 = f * IPG_i, then S_i+1 = S_i / f.
- Example:
Further Refinements:
The authors added several refinements to make the RAP protocol more stable:
- Clustered losses detection
- Fine grain rate adaptation
Clustered Losses Detection:
- Fact: each packet loss will trigger RAP to decrease transmission rate
- However: "clustered