Presentation on theme: "Slide Set 13: TCP. In this set.... TCP Connection Termination TCP State Transition Diagram Flow Control How does TCP control its sliding window ?"— Presentation transcript:
Slide Set 13: TCP
In this set.... TCP Connection Termination TCP State Transition Diagram Flow Control How does TCP control its sliding window ?
Connection Termination Note that after the server receives the FIN_WAIT_1, it may still have messages -- thus, connection not yet closed. DATA/ACK Active Close FIN_WAIT1 FIN_WAIT2 CLOSE_WAIT LAST_ACK CLOSED FIN, Seq Num = M ACK= M+1 FIN, Seq Num = N ACK N+1
TCP State Transitions Note: Retransmissions and Data Packet /ACK exchanges are not represented in the state transition diagram. Final Wait time needed to ensure that the ACK is not lost. Simultaneous Connection Inceptions/ Terminations possible !
An Simpler View of the Client Side CLOSED TIME_WAIT FIN_WAIT2 FIN_WAIT1 ESTABLISHED SYN_SENT SYN (Send) Rcv. SYN+ACK, Send ACK Send FIN Rcv. ACK, Send Nothing Rcv. FIN, Send ACK 120 secs
Simpler Server Model CLOSED LAST ACK CLOSE_WAIT ESTABLISHED SYN_RCVD LISTEN Passive OPEN, Create Listen socket Rcv. SYN, Send SYN+ACK RCV ACK Rcv. FIN, Send ACK Send FIN Rcv. ACK, Send nothing
More about Termination Applications on both sides have to “independently” close their half of the connection. If one side does it, this means that this side has no data to send but it is willing to receive. In the TIME_WAIT state, a client waits for 2 X MSL (typically). During this time the socket cannot be reused. – If ACK is lost, a new FIN may be forthcoming and this second FIN may be delayed. –Thus, if a new connection uses the same connection i.e., the same port numbers, this FIN would initiate termination of later connection !
Sequence Numbers and ACKs How does one set Sequence numbers ? – Implicitly a number in every byte in the stream. –If we have bytes and each segment = MSS and = 1000 bytes, SN of 1st segment = 0, SN of second segment = 1000 and so on. Note that ACK number is the number that the receiving host puts in -- indicates the “next” byte that it is expecting. ACKs are cumulative -- ACK up to all the bytes that are received.
Flow Control Flow control ensures that the sender does not send at a rate that causes the receiver buffer to overflow. Note that flow control is “end-to- end”.
Buffers at End Hosts Sending buffer – Maintains data sent but not ACKed –Data written by application but not sent. Receive buffer –Data that arrives out of order –Data that is in correct order but not yet read by application.
Sender Side View For now, let us forget SN wrap around. Three pointers are maintained, LastByteAcked, LastByteSent, LastByteWritten. LastByteAcked ≤ LastByteSent LastByteSent ≤ LastByteWritten Sending application LastByteWritten TCP LastByteSentLastByteAcked (a)
How is Flow Control done? Receiver “advertises” a window size to the sender based on the buffer size allocated for the connection. –Remember the “Advertised Window” field in the TCP header ? Sender cannot have more than “Advertised Window” bytes of unacknowledged data. Remember -- buffers are of finite size - i.e., there is a MaxRcvBuffer and MaxSendBuffer.
Setting the Advertised Window On the TCP receive side, clearly, LastByteRcvd -LastByteRead ≤ MaxRcvBuffer Thus, it advertises the space left in the buffer i.e., Advertised Window = MaxRcvBuffer - (LastByteRcvd -LastByteRead) As more data arrives i.e., more received bytes than read bytes, LastByteRcvd increases and hence, Advertised Window reduces.
Sender Side Response At the sender side, the TCP sender should ensure that: LastByteSent - LastByteAcked ≤ Advertised Window. Thus, we define what is called the “Effective Window” which limits the amount of data that TCP can send : Effective Window = Advertised Window - (LastByteSent - LastByteAcked) Note here that ACKing does not imply that the process has read the data! In order to prevent the overflow of the Send Side buffer: LastByteWritten - LastByteAcked ≤ MaxSendBuffer –If application tries to write more, TCP blocks.
Persistency What does one do when Advertised Window = 0 ? The sender will persist by sending 1 segment. Note that this segment may not be accepted by the receiver initially. But at some point, it would trigger a response that may contain a new Advertised window.
Sequence Number Wraparound TCP Sequence Number is 32 bits long. Advertised Window is 16 bits. Since 2 32 >> 2 X 2 16, it is almost impossible for the same sequence number to exist twice -- wrap around unlikely. In addition, MSL = 120 seconds to make sure that there is no wrap-around. Time-stamps may also be used.
How long should the time- out be ? Remember, TCP has to ensure reliability. So bytes need to be resent if there is no “timely” acknowledgement. How long should the sender wait ? It should be adaptive -- fluctuation in load on the network. – If too short, false time-outs – If too long, then poor rate of sending. Depends on round trip time estimation
RTT Estimation Simple mechanism could be: –Send packet, record time T1 –When ACK is returned, record time = T2. –T2 -T1 = Estimated RTT. To avoid fluctuations, estimated RTT is a weighted average of previous time and current sample Estimated RTT = (1- ) Estimated RTT + SampleRTT In the original specification = The Time out is set to 2 * RTT.
A problem When there are retransmissions, it is unclear if the ACK is for the original transmission or for a retransmission. How do we overcome this ?
The Karn Patridge Algorithm Take SampleRTT measurements only for segments that have been sent once ! This eliminates the possibility that wrong RTT estimates are factored into the estimation. Another change -- Each time TCP retransmits, it sets the next timeout to 2 X Last timeout --> This is called the Exponential Back-off (primarily for avoiding congestion).
Jacobson Karels Algorithm The main problem with the Karn/Patridge scheme is that it does not take into account the variation between RTT samples. New method proposed -- the Jacobson Karels Algorithm. Estimated RTT = Estimated RTT + X Difference – Difference = Sample RTT - Estimated RTT Deviation = Deviation + (|Difference| - deviation) Timeout = Estimated RTT + deviation. The values of and are computed based on experience -- Typically = 1 and = 4.
Silly Window Syndrome Suppose a MSS worth of data is collected and advertised window is MSS/2. What should the sender do ? -- transmit half full segments or wait to send a full MSS when window opens ? Early implementations were aggressive -- transmit MSS/2. Aggressively doing this, would consistently result in small segment sizes -- called the Silly Window Syndrome.
Issues.. We cannot eliminate the possibility of small segments being sent. However, we can introduce methods to coalesce small chunks. – Delaying ACKs -- receiver does not send ACKs as soon as it receives segments. How long to delay ? Not very clear. –Ultimate solution falls to the sender -- when should I transmit ?
Nagle’s Algorithm If sender waits too long --> bad for interactive connections. If it does not wait long enough -- silly window syndrome. How to solve ? Timer -- clock based – If both available data and Window ≥ MSS, send full segment. –Else, if there is unACKed data in flight, buffer new data until ACK returns. –Else, send new data now. Note -- Socket interface allows some applications to turn off Nagle’s algorithm by setting the TCP-NODELAY option.
TCP Throughput If a connection sends W segments of MSS size (in bytes) in RTT seconds, then, the throughput is defined as : W *MSS / RTT bytes/second. If there is a link of capacity R, if there are K connections, what we want is for each TCP connection to have a throughput = R/K.
Throughput (cont) If a TCP session goes through n links and if link j has a rate R j and is shared by K j connections, ideally the throughput = R j /K j. Thus, a connection’s end-to-end rate is r = min (R 1 /K 1, R 2 /K 2,.. R j /K j... R n /K n ). In reality not so simple, some connections may be unable to use their share -- so the share may be higher.
Where are we ? We have covered Chapter 5 -- Sections 5.1 and 5.2. Whatever I left out from Section 5.2 is for self-study.
Where are we headed ? We will look at Congestion Control with TCP next time. –Chapter 6 -- Sections 6.3 and 6.4.