TCP provides a connection oriented, reliable, byte stream service. The term connection-oriented means the two applications using TCP must establish a TCP connection with each other before they can exchange data. It is a full duplex protocol, meaning that each TCP connection supports a pair of byte streams, one flowing in each direction. TCP includes a flow-control mechanism for each of these byte streams that allows the receiver to limit how much data the sender can transmit. TCP also implements a congestion-control mechanism.
|Two processes communicating via TCP sockets. Each side of a TCP connection has a socket which can be identified by the pair < IP_address, port_number >. Two processes communicating over TCP form a logical connection that is uniquely identifiable by the two sockets involved, that is by the combination < local_IP_address, local_port, remote_IP_address, remote_port>.|
TCP provides the following facilities to:
- Stream Data Transfer
- From the application's viewpoint, TCP transfers a contiguous stream of bytes. TCP does this by grouping the bytes in TCP segments, which are passed to IP for transmission to the destination. TCP itself decides how to segment the data and it may forward the data at its own convenience.
- Reliability TCP assigns a sequence number to each byte transmitted, and expects a positive acknowledgment (ACK) from the receiving TCP. If the ACK is not received within a timeout interval, the data is retransmitted. The receiving TCP uses the sequence numbers to rearrange the segments when they arrive out of order, and to eliminate duplicate segments.
- Flow Control The receiving TCP, when sending an ACK back to the sender, also indicates to the sender the number of bytes it can receive beyond the last received TCP segment, without causing overrun and overflow in its internal buffers. This is sent in the ACK in the form of the highest sequence number it can receive without problems.
- Multiplexing To allow for many processes within a single host to use TCP communication facilities simultaneously, the TCP provides a set of addresses or ports within each host. Concatenated with the network and host addresses from the internet communication layer, this forms a socket. A pair of sockets uniquely identifies each connection.
- Logical Connections
- The reliability and flow control mechanisms described above require that TCP initializes and maintains certain status information for each data stream. The combination of this status, including sockets, sequence numbers and window sizes, is called a logical connection. Each connection is uniquely identified by the pair of sockets used by the sending and receiving processes. Full Duplex TCP provides for concurrent data streams in both directions.
TCP data is encapsulated in an IP datagram. The figure shows the format of the TCP header. Its normal size is 20 bytes unless options are present. Each of the fields is discussed below:
- The SrcPort and DstPort fields identify the source and destination ports,respectively. These two fields plus the source and destination IP addresses, combine to uniquely identify each TCP connection.
- The sequence number identifies the byte in the stream of data from the sending TCP to the receiving TCP that the first byte of data in this segment represents.
- The Acknowledgement number field contains the next sequence number that the sender of the acknowledgement expects to receive. This is therefore the sequence number plus 1 of the last successfully received byte of data. This field is valid only if the ACK flag is on. Once a connection is established the Ack flag is always on.
- The Acknowledgement, SequenceNum, and AdvertisedWindow fields are all involved in TCP's sliding window algorithm.The Acknowledgement and AdvertisedW indow fields carry information about the flow of dat going in the other direction. In TCP's sliding window algorithm the reciever advertises a window size to the sender. This is done using the AdvertisedWindow field. The sender is then limited to having no more than a value of AdvertisedWindow bytes of un acknowledged data at any given time. The receiver sets a suitable value for the AdvertisedWindow based on the amount of memory allocated to the connection for the purpose of buffering data.
- The header length gives the length of the header in 32-bit words. This is required because the length of the options field is variable.
- The 6-bit Flags field is used to relay control information between TCP peers. The possible flags include SYN, FIN, RESET, PUSH, URG, and ACK.
- The SYN and Fin flags are used when establishing and terminating a TCP connection, respectively.
- The ACK flag is set any time the Acknowledgement field is valid, implying that the receiver should pay attention to it.
- The URG flag signifies that this segment contains urgent data. When this flag is set, the UrgPtr field indicates where the non-urgent data contained in this segment begins.
- The PUSH flag signifies that the sender invoked the push operation, whic h indicates to the receiving side of TCP that it should notify the receiving process of this fact.
- Finally, the RESET flag signifies that the receiver has become confused and so wants to abort the connection.
- The Checksum covers the TCP segment: the TCP header and the TCP data. This is a mandatory field that must be calculated by the sender, and then verified by the receiver.
- The Option field is the maximum segment size option, called the MSS. Each end of the connection normally specifies this option on the first segment exchanged. It specifies the maximum sized segment the sender wants to recieve.
- The data portion of the TCP segment is optional.
The two transitions leading to the ESTABLISHED state correspond to the opening of a connection, and the two transitions leading from the ESTABLISHED state are for the termination of a connection. The ESTABLISHED state is where data transfer can occur between the two ends in both the directions.
If a connection is in the LISTEN state and a SYN segment arrives, the connection makes a transition to the SYN_RCVD state and takes the action of replying with an ACK+SYN segment. The client does an active open which causes its end of the connection to send a SYN segment to the server and to move to the SYN_SENT state. The arrival of the SYN+ACK segment causes the client to mo ve to the ESTABLISHED state and to send an ack back to the server. When this ACK arrives the server finally moves to the ESTABLISHED state. In other words, we have just traced the THREE-WAY HANDSHAKE.
In the process of terminating a connection, the important thing to kee p in mind is that the application process on both sides of the connection must independently close its half of the connection. Thus, on any one side there are three combinations of transition that get a connection from the ESTABLISHED state to the CLOSED state:
- This side closes first:
ESTABLISHED -> FIN_WAIT_1-> FIN_WAIT_2 -> TIME_WAIT -> CLOSED.
- The other side closes first:
ESTABLISHED -> CLOSE_WAIT -> LAST_ACK -> CLOSED.
- Both sides close at the same time:ESTABLISHED -> FIN_WAIT_1-> CLOSING ->TIME_WAIT -> CLOSED.
The main thing to recognize about connection teardown is that a connection in the TIME_WAIT state cannot move to the CLOSED state until it has waited for two times the maximum amount of time an IP datagram might live in the Inter net. The reason for this is that while the local side of the connection has sent an ACK in response to the other side's FIN segment, it does not know that the ACK was successfully delivered. As a consequence this other side might re transmit its FIN segment, and this second FIN segment might be delayed in the network. If the connection were allowed to move directly to the CLOSED state, then another pair of application processes might come along and open the same connection, and the delayed FIN segment from the earlier incarnation of the connection would immediately initiate the termination of the later incarnation of that connection.
The sliding window serves several purposes:
(1) it guarantees the reliable delivery of data(2) it ensures that the data is delivered in order,(3) it enforces flow control between the sender and the receiver.
Reliable and ordered delivery
The sending and receiving sides of TCP interact in the following manner to implement reliable and ordered delivery:
Each byte has a sequence number.ACKs are cumulative.Sending side
- LastByteAcked <=LastByteSent
- LastByteSent <= LastByteWritten
- bytes between LastByteAcked and LastByteWritten must be buffered.
- LastByteRead < NextByteExpected
- NextByteExpected <= LastByteRcvd + 1
- bytes between NextByteRead and LastByteRcvd must be buffered.
Sender buffer size : MaxSendBufferReceive buffer size : MaxRcvBufferReceiving side
- LastByteRcvd - NextBytteRead <= MaxRcvBuffer
- AdvertisedWindow = MaxRcvBuffer - (LastByteRcvd - NextByteRead)
- LastByteSent - LastByteAcked <= AdvertisedWindow
- EffectiveWindow = AdvertisedWindow - (LastByteSent - LastByteAcked)
- LastByteWritten - LastByteAcked <= MaxSendBuffer
- Block sender if (LastByteWritten - LastByteAcked) + y > MaxSendBuffer
Always send ACK in response to an arriving data segment
Persist when AdvertisedWindow = 0
TCP guarantees reliable delivery and so it retransmits each segment if an ACK is not received in a certainperiod of time. TCP sets this timeout as a function of the RTT it expects between the two ends of the connection. Unfortunately, given the range of possible RTT's between any pair of hosts in the Internet, as well as the variation in RTT between the same two hosts over time, choosing an appropriate timeout value is not that easy. To address this problem, TCP uses an adaptive retransmission medchanism. We describe this mechanism and how it has evolved over time.
Measure SampleRTT for each segment/ACK pair
Compute weighted average of RTT
EstimatedRTT = a*EstimatedRTT + b*SampleRTT, where a+b = 1
a between 0.8 and 0.9
b between 0.1 and 0.2
Set timeout based on EstimatedRTT
TimeOut = 2 * EstimatedRTT
Do not sample RTT when retransmitting
Double timeout after each retransmission
New calculation for average RTT
Difference = SampleRTT - EstimatedRTT
EstimatedRTT = EstimatedRTT + ( d * Difference)
Deviation = Deviation + d ( |Difference| - Deviation)), where d is a fraction between 0 and 1
Consider variance when setting timeout value
Timeout = u * EstimatedRTT + q * Deviation, where u = 1 and q = 4
It operates by observing that the rate at which new packets should be injected into the network is the rate at which the acknowledgments are returned by the other end.
Slow start adds another window to the sender's TCP: the congestion window, called "cwnd". When a new connection is established with a host on another network, the congestion window is initialized to one segment (i.e., the segment size announced by the other end, or the default, typically 536 or 512). Each time an ACK is received, the congestion window is increased by one segment. The sender can transmit up to the minimum of the congestion window and the advertised window. The congestion window is flow control imposed by the sender, while the advertised window is flow control imposed by the receiver. The former is based on the sender's assessment of perceived network congestion; the latter is related to the amount of available buffer space at the receiver for this connection.
The sender starts by transmitting one segment and waiting for its ACK. When that ACK is received, the congestion window is incremented from one to two, and two segments can be sent. When each of those two segments is acknowledged, the congestion window is increased to four. This provides an exponential growth, although it is not exactly exponential because the receiver may delay its ACKs, typically sending one ACK for every two segments that it receives.
At some point the capacity of the internet can be reached, and an intermediate router will start discarding packets. This tells the sender that its congestion window has gotten too large.
Early implementations performed slow start only if the other end was on a different network. Current implementations always perform slow start.
Congestion can occur when data arrives on a big pipe (a fast LAN) and gets sent out a smaller pipe (a slower WAN). Congestion can also occur when multiple input streams arrive at a router whose output capacity is less than the sum of the inputs. Congestion avoidance is a way to deal with lost packets.
The assumption of the algorithm is that packet loss caused by damage is very small (much less than 1%), therefore the loss of a packet signals congestion somewhere in the network between the source and destination. There are two indications of packet loss: a timeout occurring and the receipt of duplicate ACKs.
Congestion avoidance and slow start are independent algorithms with different objectives. But when congestion occurs TCP must slow down its transmission rate of packets into the network, and then invoke slow start to get things going again. In practice they are implemented together.
Congestion avoidance and slow start require that two variables be maintained for each connection: a congestion window, cwnd, and a slow start threshold size, ssthresh. The combined algorithm operates as follows:
1. Initialization for a given connection sets cwnd to one segment and ssthresh to 65535 bytes.
2. The TCP output routine never sends more than the minimum of cwnd and the receiver's advertised window.
3. When congestion occurs (indicated by a timeout or the reception of duplicate ACKs), one-half of the current window size (the minimum of cwnd and the receiver's advertised window, but at least two segments) is saved in ssthresh. Additionally, if the congestion is indicated by a timeout, cwnd is set to one segment (i.e., slow start).
4. When new data is acknowledged by the other end, increase cwnd, but the way it increases depends on whether TCP is performing slow start or congestion avoidance.
If cwnd is less than or equal to ssthresh, TCP is in slow start; otherwise TCP is performing congestion avoidance. Slow start continues until TCP is halfway to where it was when congestion occurred (since it recorded half of the window size that caused the problem in step 2), and then congestion avoidance takes over.
Slow start has cwnd begin at one segment, and be incremented by one segment every time an ACK is received. As mentioned earlier, this opens the window exponentially: send one segment, then two, then four, and so on. Congestion avoidance dictates that cwnd be incremented by segsize*segsize/cwnd each time an ACK is received, where segsize is the segment size and cwnd is maintained in bytes. This is a linear growth of cwnd, compared to slow start's exponential growth. The increase in cwnd should be at most one segment each round-trip time (regardless how many ACKs are received in that RTT), whereas slow start increments cwnd by the number of ACKs received in a round-trip time.
TCP may generate an immediate acknowledgment (a duplicate ACK) when an out- of-order segment is received. This duplicate ACK should not be delayed. The purpose of this duplicate ACK is to let the other end know that a segment was received out of order, and to tell it what sequence number is expected.
Since TCP does not know whether a duplicate ACK is caused by a lost segment or just a reordering of segments, it waits for a small number of duplicate ACKs to be received. It is assumed that if there is just a reordering of the segments, there will be only one or two duplicate ACKs before the reordered segment is processed, which will then generate a new ACK. If three or more duplicate ACKs are received in a row, it is a strong indication that a segment has been lost. TCP then performs a retransmission of what appears to be the missing segment, without waiting for a retransmission timer to expire.
After fast retransmit sends what appears to be the missing segment, congestion avoidance, but not slow start is performed. This is the fast recovery algorithm. It is an improvement that allows high throughput under moderate congestion, especially for large windows.
The reason for not performing slow start in this case is that the receipt of the duplicate ACKs tells TCP more than just a packet has been lost. Since the receiver can only generate the duplicate ACK when another segment is received, that segment has left the network and is in the receiver's buffer. That is, there is still data flowing between the two ends, and TCP does not want to reduce the flow abruptly by going into slow start.
The fast retransmit and fast recovery algorithms are usually implemented together as follows.
1. When the third duplicate ACK in a row is received, set ssthresh to one-half the current congestion window, cwnd, but no less than two segments. Retransmit the missing segment. Set cwnd to ssthresh plus 3 times the segment size. This inflates the congestion window by the number of segments that have left the network and which the other end has cached .
2. Each time another duplicate ACK arrives, increment cwnd by the segment size. This inflates the congestion window for the additional segment that has left the network. Transmit a packet, if allowed by the new value of cwnd.
3. When the next ACK arrives that acknowledges new data, set cwnd to ssthresh (the value set in step 1). This ACK should be the acknowledgment of the retransmission from step 1, one round-trip time after the retransmission. Additionally, this ACK should acknowledge all the intermediate segments sent between the lost packet and the receipt of the first duplicate ACK. This step is congestion avoidance, since TCP is down to one-half the rate it was at when the packet was lost.