----------
- snd.una: the sequence number of the first byte we did not receive an ACK for
-- snd.nxt: the sequence number of the first byte after the last one we ever sent
+- snd.nxt: the sequence number of the first byte after the last packet we sent (due to retransmission, this may go backwards)
- snd.wnd: the number of bytes we have left in our (UTCP/application?) input buffer
+- snd.last: the sequence number of the last byte that was enqueued in the TCP stream (increases only monotonically)
- rcv.nxt: the sequence number of the first byte after the last one we passed up to the application
-- rcv.wnd: the number of bytes the receives has left in its input buffer (may be more/less than our send buffer size)
+- rcv.wnd: the number of bytes the receiver has left in its input buffer (may be more/less than our send buffer size)
- The only packets that do not have ACK set must either have SYN or RST set
- Only packets received with rcv.nxt <= hdr.seq <= rcv.nxt + rcv.wnd are valid, drop others.
RETRANSMIT TIMEOUT
------------------
-(See RFC 2988, 3366)
+(See RFC 6298.)
This timer is intended to catch the case where we didn't get an ACK from the peer.
In principle, the timeout should be slightly longer than the maximum latency along the path.
+- The rtrx timer is set whenever we send a packet that must be ACKed by the peer:
+ - when it contains data
+ - when SYN or FIN is set
+- The rtrx timer is reset when we receive a packet that advances snd.una.
+ - it is cleared when snd.una == snd.last
+ - otherwise the timeout is set to the value of utcp->rto
+- If the rtrx timer expires, retransmit at least one packet, multiply the timeout by two, and rearm the timeout.
-- The rtrx timeout is set whenever snd.nxt is advanced.
-- If the rtrx timeout expires, retransmit at least one packet, and re-set the timeout.
+The value of RTO is calculated according to the RFC. At the moment, no
+timestamps are added to packets. When the RTT timer is not set, start it when
+sending a packet. When the ACK arrives, stop the timer and use the time
+difference as a measured RTT value. Use the algorithm from RFC 6298 to update
+RTO.
STATES
------
RECEIVE PACKET
--------------
-- Drop invalid packets:
- - Invalid flags or state
- - ACK always set
- - hdr.seq not within our receive window
- - hdr.ack ahead of snd.nxt
-- Handle RST packets
-- Advance snd.una?
- - reset conn timer if so
- - remove ACKed data from send buffer
-- If snd.una == snd.nxt, clear rtrx and conn timer
-- Process state changes due to SYN
-- Send new data to application
-- Process state changes due to FIN
+1 Drop invalid packets:
+ a Invalid flags or state
+ b ACK always set
+ c hdr.seq not within our receive window
+ d hdr.ack ahead of snd.nxt or behind snd.una
+2 Handle RST packets
+3 Advance snd.una?
+ a reset conn timer if so
+ b check if our SYN or FIN has been acked
+ c check if any data been acked
+ - remove ACKed data from send buffer
+ - increase cwnd
+ d no advance? NewReno
+4 If snd.una == snd.nxt, clear rtrx and conn timer
+5 Process state changes due to SYN
+6 Send new data to application
+7 Process state changes due to FIN
+
+CONGESTION AVOIDANCE
+--------------------
+
+We want to send as much packets as possible that won't cause any packets to be
+dropped. So we should not send more than the available bandwidth, and not more
+in one go than buffers along the path can handle.
+
+To start, we use "self-clocking". We send one packet, and wait for an ACK
+before sending another packet. On a network with a finite bandwidth but zero
+delay (latency), this will send packets as efficiently as possible. We don't
+need any timers to control the outgoing packet rate, that's why we call this
+self-clocked. However, latency is non-zero, and this means a number of packets
+is always on the way between the sender and receiver. The amount of packets
+"inbetween" is in principle the bandwidth times the delay (bandwidth-delay
+product, or BDP).
+
+Delay is fairly easy to measure (equal to half the round-trip time of a packet,
+which in TCP is easily obtained from the SYN and SYNACK pair, or the ACK in
+response of a segment), however bandwidth is more difficult and might change
+more rapidly than the latency.
+
+Back to the "inbetween" packets: ideally we would like to fill the available
+inbetween space completely. It should be easy to see that in that case,
+self-clocking will still work as intended. Our estimate of the amount of
+packets in the inbetween space is called the congestion window (CWND). If we
+know the BDP, we can set the CWND to it, however if we don't know it, we can
+start with a small CWND and gradually increase it (for example, every time we
+receive an ACK, send the next 2 segments). At some point, we will start sending
+at a higher rate than the available bandwidth, in which case packets will
+inevitably be lost. We detect that because we do not receive an ACK for our
+data, and then we have to reduce the CWND (for example, by half).
+
+The trick is to choose an algorithm that best keeps the CWND to the effective
+BDP.
+
+A nice introduction is RFC 2001.
+
+snd.cwnd: size of the congestion window.
+snd.nxt - snd.una: number of unacknowledged bytes, = number of bytes in flight.
+snd.cwnd - (snd.nxt - snd.una): unused size of congestion window
projects / utcp / blobdiff