This is a description of the modified TCP congestion avoidance
algorithm that I promised at the teleconference.

BTW, on re-reading, I noticed there were several errors in
Lixia's note besides the problem I noted at the teleconference.
I don't know whether that's because I mis-communicated the
algorithm at dinner (as I recall, I'd had some wine) or because
she's convinced that TCP is ultimately irrelevant :). Either
way, you will probably be disappointed if you experiment with
what's in that note.

First, I should point out once again that there are two
completely independent window adjustment algorithms running in
the sender: Slow-start is run when the pipe is empty (i.e.,
when first starting or re-starting after a timeout). Its goal
is to get the "ack clock" started so packets will be metered
into the network at a reasonable rate. The other algorithm,
congestion avoidance, is run any time *but* when (re-)starting
and is responsible for estimating the (dynamically varying)
pipesize. You will cause yourself, or me, no end of confusion
if you lump these separate algorithms (as Lixia's message did).

The modifications described here are only to the congestion
avoidance algorithm, not to slow-start, and they are intended to
apply to large bandwidth-delay product paths (though they don't
do any harm on other paths). Remember that with regular TCP (or
with slow-start/c-a TCP), throughput really starts to go to hell
when the probability of packet loss is on the order of the
bandwidth-delay product. E.g., you might expect a 1% packet
loss rate to translate into a 1% lower throughput but for, say,
a TCP connection with a 100 packet b-d p. (= window), it results
in a 50-75% throughput loss. To make TCP effective on fat
pipes, it would be nice if throughput degraded only as function
of loss probability rather than as the product of the loss
probabilty and the b-d p. (Assuming, of course, that we can do
this without sacrificing congestion avoidance.)

These mods do two things: (1) prevent the pipe from going empty
after a loss (if the pipe doesn't go empty, you won't have to
waste round-trip times re-filling it) and (2) correctly account
for the amount of data actually in the pipe (since that's what
congestion avoidance is supposed to be estimating and adapting to).

For (1), remember that we use a packet loss as a signal that the
pipe is overfull (congested) and that packet loss can be
detected one of two different ways: (a) via a retransmit
timeout or (b) when some small number (3-4) of consecutive
duplicate acks has been received (the "fast retransmit"
algorithm). In case (a), the pipe is guaranteed to be empty so
we must slow-start. In case (b), if the duplicate ack
threshhold is small compared to the bandwidth-delay product, we
will detect the loss with the pipe almost full. I.e., given a
threshhold of 3 packets and an LBL-MIT bandwidth-delay of around
24KB or 16 packets (assuming 1500 byte MTUs), the pipe is 75%
full when fast-retransmit detects a loss (actually, until
gateways start doing some sort of congestion control, the pipe
is overfull when the loss is detected so *at least* 75% of the
packets needed for ack clocking are in transit when
fast-retransmit happens). Since the pipe is full, there's no
need to slow-start after a fast-retransmit.

For (2), consider what a duplicate ack means: either the
network duplicated a packet (i.e., the NSFNet braindead IBM
token ring adapters) or the receiver got an out-of-order packet.
The usual cause of out-of-order packets at the receiver is a
missing packet. I.e., if there are W packets in transit and one
is dropped, the receiver will get W-1 out-of-order and
(4.3-tahoe TCP will) generate W-1 duplicate acks. If the
`consecutive duplicates' threshhold is set high enough, we can
reasonably assume that duplicate acks mean dropped packets.

But there's more information in the ack: The receiver can only
generate one in response to a packet arrival. I.e., a duplicate
ack means that a packet has left the network (it is now cached
at the receiver). If the sender is limitted by the congestion
window, a packet can now be sent. (The congestion window is a
count of how many packets will fit in the pipe. The ack says a
packet has left the pipe so a new one can be added to take its
place.) To put this another way, say the current congestion
window is C (i.e, C packets will fit in the pipe) and D
duplicate acks have been received. Then only C-D packets are
actually in the pipe and the sender wants to use a window of C+D
packets to fill the pipe to its estimated capacity (C+D sent -
D received = C in pipe).

So, conceptually, the slow-start/cong.avoid/fast-rexmit changes
are:

- The sender's input routine is changed to set `cwnd' to `ssthresh'
when the dup ack threshhold is reached. [It used to set cwnd to
mss to force a slow-start.] Everything else stays the same.

- The sender's output routine is changed to use an effective window
of min(snd_wnd, cwnd + dupacks*mss) [the change is the addition
of the `dupacks*mss' term.] `Dupacks' is zero until the rexmit
threshhold is reached and zero except when receiving a sequence
of duplicate acks.

The actual implementation is slightly different than the above
because I wanted to avoid the multiply in the output routine
(multiplies are expensive on some risc machines). A diff of the
old and new fastrexmit code is attached (your line numbers will
vary).

Note that we still do congestion avoidance (i.e., the window is
reduced by 50% when we detect the packet loss). But, as long as
the receiver's offered window is large enough (it needs to be at
most twice the bandwidth-delay product), we continue sending
packets (at exactly half the rate we were sending before the
loss) even after the loss is detected so the pipe stays full at
exactly the level we want and a slow-start isn't necessary.

Some algebra might make this last clear: Say U is the sequence
number of the first un-acked packet and we are using a window
size of W when packet U is dropped. Packets [U..U+W) are in
transit. When the loss is detected, we send packet U and pull
the window back to W/2. But in the round-trip time it takes
the U retransmit to fill the receiver's hole and an ack to get
back, W-1 dup acks will arrive (one for each packet in transit).
The window is effectively inflated by one packet for each of
these acks so packets [U..U+W/2+W-1) are sent. But we don't
re-send packets unless we know they've been lost so the amount
actually sent between the loss detection and the recovery ack is
U+W/2+W-1 - U+W = W/2-1 which is exactly the amount congestion
avoidance allows us to send (if we add in the rexmit of U). The
recovery ack is for packet U+W so when the effective window is
pulled back from W/2+W-1 to W/2 (which happens because the
recovery ack is `new' and sets dupack to zero), we are allowed
to send up to packet U+W+W/2 which is exactly the first packet
we haven't yet sent. (I.e., there is no sudden burst of packets
as the `hole' is filled.) Also, when sending packets between
the loss detection and the recovery ack, we do nothing for the
first W/2 dup acks (because they only allow us to send packets
we've already sent) and the bottleneck gateway is given W/2
packet times to clean out its backlog. Thus when we start
sending our W/2-1 new packets, the bottleneck queue is as empty
as it can be.

[I don't know if you can get the flavor of what happens from
this description -- it's hard to see without a picture. But I
was delighted by how beautifully it worked -- it was like
watching the innards of an engine when all the separate motions
of crank, pistons and valves suddenly fit together and
everything appears in exactly the right place at just the right
time.]

Also note that this algorithm interoperates with old tcp's: Most
pre-tahoe tcp's don't generate the dup acks on out-of-order packets.
If we don't get the dup acks, fast retransmit never fires and the
window is never inflated so everything happens in the old way (via
timeouts). Everything works just as it did without the new algorithm
(and just as slow).

If you want to simulate this, the intended environment is:

- large bandwidth-delay product (say 20 or more packets)

- receiver advertising window of two b-d p (or, equivalently,
advertised window of the unloaded b-d p but two or more
connections simultaneously sharing the path).

- average loss rate (from congestion or other source) less than
one lost packet per round-trip-time per active connection.
(The algorithm works at higher loss rate but the TCP selective
ack option has to be implemented otherwise the pipe will go empty
waiting to fill the second hole and throughput will once again
degrade at the product of the loss rate and b-d p. With selective
ack, throughput is insensitive to b-d p at any loss rate.)

And, of course, we should always remember that good engineering
practise suggests a b-d p worth of buffer at each bottleneck --
less buffer and your simulation will exhibit the interesting
pathologies of a poorly engineered network but will probably
tell you little about the workings of the algorithm (unless the
algorithm misbehaves badly under these conditions but my
simulations and measurements say that it doesn't). In these
days of $100/megabyte memory, I dearly hope that this particular
example of bad engineering is of historical interest only.

- Van

-----------------
*** /tmp/,RCSt1a26717 Mon Apr 30 01:35:17 1990
--- tcp_input.c Mon Apr 30 01:33:30 1990
***************
*** 834,850 ****
* Kludge snd_nxt & the congestion
* window so we send only this one
! * packet. If this packet fills the
! * only hole in the receiver's seq.
! * space, the next real ack will fully
! * open our window. This means we
! * have to do the usual slow-start to
! * not overwhelm an intermediate gateway
! * with a burst of packets. Leave
! * here with the congestion window set
! * to allow 2 packets on the next real
! * ack and the exp-to-linear thresh
! * set for half the current window
! * size (since we know we're losing at
! * the current window size).
*/
if (tp->t_timer[TCPT_REXMT] == 0 ||
--- 834,850 ----
* Kludge snd_nxt & the congestion
* window so we send only this one
! * packet.
! *
! * We know we're losing at the current
! * window size so do congestion avoidance
! * (set ssthresh to half the current window
! * and pull our congestion window back to
! * the new ssthresh).
! *
! * Dup acks mean that packets have left the
! * network (they're now cached at the receiver)
! * so bump cwnd by the amount in the receiver
! * to keep a constant cwnd packets in the
! * network.
*/
if (tp->t_timer[TCPT_REXMT] == 0 ||
***************
*** 853,864 ****
else if (++tp->t_dupacks == tcprexmtthresh) {
tcp_seq onxt = tp->snd_nxt;
! u_int win =
! MIN(tp->snd_wnd, tp->snd_cwnd) / 2 /
! tp->t_maxseg;

if (win < 2)
win = 2;
tp->snd_ssthresh = win * tp->t_maxseg;
-
tp->t_timer[TCPT_REXMT] = 0;
tp->t_rtt = 0;
--- 853,864 ----
else if (++tp->t_dupacks == tcprexmtthresh) {
tcp_seq onxt = tp->snd_nxt;
! u_int win = MIN(tp->snd_wnd,
! tp->snd_cwnd);

+ win /= tp->t_maxseg;
+ win >>= 1;
if (win < 2)
win = 2;
tp->snd_ssthresh = win * tp->t_maxseg;
tp->t_timer[TCPT_REXMT] = 0;
tp->t_rtt = 0;
***************
*** 866,873 ****
tp->snd_cwnd = tp->t_maxseg;
(void) tcp_output(tp);
!
if (SEQ_GT(onxt, tp->snd_nxt))
tp->snd_nxt = onxt;
goto drop;
}
} else
--- 866,879 ----
tp->snd_cwnd = tp->t_maxseg;
(void) tcp_output(tp);
! tp->snd_cwnd = tp->snd_ssthresh +
! tp->t_maxseg *
! tp->t_dupacks;
if (SEQ_GT(onxt, tp->snd_nxt))
tp->snd_nxt = onxt;
goto drop;
+ } else if (tp->t_dupacks > tcprexmtthresh) {
+ tp->snd_cwnd += tp->t_maxseg;
+ (void) tcp_output(tp);
+ goto drop;
}
} else
***************
*** 874,877 ****
--- 880,890 ----
tp->t_dupacks = 0;
break;
+ }
+ if (tp->t_dupacks) {
+ /*
+ * the congestion window was inflated to account for
+ * the other side's cached packets - retract it.
+ */
+ tp->snd_cwnd = tp->snd_ssthresh;
}
tp->t_dupacks = 0;
*** /tmp/,RCSt1a26725 Mon Apr 30 01:35:23 1990
--- tcp_timer.c Mon Apr 30 00:36:29 1990
***************
*** 223,226 ****
--- 223,227 ----
tp->snd_cwnd = tp->t_maxseg;
tp->snd_ssthresh = win * tp->t_maxseg;
+ tp->t_dupacks = 0;
}
(void) tcp_output(tp);

From van@helios.ee.lbl.gov Mon Apr 30 10:37:36 1990
To: end2end-interest@ISI.EDU
Subject: modified TCP congestion avoidance algorithm (correction)
Date: Mon, 30 Apr 90 10:36:12 PDT
From: Van Jacobson <van@helios.ee.lbl.gov>
Status: RO

I shouldn't make last minute 'fixes'. The code I sent out last
night had a small error:

*** t.c Mon Apr 30 10:28:52 1990
--- tcp_input.c Mon Apr 30 10:30:41 1990
***************
*** 885,893 ****
* the congestion window was inflated to account for
* the other side's cached packets - retract it.
*/
! tp->snd_cwnd = tp->snd_ssthresh;
}
- tp->t_dupacks = 0;
if (SEQ_GT(ti->ti_ack, tp->snd_max)) {
tcpstat.tcps_rcvacktoomuch++;
goto dropafterack;
--- 885,894 ----
* the congestion window was inflated to account for
* the other side's cached packets - retract it.
*/
! if (tp->snd_cwnd > tp->snd_ssthresh)
! tp->snd_cwnd = tp->snd_ssthresh;
! tp->t_dupacks = 0;
}
if (SEQ_GT(ti->ti_ack, tp->snd_max)) {
tcpstat.tcps_rcvacktoomuch++;
goto dropafterack;