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author | Marcelo Leitner <mleitner@redhat.com> | 2014-11-04 17:15:08 -0200 |
---|---|---|
committer | David S. Miller <davem@davemloft.net> | 2014-11-05 16:59:49 -0500 |
commit | 1f37bf87aa7523d28e7e4c4f7bb5dba98faa3e00 (patch) | |
tree | 587ea2b8eaff79f04b9f6cadce4089c1dd765b32 /net/ipv4/tcp_input.c | |
parent | 46d3802627f60be1c17659a44c8d2d7a5e247023 (diff) | |
download | op-kernel-dev-1f37bf87aa7523d28e7e4c4f7bb5dba98faa3e00.zip op-kernel-dev-1f37bf87aa7523d28e7e4c4f7bb5dba98faa3e00.tar.gz |
tcp: zero retrans_stamp if all retrans were acked
Ueki Kohei reported that when we are using NewReno with connections that
have a very low traffic, we may timeout the connection too early if a
second loss occurs after the first one was successfully acked but no
data was transfered later. Below is his description of it:
When SACK is disabled, and a socket suffers multiple separate TCP
retransmissions, that socket's ETIMEDOUT value is calculated from the
time of the *first* retransmission instead of the *latest*
retransmission.
This happens because the tcp_sock's retrans_stamp is set once then never
cleared.
Take the following connection:
Linux remote-machine
| |
send#1---->(*1)|--------> data#1 --------->|
| | |
RTO : :
| | |
---(*2)|----> data#1(retrans) ---->|
| (*3)|<---------- ACK <----------|
| | |
| : :
| : :
| : :
16 minutes (or more) :
| : :
| : :
| : :
| | |
send#2---->(*4)|--------> data#2 --------->|
| | |
RTO : :
| | |
---(*5)|----> data#2(retrans) ---->|
| | |
| | |
RTO*2 : :
| | |
| | |
ETIMEDOUT<----(*6)| |
(*1) One data packet sent.
(*2) Because no ACK packet is received, the packet is retransmitted.
(*3) The ACK packet is received. The transmitted packet is acknowledged.
At this point the first "retransmission event" has passed and been
recovered from. Any future retransmission is a completely new "event".
(*4) After 16 minutes (to correspond with retries2=15), a new data
packet is sent. Note: No data is transmitted between (*3) and (*4).
The socket's timeout SHOULD be calculated from this point in time, but
instead it's calculated from the prior "event" 16 minutes ago.
(*5) Because no ACK packet is received, the packet is retransmitted.
(*6) At the time of the 2nd retransmission, the socket returns
ETIMEDOUT.
Therefore, now we clear retrans_stamp as soon as all data during the
loss window is fully acked.
Reported-by: Ueki Kohei
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Marcelo Ricardo Leitner <mleitner@redhat.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Tested-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Diffstat (limited to 'net/ipv4/tcp_input.c')
-rw-r--r-- | net/ipv4/tcp_input.c | 60 |
1 files changed, 31 insertions, 29 deletions
diff --git a/net/ipv4/tcp_input.c b/net/ipv4/tcp_input.c index a12b455..88fa2d1 100644 --- a/net/ipv4/tcp_input.c +++ b/net/ipv4/tcp_input.c @@ -2315,6 +2315,35 @@ static inline bool tcp_packet_delayed(const struct tcp_sock *tp) /* Undo procedures. */ +/* We can clear retrans_stamp when there are no retransmissions in the + * window. It would seem that it is trivially available for us in + * tp->retrans_out, however, that kind of assumptions doesn't consider + * what will happen if errors occur when sending retransmission for the + * second time. ...It could the that such segment has only + * TCPCB_EVER_RETRANS set at the present time. It seems that checking + * the head skb is enough except for some reneging corner cases that + * are not worth the effort. + * + * Main reason for all this complexity is the fact that connection dying + * time now depends on the validity of the retrans_stamp, in particular, + * that successive retransmissions of a segment must not advance + * retrans_stamp under any conditions. + */ +static bool tcp_any_retrans_done(const struct sock *sk) +{ + const struct tcp_sock *tp = tcp_sk(sk); + struct sk_buff *skb; + + if (tp->retrans_out) + return true; + + skb = tcp_write_queue_head(sk); + if (unlikely(skb && TCP_SKB_CB(skb)->sacked & TCPCB_EVER_RETRANS)) + return true; + + return false; +} + #if FASTRETRANS_DEBUG > 1 static void DBGUNDO(struct sock *sk, const char *msg) { @@ -2410,6 +2439,8 @@ static bool tcp_try_undo_recovery(struct sock *sk) * is ACKed. For Reno it is MUST to prevent false * fast retransmits (RFC2582). SACK TCP is safe. */ tcp_moderate_cwnd(tp); + if (!tcp_any_retrans_done(sk)) + tp->retrans_stamp = 0; return true; } tcp_set_ca_state(sk, TCP_CA_Open); @@ -2430,35 +2461,6 @@ static bool tcp_try_undo_dsack(struct sock *sk) return false; } -/* We can clear retrans_stamp when there are no retransmissions in the - * window. It would seem that it is trivially available for us in - * tp->retrans_out, however, that kind of assumptions doesn't consider - * what will happen if errors occur when sending retransmission for the - * second time. ...It could the that such segment has only - * TCPCB_EVER_RETRANS set at the present time. It seems that checking - * the head skb is enough except for some reneging corner cases that - * are not worth the effort. - * - * Main reason for all this complexity is the fact that connection dying - * time now depends on the validity of the retrans_stamp, in particular, - * that successive retransmissions of a segment must not advance - * retrans_stamp under any conditions. - */ -static bool tcp_any_retrans_done(const struct sock *sk) -{ - const struct tcp_sock *tp = tcp_sk(sk); - struct sk_buff *skb; - - if (tp->retrans_out) - return true; - - skb = tcp_write_queue_head(sk); - if (unlikely(skb && TCP_SKB_CB(skb)->sacked & TCPCB_EVER_RETRANS)) - return true; - - return false; -} - /* Undo during loss recovery after partial ACK or using F-RTO. */ static bool tcp_try_undo_loss(struct sock *sk, bool frto_undo) { |