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authorMarcelo Leitner <mleitner@redhat.com>2014-11-04 17:15:08 -0200
committerDavid S. Miller <davem@davemloft.net>2014-11-05 16:59:49 -0500
commit1f37bf87aa7523d28e7e4c4f7bb5dba98faa3e00 (patch)
tree587ea2b8eaff79f04b9f6cadce4089c1dd765b32 /net
parent46d3802627f60be1c17659a44c8d2d7a5e247023 (diff)
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')
-rw-r--r--net/ipv4/tcp_input.c60
1 files changed, 31 insertions, 29 deletions
diff --git a/net/ipv4/tcp_input.c b/net/ipv4/tcp_input.c
index a12b455928e5..88fa2d160685 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)
{