- 20 5月, 2013 1 次提交
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由 Yuchung Cheng 提交于
tcp_timeout_skb() was intended to trigger fast recovery on timeout, unfortunately in reality it often causes spurious retransmission storms during fast recovery. The particular sign is a fast retransmit over the highest sacked sequence (SND.FACK). Currently the RTO timer re-arming (as in RFC6298) offers a nice cushion to avoid spurious timeout: when SND.UNA advances the sender re-arms RTO and extends the timeout by icsk_rto. The sender does not offset the time elapsed since the packet at SND.UNA was sent. But if the next (DUP)ACK arrives later than ~RTTVAR and triggers tcp_fastretrans_alert(), then tcp_timeout_skb() will mark any packet sent before the icsk_rto interval lost, including one that's above the highest sacked sequence. Most likely a large part of scorebard will be marked. If most packets are not lost then the subsequent DUPACKs with new SACK blocks will cause the sender to continue to retransmit packets beyond SND.FACK spuriously. Even if only one packet is lost the sender may falsely retransmit almost the entire window. The situation becomes common in the world of bufferbloat: the RTT continues to grow as the queue builds up but RTTVAR remains small and close to the minimum 200ms. If a data packet is lost and the DUPACK triggered by the next data packet is slightly delayed, then a spurious retransmission storm forms. As the original comment on tcp_timeout_skb() suggests: the usefulness of this feature is questionable. It also wastes cycles walking the sack scoreboard and is actually harmful because of false recovery. It's time to remove this. Signed-off-by: NYuchung Cheng <ycheng@google.com> Acked-by: NEric Dumazet <edumazet@google.com> Acked-by: NNeal Cardwell <ncardwell@google.com> Acked-by: NNandita Dukkipati <nanditad@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 21 3月, 2013 2 次提交
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由 Yuchung Cheng 提交于
This patch implements F-RTO (foward RTO recovery): When the first retransmission after timeout is acknowledged, F-RTO sends new data instead of old data. If the next ACK acknowledges some never-retransmitted data, then the timeout was spurious and the congestion state is reverted. Otherwise if the next ACK selectively acknowledges the new data, then the timeout was genuine and the loss recovery continues. This idea applies to recurring timeouts as well. While F-RTO sends different data during timeout recovery, it does not (and should not) change the congestion control. The implementaion follows the three steps of SACK enhanced algorithm (section 3) in RFC5682. Step 1 is in tcp_enter_loss(). Step 2 and 3 are in tcp_process_loss(). The basic version is not supported because SACK enhanced version also works for non-SACK connections. The new implementation is functionally in parity with the old F-RTO implementation except the one case where it increases undo events: In addition to the RFC algorithm, a spurious timeout may be detected without sending data in step 2, as long as the SACK confirms not all the original data are dropped. When this happens, the sender will undo the cwnd and perhaps enter fast recovery instead. This additional check increases the F-RTO undo events by 5x compared to the prior implementation on Google Web servers, since the sender often does not have new data to send for HTTP. Note F-RTO may detect spurious timeout before Eifel with timestamps does so. Signed-off-by: NYuchung Cheng <ycheng@google.com> Acked-by: NEric Dumazet <edumazet@google.com> Acked-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Yuchung Cheng 提交于
The patch series refactor the F-RTO feature (RFC4138/5682). This is to simplify the loss recovery processing. Existing F-RTO was developed during the experimental stage (RFC4138) and has many experimental features. It takes a separate code path from the traditional timeout processing by overloading CA_Disorder instead of using CA_Loss state. This complicates CA_Disorder state handling because it's also used for handling dubious ACKs and undos. While the algorithm in the RFC does not change the congestion control, the implementation intercepts congestion control in various places (e.g., frto_cwnd in tcp_ack()). The new code implements newer F-RTO RFC5682 using CA_Loss processing path. F-RTO becomes a small extension in the timeout processing and interfaces with congestion control and Eifel undo modules. It lets congestion control (module) determines how many to send independently. F-RTO only chooses what to send in order to detect spurious retranmission. If timeout is found spurious it invokes existing Eifel undo algorithms like DSACK or TCP timestamp based detection. The first patch removes all F-RTO code except the sysctl_tcp_frto is left for the new implementation. Since CA_EVENT_FRTO is removed, TCP westwood now computes ssthresh on regular timeout CA_EVENT_LOSS event. Signed-off-by: NYuchung Cheng <ycheng@google.com> Acked-by: NNeal Cardwell <ncardwell@google.com> Acked-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 18 3月, 2013 1 次提交
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由 Christoph Paasch 提交于
TCPCT uses option-number 253, reserved for experimental use and should not be used in production environments. Further, TCPCT does not fully implement RFC 6013. As a nice side-effect, removing TCPCT increases TCP's performance for very short flows: Doing an apache-benchmark with -c 100 -n 100000, sending HTTP-requests for files of 1KB size. before this patch: average (among 7 runs) of 20845.5 Requests/Second after: average (among 7 runs) of 21403.6 Requests/Second Signed-off-by: NChristoph Paasch <christoph.paasch@uclouvain.be> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 12 3月, 2013 2 次提交
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由 Nandita Dukkipati 提交于
This is the second of the TLP patch series; it augments the basic TLP algorithm with a loss detection scheme. This patch implements a mechanism for loss detection when a Tail loss probe retransmission plugs a hole thereby masking packet loss from the sender. The loss detection algorithm relies on counting TLP dupacks as outlined in Sec. 3 of: http://tools.ietf.org/html/draft-dukkipati-tcpm-tcp-loss-probe-01 The basic idea is: Sender keeps track of TLP "episode" upon retransmission of a TLP packet. An episode ends when the sender receives an ACK above the SND.NXT (tracked by tlp_high_seq) at the time of the episode. We want to make sure that before the episode ends the sender receives a "TLP dupack", indicating that the TLP retransmission was unnecessary, so there was no loss/hole that needed plugging. If the sender gets no TLP dupack before the end of the episode, then it reduces ssthresh and the congestion window, because the TLP packet arriving at the receiver probably plugged a hole. Signed-off-by: NNandita Dukkipati <nanditad@google.com> Acked-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Nandita Dukkipati 提交于
This patch series implement the Tail loss probe (TLP) algorithm described in http://tools.ietf.org/html/draft-dukkipati-tcpm-tcp-loss-probe-01. The first patch implements the basic algorithm. TLP's goal is to reduce tail latency of short transactions. It achieves this by converting retransmission timeouts (RTOs) occuring due to tail losses (losses at end of transactions) into fast recovery. TLP transmits one packet in two round-trips when a connection is in Open state and isn't receiving any ACKs. The transmitted packet, aka loss probe, can be either new or a retransmission. When there is tail loss, the ACK from a loss probe triggers FACK/early-retransmit based fast recovery, thus avoiding a costly RTO. In the absence of loss, there is no change in the connection state. PTO stands for probe timeout. It is a timer event indicating that an ACK is overdue and triggers a loss probe packet. The PTO value is set to max(2*SRTT, 10ms) and is adjusted to account for delayed ACK timer when there is only one oustanding packet. TLP Algorithm On transmission of new data in Open state: -> packets_out > 1: schedule PTO in max(2*SRTT, 10ms). -> packets_out == 1: schedule PTO in max(2*RTT, 1.5*RTT + 200ms) -> PTO = min(PTO, RTO) Conditions for scheduling PTO: -> Connection is in Open state. -> Connection is either cwnd limited or no new data to send. -> Number of probes per tail loss episode is limited to one. -> Connection is SACK enabled. When PTO fires: new_segment_exists: -> transmit new segment. -> packets_out++. cwnd remains same. no_new_packet: -> retransmit the last segment. Its ACK triggers FACK or early retransmit based recovery. ACK path: -> rearm RTO at start of ACK processing. -> reschedule PTO if need be. In addition, the patch includes a small variation to the Early Retransmit (ER) algorithm, such that ER and TLP together can in principle recover any N-degree of tail loss through fast recovery. TLP is controlled by the same sysctl as ER, tcp_early_retrans sysctl. tcp_early_retrans==0; disables TLP and ER. ==1; enables RFC5827 ER. ==2; delayed ER. ==3; TLP and delayed ER. [DEFAULT] ==4; TLP only. The TLP patch series have been extensively tested on Google Web servers. It is most effective for short Web trasactions, where it reduced RTOs by 15% and improved HTTP response time (average by 6%, 99th percentile by 10%). The transmitted probes account for <0.5% of the overall transmissions. Signed-off-by: NNandita Dukkipati <nanditad@google.com> Acked-by: NNeal Cardwell <ncardwell@google.com> Acked-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 11 3月, 2013 1 次提交
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由 Christoph Paasch 提交于
tw_cookie_values is never used in the TCP-stack. It was added by 435cf559 (TCPCT part 1d: define TCP cookie option, extend existing struct's), but already at that time it was not used at all, nor mentioned in the commit-message. Signed-off-by: NChristoph Paasch <christoph.paasch@uclouvain.be> Acked-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 14 2月, 2013 1 次提交
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由 Andrey Vagin 提交于
This functionality is used for restoring tcp sockets. A tcp timestamp depends on how long a system has been running, so it's differ for each host. The solution is to set a per-socket offset. A per-socket offset for a TIME_WAIT socket is inherited from a proper tcp socket. tcp_request_sock doesn't have a timestamp offset, because the repair mode for them are not implemented. Cc: "David S. Miller" <davem@davemloft.net> Cc: Alexey Kuznetsov <kuznet@ms2.inr.ac.ru> Cc: James Morris <jmorris@namei.org> Cc: Hideaki YOSHIFUJI <yoshfuji@linux-ipv6.org> Cc: Patrick McHardy <kaber@trash.net> Cc: Eric Dumazet <edumazet@google.com> Cc: Pavel Emelyanov <xemul@parallels.com> Signed-off-by: NAndrey Vagin <avagin@openvz.org> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 06 2月, 2013 1 次提交
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由 Stephen Hemminger 提交于
TCP Appropriate Byte Count was added by me, but later disabled. There is no point in maintaining it since it is a potential source of bugs and Linux already implements other better window protection heuristics. Signed-off-by: NStephen Hemminger <stephen@networkplumber.org> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 09 12月, 2012 1 次提交
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由 Joseph Gasparakis 提交于
This patch adds support in the kernel for offloading in the NIC Tx and Rx checksumming for encapsulated packets (such as VXLAN and IP GRE). For Tx encapsulation offload, the driver will need to set the right bits in netdev->hw_enc_features. The protocol driver will have to set the skb->encapsulation bit and populate the inner headers, so the NIC driver will use those inner headers to calculate the csum in hardware. For Rx encapsulation offload, the driver will need to set again the skb->encapsulation flag and the skb->ip_csum to CHECKSUM_UNNECESSARY. In that case the protocol driver should push the decapsulated packet up to the stack, again with CHECKSUM_UNNECESSARY. In ether case, the protocol driver should set the skb->encapsulation flag back to zero. Finally the protocol driver should have NETIF_F_RXCSUM flag set in its features. Signed-off-by: NJoseph Gasparakis <joseph.gasparakis@intel.com> Signed-off-by: NPeter P Waskiewicz Jr <peter.p.waskiewicz.jr@intel.com> Signed-off-by: NAlexander Duyck <alexander.h.duyck@intel.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 23 10月, 2012 1 次提交
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由 Yuchung Cheng 提交于
Add a bit TCPI_OPT_SYN_DATA (32) to the socket option TCP_INFO:tcpi_options. It's set if the data in SYN (sent or received) is acked by SYN-ACK. Server or client application can use this information to check Fast Open success rate. Signed-off-by: NYuchung Cheng <ycheng@google.com> Acked-by: NNeal Cardwell <ncardwell@google.com> Acked-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 13 10月, 2012 1 次提交
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由 David Howells 提交于
Signed-off-by: NDavid Howells <dhowells@redhat.com> Acked-by: NArnd Bergmann <arnd@arndb.de> Acked-by: NThomas Gleixner <tglx@linutronix.de> Acked-by: NMichael Kerrisk <mtk.manpages@gmail.com> Acked-by: NPaul E. McKenney <paulmck@linux.vnet.ibm.com> Acked-by: NDave Jones <davej@redhat.com>
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- 21 9月, 2012 1 次提交
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由 Christoph Paasch 提交于
Signed-off-by: NChristoph Paasch <christoph.paasch@uclouvain.be> Acked-by: NH.K. Jerry Chu <hkchu@google.com> Acked-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 01 9月, 2012 1 次提交
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由 Jerry Chu 提交于
This patch adds all the necessary data structure and support functions to implement TFO server side. It also documents a number of flags for the sysctl_tcp_fastopen knob, and adds a few Linux extension MIBs. In addition, it includes the following: 1. a new TCP_FASTOPEN socket option an application must call to supply a max backlog allowed in order to enable TFO on its listener. 2. A number of key data structures: "fastopen_rsk" in tcp_sock - for a big socket to access its request_sock for retransmission and ack processing purpose. It is non-NULL iff 3WHS not completed. "fastopenq" in request_sock_queue - points to a per Fast Open listener data structure "fastopen_queue" to keep track of qlen (# of outstanding Fast Open requests) and max_qlen, among other things. "listener" in tcp_request_sock - to point to the original listener for book-keeping purpose, i.e., to maintain qlen against max_qlen as part of defense against IP spoofing attack. 3. various data structure and functions, many in tcp_fastopen.c, to support server side Fast Open cookie operations, including /proc/sys/net/ipv4/tcp_fastopen_key to allow manual rekeying. Signed-off-by: NH.K. Jerry Chu <hkchu@google.com> Cc: Yuchung Cheng <ycheng@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Eric Dumazet <edumazet@google.com> Cc: Tom Herbert <therbert@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 23 7月, 2012 1 次提交
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由 Eric Dumazet 提交于
ICMP messages generated in output path if frame length is bigger than mtu are actually lost because socket is owned by user (doing the xmit) One example is the ipgre_tunnel_xmit() calling icmp_send(skb, ICMP_DEST_UNREACH, ICMP_FRAG_NEEDED, htonl(mtu)); We had a similar case fixed in commit a34a101e (ipv6: disable GSO on sockets hitting dst_allfrag). Problem of such fix is that it relied on retransmit timers, so short tcp sessions paid a too big latency increase price. This patch uses the tcp_release_cb() infrastructure so that MTU reduction messages (ICMP messages) are not lost, and no extra delay is added in TCP transmits. Reported-by: NMaciej Żenczykowski <maze@google.com> Diagnosed-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NEric Dumazet <edumazet@google.com> Cc: Nandita Dukkipati <nanditad@google.com> Cc: Tom Herbert <therbert@google.com> Cc: Tore Anderson <tore@fud.no> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 21 7月, 2012 1 次提交
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由 Eric Dumazet 提交于
Modern TCP stack highly depends on tcp_write_timer() having a small latency, but current implementation doesn't exactly meet the expectations. When a timer fires but finds the socket is owned by the user, it rearms itself for an additional delay hoping next run will be more successful. tcp_write_timer() for example uses a 50ms delay for next try, and it defeats many attempts to get predictable TCP behavior in term of latencies. Use the recently introduced tcp_release_cb(), so that the user owning the socket will call various handlers right before socket release. This will permit us to post a followup patch to address the tcp_tso_should_defer() syndrome (some deferred packets have to wait RTO timer to be transmitted, while cwnd should allow us to send them sooner) Signed-off-by: NEric Dumazet <edumazet@google.com> Cc: Tom Herbert <therbert@google.com> Cc: Yuchung Cheng <ycheng@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Nandita Dukkipati <nanditad@google.com> Cc: H.K. Jerry Chu <hkchu@google.com> Cc: John Heffner <johnwheffner@gmail.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 20 7月, 2012 3 次提交
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由 Yuchung Cheng 提交于
In trusted networks, e.g., intranet, data-center, the client does not need to use Fast Open cookie to mitigate DoS attacks. In cookie-less mode, sendmsg() with MSG_FASTOPEN flag will send SYN-data regardless of cookie availability. Signed-off-by: NYuchung Cheng <ycheng@google.com> Acked-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Yuchung Cheng 提交于
This patch implements sending SYN-data in tcp_connect(). The data is from tcp_sendmsg() with flag MSG_FASTOPEN (implemented in a later patch). The length of the cookie in tcp_fastopen_req, init'd to 0, controls the type of the SYN. If the cookie is not cached (len==0), the host sends data-less SYN with Fast Open cookie request option to solicit a cookie from the remote. If cookie is not available (len > 0), the host sends a SYN-data with Fast Open cookie option. If cookie length is negative, the SYN will not include any Fast Open option (for fall back operations). To deal with middleboxes that may drop SYN with data or experimental TCP option, the SYN-data is only sent once. SYN retransmits do not include data or Fast Open options. The connection will fall back to regular TCP handshake. Signed-off-by: NYuchung Cheng <ycheng@google.com> Acked-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Yuchung Cheng 提交于
This patch impelements the common code for both the client and server. 1. TCP Fast Open option processing. Since Fast Open does not have an option number assigned by IANA yet, it shares the experiment option code 254 by implementing draft-ietf-tcpm-experimental-options with a 16 bits magic number 0xF989. This enables global experiments without clashing the scarce(2) experimental options available for TCP. When the draft status becomes standard (maybe), the client should switch to the new option number assigned while the server supports both numbers for transistion. 2. The new sysctl tcp_fastopen 3. A place holder init function Signed-off-by: NYuchung Cheng <ycheng@google.com> Acked-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 12 7月, 2012 1 次提交
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由 Eric Dumazet 提交于
This introduce TSQ (TCP Small Queues) TSQ goal is to reduce number of TCP packets in xmit queues (qdisc & device queues), to reduce RTT and cwnd bias, part of the bufferbloat problem. sk->sk_wmem_alloc not allowed to grow above a given limit, allowing no more than ~128KB [1] per tcp socket in qdisc/dev layers at a given time. TSO packets are sized/capped to half the limit, so that we have two TSO packets in flight, allowing better bandwidth use. As a side effect, setting the limit to 40000 automatically reduces the standard gso max limit (65536) to 40000/2 : It can help to reduce latencies of high prio packets, having smaller TSO packets. This means we divert sock_wfree() to a tcp_wfree() handler, to queue/send following frames when skb_orphan() [2] is called for the already queued skbs. Results on my dev machines (tg3/ixgbe nics) are really impressive, using standard pfifo_fast, and with or without TSO/GSO. Without reduction of nominal bandwidth, we have reduction of buffering per bulk sender : < 1ms on Gbit (instead of 50ms with TSO) < 8ms on 100Mbit (instead of 132 ms) I no longer have 4 MBytes backlogged in qdisc by a single netperf session, and both side socket autotuning no longer use 4 Mbytes. As skb destructor cannot restart xmit itself ( as qdisc lock might be taken at this point ), we delegate the work to a tasklet. We use one tasklest per cpu for performance reasons. If tasklet finds a socket owned by the user, it sets TSQ_OWNED flag. This flag is tested in a new protocol method called from release_sock(), to eventually send new segments. [1] New /proc/sys/net/ipv4/tcp_limit_output_bytes tunable [2] skb_orphan() is usually called at TX completion time, but some drivers call it in their start_xmit() handler. These drivers should at least use BQL, or else a single TCP session can still fill the whole NIC TX ring, since TSQ will have no effect. Signed-off-by: NEric Dumazet <edumazet@google.com> Cc: Dave Taht <dave.taht@bufferbloat.net> Cc: Tom Herbert <therbert@google.com> Cc: Matt Mathis <mattmathis@google.com> Cc: Yuchung Cheng <ycheng@google.com> Cc: Nandita Dukkipati <nanditad@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 11 7月, 2012 1 次提交
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由 David S. Miller 提交于
No longer used. Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 10 6月, 2012 2 次提交
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由 Paul Pluzhnikov 提交于
I originally sent this patch to <trivial@kernel.org>, but Jiri Kosina did not feel that this is fully appropriate for the trivial tree. Using linux/tcp.h from C++ results in: cat t.cc #include <linux/tcp.h> int main() { } g++ -c t.cc In file included from t.cc:1: /usr/include/linux/tcp.h:72: error: '__u32 __fswab32(__u32)' cannot appear in a constant-expression /usr/include/linux/tcp.h:72: error: a function call cannot appear in a constant-expression ... Attached trivial patch fixes this problem. Tested: - the t.cc above compiles with g++ and - the following program generates the same output before/after the patch: #include <linux/tcp.h> #include <stdio.h> int main () { #define P(a) printf("%s: %08x\n", #a, (int)a) P(TCP_FLAG_CWR); P(TCP_FLAG_ECE); P(TCP_FLAG_URG); P(TCP_FLAG_ACK); P(TCP_FLAG_PSH); P(TCP_FLAG_RST); P(TCP_FLAG_SYN); P(TCP_FLAG_FIN); P(TCP_RESERVED_BITS); P(TCP_DATA_OFFSET); #undef P return 0; } Signed-off-by: NPaul Pluzhnikov <ppluzhnikov@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 David S. Miller 提交于
Since it's guarenteed that we will access the inetpeer if we're trying to do timewait recycling and TCP options were enabled on the connection, just cache the peer in the timewait socket. In the future, inetpeer lookups will be context dependent (per routing realm), and this helps facilitate that as well. Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 09 5月, 2012 1 次提交
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由 Kyle McMartin 提交于
Noticed this comment didn't get updated when tcp_build_and_update_options was refactored. Signed-off-by: NKyle McMartin <kyle@redhat.com> Signed-off-by: NJiri Kosina <jkosina@suse.cz>
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- 03 5月, 2012 2 次提交
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由 Yuchung Cheng 提交于
Implementing the advanced early retransmit (sysctl_tcp_early_retrans==2). Delays the fast retransmit by an interval of RTT/4. We borrow the RTO timer to implement the delay. If we receive another ACK or send a new packet, the timer is cancelled and restored to original RTO value offset by time elapsed. When the delayed-ER timer fires, we enter fast recovery and perform fast retransmit. Signed-off-by: NYuchung Cheng <ycheng@google.com> Acked-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Yuchung Cheng 提交于
This patch implements RFC 5827 early retransmit (ER) for TCP. It reduces DUPACK threshold (dupthresh) if outstanding packets are less than 4 to recover losses by fast recovery instead of timeout. While the algorithm is simple, small but frequent network reordering makes this feature dangerous: the connection repeatedly enter false recovery and degrade performance. Therefore we implement a mitigation suggested in the appendix of the RFC that delays entering fast recovery by a small interval, i.e., RTT/4. Currently ER is conservative and is disabled for the rest of the connection after the first reordering event. A large scale web server experiment on the performance impact of ER is summarized in section 6 of the paper "Proportional Rate Reduction for TCP”, IMC 2011. http://conferences.sigcomm.org/imc/2011/docs/p155.pdf Note that Linux has a similar feature called THIN_DUPACK. The differences are THIN_DUPACK do not mitigate reorderings and is only used after slow start. Currently ER is disabled if THIN_DUPACK is enabled. I would be happy to merge THIN_DUPACK feature with ER if people think it's a good idea. ER is enabled by sysctl_tcp_early_retrans: 0: Disables ER 1: Reduce dupthresh to packets_out - 1 when outstanding packets < 4. 2: (Default) reduce dupthresh like mode 1. In addition, delay entering fast recovery by RTT/4. Note: mode 2 is implemented in the third part of this patch series. Signed-off-by: NYuchung Cheng <ycheng@google.com> Acked-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 26 4月, 2012 1 次提交
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由 Pavel Emelyanov 提交于
Don't pick __u8/__u16 values directly from raw pointers, but instead use an array of structures of code:value pairs. This is OK, since the buffer we take options from is not an skb memory, but a user-to-kernel one. For those options which don't require any value now, require this to be zero (for potential future extension of this API). v2: Changed tcp_repair_opt to use two __u32-s as spotted by David Laight. Signed-off-by: NPavel Emelyanov <xemul@parallels.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 22 4月, 2012 2 次提交
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由 Pavel Emelyanov 提交于
There are options, which are set up on a socket while performing TCP handshake. Need to resurrect them on a socket while repairing. A new sockoption accepts a buffer and parses it. The buffer should be CODE:VALUE sequence of bytes, where CODE is standard option code and VALUE is the respective value. Only 4 options should be handled on repaired socket. To read 3 out of 4 of these options the TCP_INFO sockoption can be used. An ability to get the last one (the mss_clamp) was added by the previous patch. Now the restore. Three of these options -- timestamp_ok, mss_clamp and snd_wscale -- are just restored on a coket. The sack_ok flags has 2 issues. First, whether or not to do sacks at all. This flag is just read and set back. No other sack info is saved or restored, since according to the standart and the code dropping all sack-ed segments is OK, the sender will resubmit them again, so after the repair we will probably experience a pause in connection. Next, the fack bit. It's just set back on a socket if the respective sysctl is set. No collected stats about packets flow is preserved. As far as I see (plz, correct me if I'm wrong) the fack-based congestion algorithm survives dropping all of the stats and repairs itself eventually, probably losing the performance for that period. Signed-off-by: NPavel Emelyanov <xemul@openvz.org> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Pavel Emelyanov 提交于
This includes (according the the previous description): * TCP_REPAIR sockoption This one just puts the socket in/out of the repair mode. Allowed for CAP_NET_ADMIN and for closed/establised sockets only. When repair mode is turned off and the socket happens to be in the established state the window probe is sent to the peer to 'unlock' the connection. * TCP_REPAIR_QUEUE sockoption This one sets the queue which we're about to repair. The 'no-queue' is set by default. * TCP_QUEUE_SEQ socoption Sets the write_seq/rcv_nxt of a selected repaired queue. Allowed for TCP_CLOSE-d sockets only. When the socket changes its state the other seq-s are changed by the kernel according to the protocol rules (most of the existing code is actually reused). * Ability to forcibly bind a socket to a port The sk->sk_reuse is set to SK_FORCE_REUSE. * Immediate connect modification The connect syscall initializes the connection, then directly jumps to the code which finalizes it. * Silent close modification The close just aborts the connection (similar to SO_LINGER with 0 time) but without sending any FIN/RST-s to peer. Signed-off-by: NPavel Emelyanov <xemul@parallels.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 29 2月, 2012 1 次提交
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由 Neal Cardwell 提交于
There was an off-by-one error in the comments describing the highest_sack field in struct tcp_sock. The comments previously claimed that it was the "start sequence of the highest skb with SACKed bit". This commit fixes the comments to note that it is the "start sequence of the skb just *after* the highest skb with SACKed bit". Signed-off-by: NNeal Cardwell <ncardwell@google.com> Acked-by: NIlpo Järvinen <ilpo.jarvinen@helsinki.fi> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 01 2月, 2012 2 次提交
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由 Eric Dumazet 提交于
This patch makes sure we use appropriate memory barriers before publishing tp->md5sig_info, allowing tcp_md5_do_lookup() being used from tcp_v4_send_reset() without holding socket lock (upcoming patch from Shawn Lu) Note we also need to respect rcu grace period before its freeing, since we can free socket without this grace period thanks to SLAB_DESTROY_BY_RCU Signed-off-by: NEric Dumazet <eric.dumazet@gmail.com> Cc: Shawn Lu <shawn.lu@ericsson.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Eric Dumazet 提交于
In order to be able to support proper RST messages for TCP MD5 flows, we need to allow access to MD5 keys without locking listener socket. This conversion is a nice cleanup, and shrinks size of timewait sockets by 80 bytes. IPv6 code reuses generic code found in IPv4 instead of duplicating it. Control path uses GFP_KERNEL allocations instead of GFP_ATOMIC. Signed-off-by: NEric Dumazet <eric.dumazet@gmail.com> Cc: Shawn Lu <shawn.lu@ericsson.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 21 12月, 2011 1 次提交
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由 Vijay Subramanian 提交于
to record the state of SACK/FACK and DSACK for better readability and maintenance. Signed-off-by: NVijay Subramanian <subramanian.vijay@gmail.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 04 10月, 2011 1 次提交
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由 Eric Dumazet 提交于
Allows ss command (iproute2) to display "ecnseen" if at least one packet with ECT(0) or ECT(1) or ECN was received by this socket. "ecn" means ECN was negotiated at session establishment (TCP level) "ecnseen" means we received at least one packet with ECT fields set (IP level) ss -i ... ESTAB 0 0 192.168.20.110:22 192.168.20.144:38016 ino:5950 sk:f178e400 mem:(r0,w0,f0,t0) ts sack ecn ecnseen bic wscale:7,8 rto:210 rtt:12.5/7.5 cwnd:10 send 9.3Mbps rcv_space:14480 Signed-off-by: NEric Dumazet <eric.dumazet@gmail.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 25 8月, 2011 1 次提交
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由 Nandita Dukkipati 提交于
This patch implements Proportional Rate Reduction (PRR) for TCP. PRR is an algorithm that determines TCP's sending rate in fast recovery. PRR avoids excessive window reductions and aims for the actual congestion window size at the end of recovery to be as close as possible to the window determined by the congestion control algorithm. PRR also improves accuracy of the amount of data sent during loss recovery. The patch implements the recommended flavor of PRR called PRR-SSRB (Proportional rate reduction with slow start reduction bound) and replaces the existing rate halving algorithm. PRR improves upon the existing Linux fast recovery under a number of conditions including: 1) burst losses where the losses implicitly reduce the amount of outstanding data (pipe) below the ssthresh value selected by the congestion control algorithm and, 2) losses near the end of short flows where application runs out of data to send. As an example, with the existing rate halving implementation a single loss event can cause a connection carrying short Web transactions to go into the slow start mode after the recovery. This is because during recovery Linux pulls the congestion window down to packets_in_flight+1 on every ACK. A short Web response often runs out of new data to send and its pipe reduces to zero by the end of recovery when all its packets are drained from the network. Subsequent HTTP responses using the same connection will have to slow start to raise cwnd to ssthresh. PRR on the other hand aims for the cwnd to be as close as possible to ssthresh by the end of recovery. A description of PRR and a discussion of its performance can be found at the following links: - IETF Draft: http://tools.ietf.org/html/draft-mathis-tcpm-proportional-rate-reduction-01 - IETF Slides: http://www.ietf.org/proceedings/80/slides/tcpm-6.pdf http://tools.ietf.org/agenda/81/slides/tcpm-2.pdf - Paper to appear in Internet Measurements Conference (IMC) 2011: Improving TCP Loss Recovery Nandita Dukkipati, Matt Mathis, Yuchung Cheng Signed-off-by: NNandita Dukkipati <nanditad@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 09 6月, 2011 1 次提交
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由 Jerry Chu 提交于
This patch lowers the default initRTO from 3secs to 1sec per RFC2988bis. It falls back to 3secs if the SYN or SYN-ACK packet has been retransmitted, AND the TCP timestamp option is not on. It also adds support to take RTT sample during 3WHS on the passive open side, just like its active open counterpart, and uses it, if valid, to seed the initRTO for the data transmission phase. The patch also resets ssthresh to its initial default at the beginning of the data transmission phase, and reduces cwnd to 1 if there has been MORE THAN ONE retransmission during 3WHS per RFC5681. Signed-off-by: NH.K. Jerry Chu <hkchu@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 31 8月, 2010 1 次提交
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由 Jerry Chu 提交于
This patch provides a "user timeout" support as described in RFC793. The socket option is also needed for the the local half of RFC5482 "TCP User Timeout Option". TCP_USER_TIMEOUT is a TCP level socket option that takes an unsigned int, when > 0, to specify the maximum amount of time in ms that transmitted data may remain unacknowledged before TCP will forcefully close the corresponding connection and return ETIMEDOUT to the application. If 0 is given, TCP will continue to use the system default. Increasing the user timeouts allows a TCP connection to survive extended periods without end-to-end connectivity. Decreasing the user timeouts allows applications to "fail fast" if so desired. Otherwise it may take upto 20 minutes with the current system defaults in a normal WAN environment. The socket option can be made during any state of a TCP connection, but is only effective during the synchronized states of a connection (ESTABLISHED, FIN-WAIT-1, FIN-WAIT-2, CLOSE-WAIT, CLOSING, or LAST-ACK). Moreover, when used with the TCP keepalive (SO_KEEPALIVE) option, TCP_USER_TIMEOUT will overtake keepalive to determine when to close a connection due to keepalive failure. The option does not change in anyway when TCP retransmits a packet, nor when a keepalive probe will be sent. This option, like many others, will be inherited by an acceptor from its listener. Signed-off-by: NH.K. Jerry Chu <hkchu@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 19 2月, 2010 2 次提交
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由 Andreas Petlund 提交于
This patch enables fast retransmissions after one dupACK for TCP if the stream is identified as thin. This will reduce latencies for thin streams that are not able to trigger fast retransmissions due to high packet interarrival time. This mechanism is only active if enabled by iocontrol or syscontrol and the stream is identified as thin. Signed-off-by: NAndreas Petlund <apetlund@simula.no> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Andreas Petlund 提交于
This patch will make TCP use only linear timeouts if the stream is thin. This will help to avoid the very high latencies that thin stream suffer because of exponential backoff. This mechanism is only active if enabled by iocontrol or syscontrol and the stream is identified as thin. A maximum of 6 linear timeouts is tried before exponential backoff is resumed. Signed-off-by: NAndreas Petlund <apetlund@simula.no> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 03 12月, 2009 1 次提交
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由 William Allen Simpson 提交于
Data structures are carefully composed to require minimal additions. For example, the struct tcp_options_received cookie_plus variable fits between existing 16-bit and 8-bit variables, requiring no additional space (taking alignment into consideration). There are no additions to tcp_request_sock, and only 1 pointer in tcp_sock. This is a significantly revised implementation of an earlier (year-old) patch that no longer applies cleanly, with permission of the original author (Adam Langley): http://thread.gmane.org/gmane.linux.network/102586 The principle difference is using a TCP option to carry the cookie nonce, instead of a user configured offset in the data. This is more flexible and less subject to user configuration error. Such a cookie option has been suggested for many years, and is also useful without SYN data, allowing several related concepts to use the same extension option. "Re: SYN floods (was: does history repeat itself?)", September 9, 1996. http://www.merit.net/mail.archives/nanog/1996-09/msg00235.html "Re: what a new TCP header might look like", May 12, 1998. ftp://ftp.isi.edu/end2end/end2end-interest-1998.mail These functions will also be used in subsequent patches that implement additional features. Requires: TCPCT part 1a: add request_values parameter for sending SYNACK TCPCT part 1b: generate Responder Cookie secret TCPCT part 1c: sysctl_tcp_cookie_size, socket option TCP_COOKIE_TRANSACTIONS Signed-off-by: William.Allen.Simpson@gmail.com Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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