summaryrefslogtreecommitdiffstats
path: root/include/net/tcp.h
Commit message (Collapse)AuthorAgeFilesLines
* net: tcp: Define the TCP_MAX_WSCALE instead of literal number 14Gao Feng2017-04-051-0/+3
| | | | | | | | | | Define one new macro TCP_MAX_WSCALE instead of literal number '14', and use U16_MAX instead of 65535 as the max value of TCP window. There is another minor change, use rounddown(space, mss) instead of (space / mss) * mss; Signed-off-by: Gao Feng <fgao@ikuai8.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: sysctl: Fix a race to avoid unexpected 0 window from spaceGao Feng2017-03-241-3/+5
| | | | | | | | | | | | | | | | | Because sysctl_tcp_adv_win_scale could be changed any time, so there is one race in tcp_win_from_space. For example, 1.sysctl_tcp_adv_win_scale<=0 (sysctl_tcp_adv_win_scale is negative now) 2.space>>(-sysctl_tcp_adv_win_scale) (sysctl_tcp_adv_win_scale is postive now) As a result, tcp_win_from_space returns 0. It is unexpected. Certainly if the compiler put the sysctl_tcp_adv_win_scale into one register firstly, then use the register directly, it would be ok. But we could not depend on the compiler behavior. Signed-off-by: Gao Feng <fgao@ikuai8.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: remove tcp_tw_recycleSoheil Hassas Yeganeh2017-03-161-2/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | The tcp_tw_recycle was already broken for connections behind NAT, since the per-destination timestamp is not monotonically increasing for multiple machines behind a single destination address. After the randomization of TCP timestamp offsets in commit 8a5bd45f6616 (tcp: randomize tcp timestamp offsets for each connection), the tcp_tw_recycle is broken for all types of connections for the same reason: the timestamps received from a single machine is not monotonically increasing, anymore. Remove tcp_tw_recycle, since it is not functional. Also, remove the PAWSPassive SNMP counter since it is only used for tcp_tw_recycle, and simplify tcp_v4_route_req and tcp_v6_route_req since the strict argument is only set when tcp_tw_recycle is enabled. Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Cc: Lutz Vieweg <lvml@5t9.de> Cc: Florian Westphal <fw@strlen.de> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: remove per-destination timestamp cacheSoheil Hassas Yeganeh2017-03-161-5/+1
| | | | | | | | | | | | | | | | | | | | | | | Commit 8a5bd45f6616 (tcp: randomize tcp timestamp offsets for each connection) randomizes TCP timestamps per connection. After this commit, there is no guarantee that the timestamps received from the same destination are monotonically increasing. As a result, the per-destination timestamp cache in TCP metrics (i.e., tcpm_ts in struct tcp_metrics_block) is broken and cannot be relied upon. Remove the per-destination timestamp cache and all related code paths. Note that this cache was already broken for caching timestamps of multiple machines behind a NAT sharing the same address. Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Cc: Lutz Vieweg <lvml@5t9.de> Cc: Florian Westphal <fw@strlen.de> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: rename *_sequence_number() to *_seq_and_tsoff()Alexey Kodanev2017-03-091-1/+1
| | | | | | | | | | | The functions that are returning tcp sequence number also setup TS offset value, so rename them to better describe their purpose. No functional changes in this patch. Suggested-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Alexey Kodanev <alexey.kodanev@oracle.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* net/tcp-fastopen: Add new API supportWei Wang2017-01-251-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch adds a new socket option, TCP_FASTOPEN_CONNECT, as an alternative way to perform Fast Open on the active side (client). Prior to this patch, a client needs to replace the connect() call with sendto(MSG_FASTOPEN). This can be cumbersome for applications who want to use Fast Open: these socket operations are often done in lower layer libraries used by many other applications. Changing these libraries and/or the socket call sequences are not trivial. A more convenient approach is to perform Fast Open by simply enabling a socket option when the socket is created w/o changing other socket calls sequence: s = socket() create a new socket setsockopt(s, IPPROTO_TCP, TCP_FASTOPEN_CONNECT …); newly introduced sockopt If set, new functionality described below will be used. Return ENOTSUPP if TFO is not supported or not enabled in the kernel. connect() With cookie present, return 0 immediately. With no cookie, initiate 3WHS with TFO cookie-request option and return -1 with errno = EINPROGRESS. write()/sendmsg() With cookie present, send out SYN with data and return the number of bytes buffered. With no cookie, and 3WHS not yet completed, return -1 with errno = EINPROGRESS. No MSG_FASTOPEN flag is needed. read() Return -1 with errno = EWOULDBLOCK/EAGAIN if connect() is called but write() is not called yet. Return -1 with errno = EWOULDBLOCK/EAGAIN if connection is established but no msg is received yet. Return number of bytes read if socket is established and there is msg received. The new API simplifies life for applications that always perform a write() immediately after a successful connect(). Such applications can now take advantage of Fast Open by merely making one new setsockopt() call at the time of creating the socket. Nothing else about the application's socket call sequence needs to change. Signed-off-by: Wei Wang <weiwan@google.com> Acked-by: Eric Dumazet <edumazet@google.com> Acked-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* net/tcp-fastopen: refactor cookie check logicWei Wang2017-01-251-0/+2
| | | | | | | | | | Refactor the cookie check logic in tcp_send_syn_data() into a function. This function will be called else where in later changes. Signed-off-by: Wei Wang <weiwan@google.com> Acked-by: Eric Dumazet <edumazet@google.com> Acked-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: remove early retransmitYuchung Cheng2017-01-131-19/+0
| | | | | | | | | | | | | | | This patch removes the support of RFC5827 early retransmit (i.e., fast recovery on small inflight with <3 dupacks) because it is subsumed by the new RACK loss detection. More specifically when RACK receives DUPACKs, it'll arm a reordering timer to start fast recovery after a quarter of (min)RTT, hence it covers the early retransmit except RACK does not limit itself to specific inflight or dupack numbers. Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Acked-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: enable RACK loss detection to trigger recoveryYuchung Cheng2017-01-131-7/+4
| | | | | | | | | | | | | | | | | | This patch changes two things: 1. Start fast recovery with RACK in addition to other heuristics (e.g., DUPACK threshold, FACK). Prior to this change RACK is enabled to detect losses only after the recovery has started by other algorithms. 2. Disable TCP early retransmit. RACK subsumes the early retransmit with the new reordering timer feature. A latter patch in this series removes the early retransmit code. Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Acked-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: use sequence to break TS ties for RACK loss detectionYuchung Cheng2017-01-131-1/+1
| | | | | | | | | | | | | | | | | | | The packets inside a jumbo skb (e.g., TSO) share the same skb timestamp, even though they are sent sequentially on the wire. Since RACK is based on time, it can not detect some packets inside the same skb are lost. However, we can leverage the packet sequence numbers as extended timestamps to detect losses. Therefore, when RACK timestamp is identical to skb's timestamp (i.e., one of the packets of the skb is acked or sacked), we use the sequence numbers of the acked and unacked packets to break ties. We can use the same sequence logic to advance RACK xmit time as well to detect more losses and avoid timeout. Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Acked-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: add reordering timer in RACK loss detectionYuchung Cheng2017-01-131-0/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch makes RACK install a reordering timer when it suspects some packets might be lost, but wants to delay the decision a little bit to accomodate reordering. It does not create a new timer but instead repurposes the existing RTO timer, because both are meant to retransmit packets. Specifically it arms a timer ICSK_TIME_REO_TIMEOUT when the RACK timing check fails. The wait time is set to RACK.RTT + RACK.reo_wnd - (NOW - Packet.xmit_time) + fudge This translates to expecting a packet (Packet) should take (RACK.RTT + RACK.reo_wnd + fudge) to deliver after it was sent. When there are multiple packets that need a timer, we use one timer with the maximum timeout. Therefore the timer conservatively uses the maximum window to expire N packets by one timeout, instead of N timeouts to expire N packets sent at different times. The fudge factor is 2 jiffies to ensure when the timer fires, all the suspected packets would exceed the deadline and be marked lost by tcp_rack_detect_loss(). It has to be at least 1 jiffy because the clock may tick between calling icsk_reset_xmit_timer(timeout) and actually hang the timer. The next jiffy is to lower-bound the timeout to 2 jiffies when reo_wnd is < 1ms. When the reordering timer fires (tcp_rack_reo_timeout): If we aren't in Recovery we'll enter fast recovery and force fast retransmit. This is very similar to the early retransmit (RFC5827) except RACK is not constrained to only enter recovery for small outstanding flights. Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Acked-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: record most recent RTT in RACK loss detectionYuchung Cheng2017-01-131-3/+4
| | | | | | | | | | | | | | | | | | | Record the most recent RTT in RACK. It is often identical to the "ca_rtt_us" values in tcp_clean_rtx_queue. But when the packet has been retransmitted, RACK choses to believe the ACK is for the (latest) retransmitted packet if the RTT is over minimum RTT. This requires passing the arrival time of the most recent ACK to RACK routines. The timestamp is now recorded in the "ack_time" in tcp_sacktag_state during the ACK processing. This patch does not change the RACK algorithm itself. It only adds the RTT variable to prepare the next main patch. Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Acked-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: new helper for RACK to detect lossYuchung Cheng2017-01-131-2/+1
| | | | | | | | | | Create a new helper tcp_rack_detect_loss to prepare the upcoming RACK reordering timer patch. Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Acked-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* ipv4: Namespaceify tcp_tw_recycle and tcp_max_tw_buckets knobHaishuang Yan2016-12-291-1/+0
| | | | | | | | Different namespace application might require fast recycling TIME-WAIT sockets independently of the host. Signed-off-by: Haishuang Yan <yanhaishuang@cmss.chinamobile.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* ipv4: Namespaceify tcp_tw_reuse knobHaishuang Yan2016-12-271-1/+0
| | | | | | | | | | | Different namespaces might have different requirements to reuse TIME-WAIT sockets for new connections. This might be required in cases where different namespace applications are in place which require TIME_WAIT socket connections to be reduced independently of the host. Signed-off-by: Haishuang Yan <yanhaishuang@cmss.chinamobile.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: randomize tcp timestamp offsets for each connectionFlorian Westphal2016-12-021-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | jiffies based timestamps allow for easy inference of number of devices behind NAT translators and also makes tracking of hosts simpler. commit ceaa1fef65a7c2e ("tcp: adding a per-socket timestamp offset") added the main infrastructure that is needed for per-connection ts randomization, in particular writing/reading the on-wire tcp header format takes the offset into account so rest of stack can use normal tcp_time_stamp (jiffies). So only two items are left: - add a tsoffset for request sockets - extend the tcp isn generator to also return another 32bit number in addition to the ISN. Re-use of ISN generator also means timestamps are still monotonically increasing for same connection quadruple, i.e. PAWS will still work. Includes fixes from Eric Dumazet. Signed-off-by: Florian Westphal <fw@strlen.de> Acked-by: Eric Dumazet <edumazet@google.com> Acked-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: instrument how long TCP is busy sendingFrancis Yan2016-11-301-1/+5
| | | | | | | | | | | | | | | | | | | This patch measures TCP busy time, which is defined as the period of time when sender has data (or FIN) to send. The time starts when data is buffered and stops when the write queue is flushed by ACKs or error events. Note the busy time does not include SYN time, unless data is included in SYN (i.e. Fast Open). It does include FIN time even if the FIN carries no payload. Excluding pure FIN is possible but would incur one additional test in the fast path, which may not be worth it. Signed-off-by: Francis Yan <francisyyan@gmail.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Acked-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: instrument tcp sender limits chronographsFrancis Yan2016-11-301-0/+14
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch implements the skeleton of the TCP chronograph instrumentation on sender side limits: 1) idle (unspec) 2) busy sending data other than 3-4 below 3) rwnd-limited 4) sndbuf-limited The limits are enumerated 'tcp_chrono'. Since a connection in theory can idle forever, we do not track the actual length of this uninteresting idle period. For the rest we track how long the sender spends in each limit. At any point during the life time of a connection, the sender must be in one of the four states. If there are multiple conditions worthy of tracking in a chronograph then the highest priority enum takes precedence over the other conditions. So that if something "more interesting" starts happening, stop the previous chrono and start a new one. The time unit is jiffy(u32) in order to save space in tcp_sock. This implies application must sample the stats no longer than every 49 days of 1ms jiffy. Signed-off-by: Francis Yan <francisyyan@gmail.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Acked-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: make undo_cwnd mandatory for congestion modulesFlorian Westphal2016-11-211-0/+1
| | | | | | | | | | | | | | | The undo_cwnd fallback in the stack doubles cwnd based on ssthresh, which un-does reno halving behaviour. It seems more appropriate to let congctl algorithms pair .ssthresh and .undo_cwnd properly. Add a 'tcp_reno_undo_cwnd' function and wire it up for all congestion algorithms that used to rely on the fallback. Cc: Eric Dumazet <edumazet@google.com> Cc: Yuchung Cheng <ycheng@google.com> Cc: Neal Cardwell <ncardwell@google.com> Signed-off-by: Florian Westphal <fw@strlen.de> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: take care of truncations done by sk_filter()Eric Dumazet2016-11-131-0/+1
| | | | | | | | | | | | | | | | | | | With syzkaller help, Marco Grassi found a bug in TCP stack, crashing in tcp_collapse() Root cause is that sk_filter() can truncate the incoming skb, but TCP stack was not really expecting this to happen. It probably was expecting a simple DROP or ACCEPT behavior. We first need to make sure no part of TCP header could be removed. Then we need to adjust TCP_SKB_CB(skb)->end_seq Many thanks to syzkaller team and Marco for giving us a reproducer. Signed-off-by: Eric Dumazet <edumazet@google.com> Reported-by: Marco Grassi <marco.gra@gmail.com> Reported-by: Vladis Dronov <vdronov@redhat.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* net: tcp: check skb is non-NULL for exact match on lookupsDavid Ahern2016-11-031-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Andrey reported the following error report while running the syzkaller fuzzer: general protection fault: 0000 [#1] SMP KASAN Dumping ftrace buffer: (ftrace buffer empty) Modules linked in: CPU: 0 PID: 648 Comm: syz-executor Not tainted 4.9.0-rc3+ #333 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 task: ffff8800398c4480 task.stack: ffff88003b468000 RIP: 0010:[<ffffffff83091106>] [< inline >] inet_exact_dif_match include/net/tcp.h:808 RIP: 0010:[<ffffffff83091106>] [<ffffffff83091106>] __inet_lookup_listener+0xb6/0x500 net/ipv4/inet_hashtables.c:219 RSP: 0018:ffff88003b46f270 EFLAGS: 00010202 RAX: 0000000000000004 RBX: 0000000000004242 RCX: 0000000000000001 RDX: 0000000000000000 RSI: ffffc90000e3c000 RDI: 0000000000000054 RBP: ffff88003b46f2d8 R08: 0000000000004000 R09: ffffffff830910e7 R10: 0000000000000000 R11: 000000000000000a R12: ffffffff867fa0c0 R13: 0000000000004242 R14: 0000000000000003 R15: dffffc0000000000 FS: 00007fb135881700(0000) GS:ffff88003ec00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000000020cc3000 CR3: 000000006d56a000 CR4: 00000000000006f0 Stack: 0000000000000000 000000000601a8c0 0000000000000000 ffffffff00004242 424200003b9083c2 ffff88003def4041 ffffffff84e7e040 0000000000000246 ffff88003a0911c0 0000000000000000 ffff88003a091298 ffff88003b9083ae Call Trace: [<ffffffff831100f4>] tcp_v4_send_reset+0x584/0x1700 net/ipv4/tcp_ipv4.c:643 [<ffffffff83115b1b>] tcp_v4_rcv+0x198b/0x2e50 net/ipv4/tcp_ipv4.c:1718 [<ffffffff83069d22>] ip_local_deliver_finish+0x332/0xad0 net/ipv4/ip_input.c:216 ... MD5 has a code path that calls __inet_lookup_listener with a null skb, so inet{6}_exact_dif_match needs to check skb against null before pulling the flag. Fixes: a04a480d4392 ("net: Require exact match for TCP socket lookups if dif is l3mdev") Reported-by: Andrey Konovalov <andreyknvl@google.com> Signed-off-by: David Ahern <dsa@cumulusnetworks.com> Tested-by: Andrey Konovalov <andreyknvl@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* net: Require exact match for TCP socket lookups if dif is l3mdevDavid Ahern2016-10-171-1/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently, socket lookups for l3mdev (vrf) use cases can match a socket that is bound to a port but not a device (ie., a global socket). If the sysctl tcp_l3mdev_accept is not set this leads to ack packets going out based on the main table even though the packet came in from an L3 domain. The end result is that the connection does not establish creating confusion for users since the service is running and a socket shows in ss output. Fix by requiring an exact dif to sk_bound_dev_if match if the skb came through an interface enslaved to an l3mdev device and the tcp_l3mdev_accept is not set. skb's through an l3mdev interface are marked by setting a flag in inet{6}_skb_parm. The IPv6 variant is already set; this patch adds the flag for IPv4. Using an skb flag avoids a device lookup on the dif. The flag is set in the VRF driver using the IP{6}CB macros. For IPv4, the inet_skb_parm struct is moved in the cb per commit 971f10eca186, so the match function in the TCP stack needs to use TCP_SKB_CB. For IPv6, the move is done after the socket lookup, so IP6CB is used. The flags field in inet_skb_parm struct needs to be increased to add another flag. There is currently a 1-byte hole following the flags, so it can be expanded to u16 without increasing the size of the struct. Fixes: 193125dbd8eb ("net: Introduce VRF device driver") Signed-off-by: David Ahern <dsa@cumulusnetworks.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: new CC hook to set sending rate with rate_sample in any CA stateYuchung Cheng2016-09-211-0/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This commit introduces an optional new "omnipotent" hook, cong_control(), for congestion control modules. The cong_control() function is called at the end of processing an ACK (i.e., after updating sequence numbers, the SACK scoreboard, and loss detection). At that moment we have precise delivery rate information the congestion control module can use to control the sending behavior (using cwnd, TSO skb size, and pacing rate) in any CA state. This function can also be used by a congestion control that prefers not to use the default cwnd reduction approach (i.e., the PRR algorithm) during CA_Recovery to control the cwnd and sending rate during loss recovery. We take advantage of the fact that recent changes defer the retransmission or transmission of new data (e.g. by F-RTO) in recovery until the new tcp_cong_control() function is run. With this commit, we only run tcp_update_pacing_rate() if the congestion control is not using this new API. New congestion controls which use the new API do not want the TCP stack to run the default pacing rate calculation and overwrite whatever pacing rate they have chosen at initialization time. Signed-off-by: Van Jacobson <vanj@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Nandita Dukkipati <nanditad@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: allow congestion control to expand send buffer differentlyYuchung Cheng2016-09-211-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | Currently the TCP send buffer expands to twice cwnd, in order to allow limited transmits in the CA_Recovery state. This assumes that cwnd does not increase in the CA_Recovery. For some congestion control algorithms, like the upcoming BBR module, if the losses in recovery do not indicate congestion then we may continue to raise cwnd multiplicatively in recovery. In such cases the current multiplier will falsely limit the sending rate, much as if it were limited by the application. This commit adds an optional congestion control callback to use a different multiplier to expand the TCP send buffer. For congestion control modules that do not specificy this callback, TCP continues to use the previous default of 2. Signed-off-by: Van Jacobson <vanj@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Nandita Dukkipati <nanditad@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Acked-by: Stephen Hemminger <stephen@networkplumber.org> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: export tcp_tso_autosize() and parameterize minimum number of TSO segmentsNeal Cardwell2016-09-211-0/+2
| | | | | | | | | | | | | | | | | | | | | To allow congestion control modules to use the default TSO auto-sizing algorithm as one of the ingredients in their own decision about TSO sizing: 1) Export tcp_tso_autosize() so that CC modules can use it. 2) Change tcp_tso_autosize() to allow callers to specify a minimum number of segments per TSO skb, in case the congestion control module has a different notion of the best floor for TSO skbs for the connection right now. For very low-rate paths or policed connections it can be appropriate to use smaller TSO skbs. Signed-off-by: Van Jacobson <vanj@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Nandita Dukkipati <nanditad@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: allow congestion control module to request TSO skb segment countNeal Cardwell2016-09-211-0/+2
| | | | | | | | | | | | | | | | | | | | | Add the tso_segs_goal() function in tcp_congestion_ops to allow the congestion control module to specify the number of segments that should be in a TSO skb sent by tcp_write_xmit() and tcp_xmit_retransmit_queue(). The congestion control module can either request a particular number of segments in TSO skb that we transmit, or return 0 if it doesn't care. This allows the upcoming BBR congestion control module to select small TSO skb sizes if the module detects that the bottleneck bandwidth is very low, or that the connection is policed to a low rate. Signed-off-by: Van Jacobson <vanj@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Nandita Dukkipati <nanditad@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: track application-limited rate samplesSoheil Hassas Yeganeh2016-09-211-1/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This commit adds code to track whether the delivery rate represented by each rate_sample was limited by the application. Upon each transmit, we store in the is_app_limited field in the skb a boolean bit indicating whether there is a known "bubble in the pipe": a point in the rate sample interval where the sender was application-limited, and did not transmit even though the cwnd and pacing rate allowed it. This logic marks the flow app-limited on a write if *all* of the following are true: 1) There is less than 1 MSS of unsent data in the write queue available to transmit. 2) There is no packet in the sender's queues (e.g. in fq or the NIC tx queue). 3) The connection is not limited by cwnd. 4) There are no lost packets to retransmit. The tcp_rate_check_app_limited() code in tcp_rate.c determines whether the connection is application-limited at the moment. If the flow is application-limited, it sets the tp->app_limited field. If the flow is application-limited then that means there is effectively a "bubble" of silence in the pipe now, and this silence will be reflected in a lower bandwidth sample for any rate samples from now until we get an ACK indicating this bubble has exited the pipe: specifically, until we get an ACK for the next packet we transmit. When we send every skb we record in scb->tx.is_app_limited whether the resulting rate sample will be application-limited. The code in tcp_rate_gen() checks to see when it is safe to mark all known application-limited bubbles of silence as having exited the pipe. It does this by checking to see when the delivered count moves past the tp->app_limited marker. At this point it zeroes the tp->app_limited marker, as all known bubbles are out of the pipe. We make room for the tx.is_app_limited bit in the skb by borrowing a bit from the in_flight field used by NV to record the number of bytes in flight. The receive window in the TCP header is 16 bits, and the max receive window scaling shift factor is 14 (RFC 1323). So the max receive window offered by the TCP protocol is 2^(16+14) = 2^30. So we only need 30 bits for the tx.in_flight used by NV. Signed-off-by: Van Jacobson <vanj@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Nandita Dukkipati <nanditad@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: track data delivery rate for a TCP connectionYuchung Cheng2016-09-211-1/+34
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch generates data delivery rate (throughput) samples on a per-ACK basis. These rate samples can be used by congestion control modules, and specifically will be used by TCP BBR in later patches in this series. Key state: tp->delivered: Tracks the total number of data packets (original or not) delivered so far. This is an already-existing field. tp->delivered_mstamp: the last time tp->delivered was updated. Algorithm: A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis: d1: the current tp->delivered after processing the ACK t1: the current time after processing the ACK d0: the prior tp->delivered when the acked skb was transmitted t0: the prior tp->delivered_mstamp when the acked skb was transmitted When an skb is transmitted, we snapshot d0 and t0 in its control block in tcp_rate_skb_sent(). When an ACK arrives, it may SACK and ACK some skbs. For each SACKed or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct to reflect the latest (d0, t0). Finally, tcp_rate_gen() generates a rate sample by storing (d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us. One caveat: if an skb was sent with no packets in flight, then tp->delivered_mstamp may be either invalid (if the connection is starting) or outdated (if the connection was idle). In that case, we'll re-stamp tp->delivered_mstamp. At first glance it seems t0 should always be the time when an skb was transmitted, but actually this could over-estimate the rate due to phase mismatch between transmit and ACK events. To track the delivery rate, we ensure that if packets are in flight then t0 and and t1 are times at which packets were marked delivered. If the initial and final RTTs are different then one may be corrupted by some sort of noise. The noise we see most often is sending gaps caused by delayed, compressed, or stretched acks. This either affects both RTTs equally or artificially reduces the final RTT. We approach this by recording the info we need to compute the initial RTT (duration of the "send phase" of the window) when we recorded the associated inflight. Then, for a filter to avoid bandwidth overestimates, we generalize the per-sample bandwidth computation from: bw = delivered / ack_phase_rtt to the following: bw = delivered / max(send_phase_rtt, ack_phase_rtt) In large-scale experiments, this filtering approach incorporating send_phase_rtt is effective at avoiding bandwidth overestimates due to ACK compression or stretched ACKs. Signed-off-by: Van Jacobson <vanj@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Nandita Dukkipati <nanditad@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: use windowed min filter library for TCP min_rtt estimationNeal Cardwell2016-09-211-1/+1
| | | | | | | | | | | | | | | | | | Refactor the TCP min_rtt code to reuse the new win_minmax library in lib/win_minmax.c to simplify the TCP code. This is a pure refactor: the functionality is exactly the same. We just moved the windowed min code to make TCP easier to read and maintain, and to allow other parts of the kernel to use the windowed min/max filter code. Signed-off-by: Van Jacobson <vanj@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Nandita Dukkipati <nanditad@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: use an RB tree for ooo receive queueYaogong Wang2016-09-081-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Over the years, TCP BDP has increased by several orders of magnitude, and some people are considering to reach the 2 Gbytes limit. Even with current window scale limit of 14, ~1 Gbytes maps to ~740,000 MSS. In presence of packet losses (or reorders), TCP stores incoming packets into an out of order queue, and number of skbs sitting there waiting for the missing packets to be received can be in the 10^5 range. Most packets are appended to the tail of this queue, and when packets can finally be transferred to receive queue, we scan the queue from its head. However, in presence of heavy losses, we might have to find an arbitrary point in this queue, involving a linear scan for every incoming packet, throwing away cpu caches. This patch converts it to a RB tree, to get bounded latencies. Yaogong wrote a preliminary patch about 2 years ago. Eric did the rebase, added ofo_last_skb cache, polishing and tests. Tested with network dropping between 1 and 10 % packets, with good success (about 30 % increase of throughput in stress tests) Next step would be to also use an RB tree for the write queue at sender side ;) Signed-off-by: Yaogong Wang <wygivan@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Cc: Yuchung Cheng <ycheng@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi> Acked-By: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi> Signed-off-by: David S. Miller <davem@davemloft.net>
* Merge git://git.kernel.org/pub/scm/linux/kernel/git/davem/netDavid S. Miller2016-08-301-0/+2
|\ | | | | | | | | | | | | All three conflicts were cases of simple overlapping changes. Signed-off-by: David S. Miller <davem@davemloft.net>
| * tcp: fix use after free in tcp_xmit_retransmit_queue()Eric Dumazet2016-08-181-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When tcp_sendmsg() allocates a fresh and empty skb, it puts it at the tail of the write queue using tcp_add_write_queue_tail() Then it attempts to copy user data into this fresh skb. If the copy fails, we undo the work and remove the fresh skb. Unfortunately, this undo lacks the change done to tp->highest_sack and we can leave a dangling pointer (to a freed skb) Later, tcp_xmit_retransmit_queue() can dereference this pointer and access freed memory. For regular kernels where memory is not unmapped, this might cause SACK bugs because tcp_highest_sack_seq() is buggy, returning garbage instead of tp->snd_nxt, but with various debug features like CONFIG_DEBUG_PAGEALLOC, this can crash the kernel. This bug was found by Marco Grassi thanks to syzkaller. Fixes: 6859d49475d4 ("[TCP]: Abstract tp->highest_sack accessing & point to next skb") Reported-by: Marco Grassi <marco.gra@gmail.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Cc: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi> Cc: Yuchung Cheng <ycheng@google.com> Cc: Neal Cardwell <ncardwell@google.com> Acked-by: Neal Cardwell <ncardwell@google.com> Reviewed-by: Cong Wang <xiyou.wangcong@gmail.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* | tcp: add tcp_add_backlog()Eric Dumazet2016-08-291-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When TCP operates in lossy environments (between 1 and 10 % packet losses), many SACK blocks can be exchanged, and I noticed we could drop them on busy senders, if these SACK blocks have to be queued into the socket backlog. While the main cause is the poor performance of RACK/SACK processing, we can try to avoid these drops of valuable information that can lead to spurious timeouts and retransmits. Cause of the drops is the skb->truesize overestimation caused by : - drivers allocating ~2048 (or more) bytes as a fragment to hold an Ethernet frame. - various pskb_may_pull() calls bringing the headers into skb->head might have pulled all the frame content, but skb->truesize could not be lowered, as the stack has no idea of each fragment truesize. The backlog drops are also more visible on bidirectional flows, since their sk_rmem_alloc can be quite big. Let's add some room for the backlog, as only the socket owner can selectively take action to lower memory needs, like collapsing receive queues or partial ofo pruning. Signed-off-by: Eric Dumazet <edumazet@google.com> Cc: Yuchung Cheng <ycheng@google.com> Cc: Neal Cardwell <ncardwell@google.com> Acked-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* | tcp: Set read_sock and peek_len proto_opsTom Herbert2016-08-281-0/+2
| | | | | | | | | | | | | | | | In inet_stream_ops we set read_sock to tcp_read_sock and peek_len to tcp_peek_len (which is just a stub function that calls tcp_inq). Signed-off-by: Tom Herbert <tom@herbertland.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* | net: Add read_sock proto_opTom Herbert2016-08-281-2/+0
| | | | | | | | | | | | | | | | | | Add new function in proto_ops structure. This includes moving the typedef got sk_read_actor into net.h and removing the definition from tcp.h. Signed-off-by: Tom Herbert <tom@herbertland.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* | net-tcp: retire TFO_SERVER_WO_SOCKOPT2 configYuchung Cheng2016-08-231-2/+1
|/ | | | | | | | | | | | TFO_SERVER_WO_SOCKOPT2 was intended for debugging purposes during Fast Open development. Remove this config option and also update/clean-up the documentation of the Fast Open sysctl. Reported-by: Piotr Jurkiewicz <piotr.jerzy.jurkiewicz@gmail.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: md5: use kmalloc() backed scratch areasEric Dumazet2016-07-011-2/+1
| | | | | | | | | | | | | | | | | Some arches have virtually mapped kernel stacks, or will soon have. tcp_md5_hash_header() uses an automatic variable to copy tcp header before mangling th->check and calling crypto function, which might be problematic on such arches. David says that using percpu storage is also problematic on non SMP builds. Just use kmalloc() to allocate scratch areas. Signed-off-by: Eric Dumazet <edumazet@google.com> Reported-by: Andy Lutomirski <luto@amacapital.net> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: increase size at which tcp_bound_to_half_wnd bounds to > TCP_MSS_DEFAULTSeymour, Shane M2016-06-301-1/+1
| | | | | | | | | | | | | | | | | In previous commit 01f83d69844d307be2aa6fea88b0e8fe5cbdb2f4 the following comments were added: "When peer uses tiny windows, there is no use in packetizing to sub-MSS pieces for the sake of SWS or making sure there are enough packets in the pipe for fast recovery." The test should be > TCP_MSS_DEFAULT not >= 512. This allows low end devices that send an MSS of 536 (TCP_MSS_DEFAULT) to see better network performance by sending it 536 bytes of data at a time instead of bounding to half window size (268). Other network stacks work this way, e.g. HP-UX. Signed-off-by: Shane Seymour <shane.seymour@hpe.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: add in_flight to tcp_skb_cbLawrence Brakmo2016-06-101-0/+2
| | | | | | | | | | | Add in_flight (bytes in flight when packet was sent) field to tx component of tcp_skb_cb and make it available to congestion modules' pkts_acked() function through the ack_sample function argument. Signed-off-by: Lawrence Brakmo <brakmo@fb.com> Acked-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* net: l3mdev: Add hook in ip and ipv6David Ahern2016-05-111-1/+3
| | | | | | | | | | | | | | | | | | | | | Currently the VRF driver uses the rx_handler to switch the skb device to the VRF device. Switching the dev prior to the ip / ipv6 layer means the VRF driver has to duplicate IP/IPv6 processing which adds overhead and makes features such as retaining the ingress device index more complicated than necessary. This patch moves the hook to the L3 layer just after the first NF_HOOK for PRE_ROUTING. This location makes exposing the original ingress device trivial (next patch) and allows adding other NF_HOOKs to the VRF driver in the future. dev_queue_xmit_nit is exported so that the VRF driver can cycle the skb with the switched device through the packet taps to maintain current behavior (tcpdump can be used on either the vrf device or the enslaved devices). Signed-off-by: David Ahern <dsa@cumulusnetworks.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: replace cnt & rtt with struct in pkts_acked()Lawrence Brakmo2016-05-111-1/+6
| | | | | | | | | | | | | | Replace 2 arguments (cnt and rtt) in the congestion control modules' pkts_acked() function with a struct. This will allow adding more information without having to modify existing congestion control modules (tcp_nv in particular needs bytes in flight when packet was sent). As proposed by Neal Cardwell in his comments to the tcp_nv patch. Signed-off-by: Lawrence Brakmo <brakmo@fb.com> Acked-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: refactor struct tcp_skb_cbLawrence Brakmo2016-05-091-3/+8
| | | | | | | | | | | | | | | | Refactor tcp_skb_cb to create two overlaping areas to store state for incoming or outgoing skbs based on comments by Neal Cardwell to tcp_nv patch: AFAICT this patch would not require an increase in the size of sk_buff cb[] if it were to take advantage of the fact that the tcp_skb_cb header.h4 and header.h6 fields are only used in the packet reception code path, and this in_flight field is only used on the transmit side. Signed-off-by: Lawrence Brakmo <brakmo@fb.com> Acked-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp: Make use of MSG_EOR in tcp_sendmsgMartin KaFai Lau2016-04-281-1/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch adds an eor bit to the TCP_SKB_CB. When MSG_EOR is passed to tcp_sendmsg, the eor bit will be set at the skb containing the last byte of the userland's msg. The eor bit will prevent data from appending to that skb in the future. The change in do_tcp_sendpages is to honor the eor set during the previous tcp_sendmsg(MSG_EOR) call. This patch handles the tcp_sendmsg case. The followup patches will handle other skb coalescing and fragment cases. One potential use case is to use MSG_EOR with SOF_TIMESTAMPING_TX_ACK to get a more accurate TCP ack timestamping on application protocol with multiple outgoing response messages (e.g. HTTP2). Packetdrill script for testing: ~~~~~~ +0 `sysctl -q -w net.ipv4.tcp_min_tso_segs=10` +0 `sysctl -q -w net.ipv4.tcp_no_metrics_save=1` +0 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3 +0 setsockopt(3, SOL_SOCKET, SO_REUSEADDR, [1], 4) = 0 +0 bind(3, ..., ...) = 0 +0 listen(3, 1) = 0 0.100 < S 0:0(0) win 32792 <mss 1460,sackOK,nop,nop,nop,wscale 7> 0.100 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 7> 0.200 < . 1:1(0) ack 1 win 257 0.200 accept(3, ..., ...) = 4 +0 setsockopt(4, SOL_TCP, TCP_NODELAY, [1], 4) = 0 0.200 write(4, ..., 14600) = 14600 0.200 sendto(4, ..., 730, MSG_EOR, ..., ...) = 730 0.200 sendto(4, ..., 730, MSG_EOR, ..., ...) = 730 0.200 > . 1:7301(7300) ack 1 0.200 > P. 7301:14601(7300) ack 1 0.300 < . 1:1(0) ack 14601 win 257 0.300 > P. 14601:15331(730) ack 1 0.300 > P. 15331:16061(730) ack 1 0.400 < . 1:1(0) ack 16061 win 257 0.400 close(4) = 0 0.400 > F. 16061:16061(0) ack 1 0.400 < F. 1:1(0) ack 16062 win 257 0.400 > . 16062:16062(0) ack 2 Signed-off-by: Martin KaFai Lau <kafai@fb.com> Cc: Eric Dumazet <edumazet@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Soheil Hassas Yeganeh <soheil@google.com> Cc: Willem de Bruijn <willemb@google.com> Cc: Yuchung Cheng <ycheng@google.com> Suggested-by: Eric Dumazet <edumazet@google.com> Acked-by: Eric Dumazet <edumazet@google.com> Acked-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* net: snmp: kill STATS_BH macrosEric Dumazet2016-04-271-1/+1
| | | | | | | | | | | | | | There is nothing related to BH in SNMP counters anymore, since linux-3.0. Rename helpers to use __ prefix instead of _BH prefix, for contexts where preemption is disabled. This more closely matches convention used to update percpu variables. Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* net: rename NET_{ADD|INC}_STATS_BH()Eric Dumazet2016-04-271-2/+2
| | | | | | | | Rename NET_INC_STATS_BH() to __NET_INC_STATS() and NET_ADD_STATS_BH() to __NET_ADD_STATS() Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* net: tcp: rename TCP_INC_STATS_BHEric Dumazet2016-04-271-1/+1
| | | | | | | Rename TCP_INC_STATS_BH() to __TCP_INC_STATS() Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* net: snmp: kill various STATS_USER() helpersEric Dumazet2016-04-271-5/+4
| | | | | | | | | | | | | | | | | | | | | In the old days (before linux-3.0), SNMP counters were duplicated, one for user context, and one for BH context. After commit 8f0ea0fe3a03 ("snmp: reduce percpu needs by 50%") we have a single copy, and what really matters is preemption being enabled or disabled, since we use this_cpu_inc() or __this_cpu_inc() respectively. We therefore kill SNMP_INC_STATS_USER(), SNMP_ADD_STATS_USER(), NET_INC_STATS_USER(), NET_ADD_STATS_USER(), SCTP_INC_STATS_USER(), SNMP_INC_STATS64_USER(), SNMP_ADD_STATS64_USER(), TCP_ADD_STATS_USER(), UDP_INC_STATS_USER(), UDP6_INC_STATS_USER(), and XFRM_INC_STATS_USER() Following patches will rename __BH helpers to make clear their usage is not tied to BH being disabled. Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* tcp-tso: do not split TSO packets at retransmit timeEric Dumazet2016-04-241-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | Linux TCP stack painfully segments all TSO/GSO packets before retransmits. This was fine back in the days when TSO/GSO were emerging, with their bugs, but we believe the dark age is over. Keeping big packets in write queues, but also in stack traversal has a lot of benefits. - Less memory overhead, because write queues have less skbs - Less cpu overhead at ACK processing. - Better SACK processing, as lot of studies mentioned how awful linux was at this ;) - Less cpu overhead to send the rtx packets (IP stack traversal, netfilter traversal, drivers...) - Better latencies in presence of losses. - Smaller spikes in fq like packet schedulers, as retransmits are not constrained by TCP Small Queues. 1 % packet losses are common today, and at 100Gbit speeds, this translates to ~80,000 losses per second. Losses are often correlated, and we see many retransmit events leading to 1-MSS train of packets, at the time hosts are already under stress. Signed-off-by: Eric Dumazet <edumazet@google.com> Acked-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* Merge git://git.kernel.org/pub/scm/linux/kernel/git/davem/netDavid S. Miller2016-04-231-0/+2
|\ | | | | | | | | | | | | | | | | | | | | Conflicts were two cases of simple overlapping changes, nothing serious. In the UDP case, we need to add a hlist_add_tail_rcu() to linux/rculist.h, because we've moved UDP socket handling away from using nulls lists. Signed-off-by: David S. Miller <davem@davemloft.net>
| * tcp: Merge tx_flags and tskey in tcp_shifted_skbMartin KaFai Lau2016-04-211-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | After receiving sacks, tcp_shifted_skb() will collapse skbs if possible. tx_flags and tskey also have to be merged. This patch reuses the tcp_skb_collapse_tstamp() to handle them. BPF Output Before: ~~~~~ <no-output-due-to-missing-tstamp-event> BPF Output After: ~~~~~ <...>-2024 [007] d.s. 88.644374: : ee_data:14599 Packetdrill Script: ~~~~~ +0 `sysctl -q -w net.ipv4.tcp_min_tso_segs=10` +0 `sysctl -q -w net.ipv4.tcp_no_metrics_save=1` +0 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3 +0 setsockopt(3, SOL_SOCKET, SO_REUSEADDR, [1], 4) = 0 +0 bind(3, ..., ...) = 0 +0 listen(3, 1) = 0 0.100 < S 0:0(0) win 32792 <mss 1460,sackOK,nop,nop,nop,wscale 7> 0.100 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 7> 0.200 < . 1:1(0) ack 1 win 257 0.200 accept(3, ..., ...) = 4 +0 setsockopt(4, SOL_TCP, TCP_NODELAY, [1], 4) = 0 0.200 write(4, ..., 1460) = 1460 +0 setsockopt(4, SOL_SOCKET, 37, [2688], 4) = 0 0.200 write(4, ..., 13140) = 13140 0.200 > P. 1:1461(1460) ack 1 0.200 > . 1461:8761(7300) ack 1 0.200 > P. 8761:14601(5840) ack 1 0.300 < . 1:1(0) ack 1 win 257 <sack 1461:14601,nop,nop> 0.300 > P. 1:1461(1460) ack 1 0.400 < . 1:1(0) ack 14601 win 257 0.400 close(4) = 0 0.400 > F. 14601:14601(0) ack 1 0.500 < F. 1:1(0) ack 14602 win 257 0.500 > . 14602:14602(0) ack 2 Signed-off-by: Martin KaFai Lau <kafai@fb.com> Cc: Eric Dumazet <edumazet@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Soheil Hassas Yeganeh <soheil@google.com> Cc: Willem de Bruijn <willemb@google.com> Cc: Yuchung Cheng <ycheng@google.com> Acked-by: Soheil Hassas Yeganeh <soheil@google.com> Tested-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-19 02:32:05 +0000