- 17 7月, 2012 1 次提交
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由 David S. Miller 提交于
This will be used so that we can compose a full flow key. Even though we have a route in this context, we need more. In the future the routes will be without destination address, source address, etc. keying. One ipv4 route will cover entire subnets, etc. In this environment we have to have a way to possess persistent storage for redirects and PMTU information. This persistent storage will exist in the FIB tables, and that's why we'll need to be able to rebuild a full lookup flow key here. Using that flow key will do a fib_lookup() and create/update the persistent entry. Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 16 7月, 2012 1 次提交
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由 David S. Miller 提交于
This is the ipv6 version of inet_csk_update_pmtu(). Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 12 7月, 2012 3 次提交
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由 David S. Miller 提交于
No longer necessary. Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 David S. Miller 提交于
Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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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 3 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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由 David S. Miller 提交于
With help from Lin Ming. Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 David S. Miller 提交于
Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 05 7月, 2012 1 次提交
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由 RongQing.Li 提交于
opt always equals np->opts, so it is meaningless to define opt, and check if opt does not equal np->opts and then try to free opt. Signed-off-by: NRongQing.Li <roy.qing.li@gmail.com> Acked-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 29 6月, 2012 3 次提交
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由 Neal Cardwell 提交于
The code in tcp_v6_conn_request() was implicitly assuming that tcp_v6_send_synack() would take care of dst_release(), much as tcp_v4_send_synack() already does. This resulted in tcp_v6_conn_request() leaking a dst if sysctl_tw_recycle is enabled. This commit restructures tcp_v6_send_synack() so that it accepts a dst pointer and takes care of releasing the dst that is passed in, to plug the leak and avoid future surprises by bringing the IPv6 behavior in line with the IPv4 side. Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Neal Cardwell 提交于
With the recent change (earlier in this patch series) to set flowi6_oif to treq->iif in inet6_csk_route_req(), the dst lookup in these two functions is now identical, so tcp_v6_send_synack() can now just call inet6_csk_route_req(), to reduce code duplication and keep things closer to the IPv4 side, which is structured this way. Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Neal Cardwell 提交于
This commit changes inet_csk_route_req() so that it uses a pointer to a struct flowi6, rather than allocating its own on the stack. This brings its behavior in line with its IPv4 cousin, inet_csk_route_req(), and allows a follow-on patch to fix a dst leak. Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 26 6月, 2012 1 次提交
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由 Neal Cardwell 提交于
If security_inet_conn_request() returns non-zero then TCP/IPv6 should drop the request, just as in TCP/IPv4 and DCCP in both IPv4 and IPv6. Signed-off-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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- 16 6月, 2012 1 次提交
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由 David S. Miller 提交于
One tricky issue on the ipv6 side vs. ipv4 is that the ICMP callouts to handle the error pass the 32-bit info cookie in network byte order whereas ipv4 passes it around in host byte order. Like the ipv4 side, we have two helper functions. One for when we have a socket context and one for when we do not. ip6ip6 tunnels are not handled here, because they handle PMTU events by essentially relaying another ICMP packet-too-big message back to the original sender. This patch allows us to get rid of rt6_do_pmtu_disc(). It handles all kinds of situations that simply cannot happen when we do the PMTU update directly using a fully resolved route. In fact, the "plen == 128" check in ip6_rt_update_pmtu() can very likely be removed or changed into a BUG_ON() check. We should never have a prefixed ipv6 route when we get there. Another piece of strange history here is that TCP and DCCP, unlike in ipv4, never invoke the update_pmtu() method from their ICMP error handlers. This is incredibly astonishing since this is the context where we have the most accurate context in which to make a PMTU update, namely we have a fully connected socket and associated cached socket route. Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 10 6月, 2012 1 次提交
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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 6月, 2012 3 次提交
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由 David S. Miller 提交于
The get_peer method TCP uses is full of special cases that make no sense accommodating, and it also gets in the way of doing more reasonable things here. First of all, if the socket doesn't have a usable cached route, there is no sense in trying to optimize timewait recycling. Likewise for the case where we have IP options, such as SRR enabled, that make the IP header destination address (and thus the destination address of the route key) differ from that of the connection's destination address. Just return a NULL peer in these cases, and thus we're also able to get rid of the clumsy inetpeer release logic. Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 David S. Miller 提交于
There's a lot of places that open-code rt{,6}_get_peer() only because they want to set 'create' to one. So add an rt{,6}_get_peer_create() for their sake. There were also a few spots open-coding plain rt{,6}_get_peer() and those are transformed here as well. Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Gao feng 提交于
add struct net as a parameter of inet_getpeer_v[4,6], use net to replace &init_net. and modify some places to provide net for inet_getpeer_v[4,6] Signed-off-by: NGao feng <gaofeng@cn.fujitsu.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 04 6月, 2012 1 次提交
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由 Eric Dumazet 提交于
tcp_make_synack() clones the dst, and callers release it. We can avoid two atomic operations per SYNACK if tcp_make_synack() consumes dst instead of cloning it. Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 02 6月, 2012 1 次提交
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由 Eric Dumazet 提交于
While testing how linux behaves on SYNFLOOD attack on multiqueue device (ixgbe), I found that SYNACK messages were dropped at Qdisc level because we send them all on a single queue. Obvious choice is to reflect incoming SYN packet @queue_mapping to SYNACK packet. Under stress, my machine could only send 25.000 SYNACK per second (for 200.000 incoming SYN per second). NIC : ixgbe with 16 rx/tx queues. After patch, not a single SYNACK is dropped. Signed-off-by: NEric Dumazet <edumazet@google.com> Cc: Hans Schillstrom <hans.schillstrom@ericsson.com> Cc: Jesper Dangaard Brouer <brouer@redhat.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Tom Herbert <therbert@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 18 5月, 2012 1 次提交
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由 Eric Dumazet 提交于
bool conversions where possible. __inline__ -> inline space cleanups Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 16 5月, 2012 1 次提交
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由 Joe Perches 提交于
Standardize the net core ratelimited logging functions. Coalesce formats, align arguments. Change a printk then vprintk sequence to use printf extension %pV. Signed-off-by: NJoe Perches <joe@perches.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 05 5月, 2012 1 次提交
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由 Eric Dumazet 提交于
It appears some networks play bad games with the two bits reserved for ECN. This can trigger false congestion notifications and very slow transferts. Since RFC 3168 (6.1.1) forbids SYN packets to carry CT bits, we can disable TCP ECN negociation if it happens we receive mangled CT bits in the SYN packet. Signed-off-by: NEric Dumazet <edumazet@google.com> Cc: Perry Lorier <perryl@google.com> Cc: Matt Mathis <mattmathis@google.com> Cc: Yuchung Cheng <ycheng@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Wilmer van der Gaast <wilmer@google.com> Cc: Ankur Jain <jankur@google.com> Cc: Tom Herbert <therbert@google.com> Cc: Dave Täht <dave.taht@bufferbloat.net> Acked-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 27 4月, 2012 1 次提交
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由 Eric Dumazet 提交于
Quoting Tore Anderson from : https://bugzilla.kernel.org/show_bug.cgi?id=42572 When RTAX_FEATURE_ALLFRAG is set on a route, the effective TCP segment size does not take into account the size of the IPv6 Fragmentation header that needs to be included in outbound packets, causing every transmitted TCP segment to be fragmented across two IPv6 packets, the latter of which will only contain 8 bytes of actual payload. RTAX_FEATURE_ALLFRAG is typically set on a route in response to receving a ICMPv6 Packet Too Big message indicating a Path MTU of less than 1280 bytes. 1280 bytes is the minimum IPv6 MTU, however ICMPv6 PTBs with MTU < 1280 are still valid, in particular when an IPv6 packet is sent to an IPv4 destination through a stateless translator. Any ICMPv4 Need To Fragment packets originated from the IPv4 part of the path will be translated to ICMPv6 PTB which may then indicate an MTU of less than 1280. The Linux kernel refuses to reduce the effective MTU to anything below 1280 bytes, instead it sets it to exactly 1280 bytes, and RTAX_FEATURE_ALLFRAG is also set. However, the TCP segment size appears to be set to 1240 bytes (1280 Path MTU - 40 bytes of IPv6 header), instead of 1232 (additionally taking into account the 8 bytes required by the IPv6 Fragmentation extension header). This in turn results in rather inefficient transmission, as every transmitted TCP segment now is split in two fragments containing 1232+8 bytes of payload. After this patch, all the outgoing packets that includes a Fragmentation header all are "atomic" or "non-fragmented" fragments, i.e., they both have Offset=0 and More Fragments=0. With help from David S. Miller Reported-by: NTore Anderson <tore@fud.no> Signed-off-by: NEric Dumazet <edumazet@google.com> Cc: Maciej Żenczykowski <maze@google.com> Cc: Tom Herbert <therbert@google.com> Tested-by: NTore Anderson <tore@fud.no> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 24 4月, 2012 2 次提交
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由 Eric Dumazet 提交于
While investigating TCP performance problems on 10Gb+ links, we found a tcp sender was dropping lot of incoming ACKS because of sk_rcvbuf limit in sk_add_backlog(), especially if receiver doesnt use GRO/LRO and sends one ACK every two MSS segments. A sender usually tweaks sk_sndbuf, but sk_rcvbuf stays at its default value (87380), allowing a too small backlog. A TCP ACK, even being small, can consume nearly same truesize space than outgoing packets. Using sk_rcvbuf + sk_sndbuf as a limit makes sense and is fast to compute. Performance results on netperf, single flow, receiver with disabled GRO/LRO : 7500 Mbits instead of 6050 Mbits, no more TCPBacklogDrop increments at sender. Signed-off-by: NEric Dumazet <edumazet@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Tom Herbert <therbert@google.com> Cc: Maciej Żenczykowski <maze@google.com> Cc: Yuchung Cheng <ycheng@google.com> Cc: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi> Cc: Rick Jones <rick.jones2@hp.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Eric Dumazet 提交于
sk_add_backlog() & sk_rcvqueues_full() hard coded sk_rcvbuf as the memory limit. We need to make this limit a parameter for TCP use. No functional change expected in this patch, all callers still using the old sk_rcvbuf limit. Signed-off-by: NEric Dumazet <edumazet@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Tom Herbert <therbert@google.com> Cc: Maciej Żenczykowski <maze@google.com> Cc: Yuchung Cheng <ycheng@google.com> Cc: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi> Cc: Rick Jones <rick.jones2@hp.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 23 4月, 2012 2 次提交
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由 David S. Miller 提交于
net/ipv4/tcp_ipv4.c: In function 'tcp_v4_init_sock': net/ipv4/tcp_ipv4.c:1891:19: warning: unused variable 'tp' [-Wunused-variable] net/ipv6/tcp_ipv6.c: In function 'tcp_v6_init_sock': net/ipv6/tcp_ipv6.c:1836:19: warning: unused variable 'tp' [-Wunused-variable] Reported-by: NStephen Rothwell <sfr@canb.auug.org.au> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Neal Cardwell 提交于
Commit f5fff5dc forgot to fix TCP_MAXSEG behavior IPv6 sockets, so IPv6 TCP server sockets that used TCP_MAXSEG would find that the advmss of child sockets would be incorrect. This commit mirrors the advmss logic from tcp_v4_syn_recv_sock in tcp_v6_syn_recv_sock. Eventually this logic should probably be shared between IPv4 and IPv6, but this at least fixes this issue. Signed-off-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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- 22 4月, 2012 1 次提交
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由 Neal Cardwell 提交于
This commit moves the (substantial) common code shared between tcp_v4_init_sock() and tcp_v6_init_sock() to a new address-family independent function, tcp_init_sock(). Centralizing this functionality should help avoid drift issues, e.g. where the IPv4 side is updated without a corresponding update to IPv6. There was already some drift: IPv4 initialized snd_cwnd to TCP_INIT_CWND, while the IPv6 side was still initializing snd_cwnd to 2 (in this case it should not matter, since snd_cwnd is also initialized in tcp_init_metrics(), but the general risks and maintenance overhead remain). When diffing the old and new code, note that new tcp_init_sock() function uses the order of steps from the tcp_v4_init_sock() implementation (the order is slightly different in tcp_v6_init_sock()). Signed-off-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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- 20 4月, 2012 1 次提交
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由 Eric Dumazet 提交于
When we need to clone skb, we dont drop a packet. Call consume_skb() to not confuse dropwatch. Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 06 4月, 2012 1 次提交
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由 Dave Jiang 提交于
This is the fallout from adding memcpy alignment workaround for certain IOATDMA hardware. NetDMA will only use DMA engine that can handle byte align ops. Acked-by: NDavid S. Miller <davem@davemloft.net> Signed-off-by: NDave Jiang <dave.jiang@intel.com> Signed-off-by: NDan Williams <dan.j.williams@intel.com>
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- 13 2月, 2012 1 次提交
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由 Jiri Benc 提交于
Currently, it is not easily possible to get TOS/DSCP value of packets from an incoming TCP stream. The mechanism is there, IP_PKTOPTIONS getsockopt with IP_RECVTOS set, the same way as incoming TTL can be queried. This is not actually implemented for TOS, though. This patch adds this functionality, both for IPv4 (IP_PKTOPTIONS) and IPv6 (IPV6_2292PKTOPTIONS). For IPv4, like in the IP_RECVTTL case, the value of the TOS field is stored from the other party's ACK. This is needed for proxies which require DSCP transparency. One such example is at http://zph.bratcheda.org/. Signed-off-by: NJiri Benc <jbenc@redhat.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 02 2月, 2012 1 次提交
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由 Shawn Lu 提交于
TCP RST mechanism is broken in TCP md5(RFC2385). When connection is gone, md5 key is lost, sending RST without md5 hash is deem to ignored by peer. This can be a problem since RST help protocal like bgp to fast recove from peer crash. In most case, users of tcp md5, such as bgp and ldp, have listener on both sides to accept connection from peer. md5 keys for peers are saved in listening socket. There are two cases in finding md5 key when connection is lost: 1.Passive receive RST: The message is send to well known port, tcp will associate it with listner. md5 key is gotten from listener. 2.Active receive RST (no sock): The message is send to ative side, there is no socket associated with the message. In this case, finding listener from source port, then find md5 key from listener. we are not loosing sercuriy here: packet is checked with md5 hash. No RST is generated if md5 hash doesn't match or no md5 key can be found. Signed-off-by: NShawn Lu <shawn.lu@ericsson.com> Signed-off-by: NEric Dumazet <eric.dumazet@gmail.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 01 2月, 2012 3 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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由 Eric Dumazet 提交于
We no longer use md5_add() method from struct tcp_sock_af_ops Signed-off-by: NEric Dumazet <eric.dumazet@gmail.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 23 1月, 2012 1 次提交
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由 shawnlu 提交于
md5 key is added in socket through remote address. remote address should be used in finding md5 key when sending out reset packet. Signed-off-by: Nshawnlu <shawn.lu@ericsson.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 13 12月, 2011 3 次提交
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由 Glauber Costa 提交于
This patch allows each namespace to independently set up its levels for tcp memory pressure thresholds. This patch alone does not buy much: we need to make this values per group of process somehow. This is achieved in the patches that follows in this patchset. Signed-off-by: NGlauber Costa <glommer@parallels.com> Reviewed-by: NKAMEZAWA Hiroyuki <kamezawa.hiroyu@jp.fujitsu.com> CC: David S. Miller <davem@davemloft.net> CC: Eric W. Biederman <ebiederm@xmission.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Glauber Costa 提交于
This patch introduces memory pressure controls for the tcp protocol. It uses the generic socket memory pressure code introduced in earlier patches, and fills in the necessary data in cg_proto struct. Signed-off-by: NGlauber Costa <glommer@parallels.com> Reviewed-by: NKAMEZAWA Hiroyuki <kamezawa.hiroyu@jp.fujtisu.com> CC: Eric W. Biederman <ebiederm@xmission.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Glauber Costa 提交于
This patch replaces all uses of struct sock fields' memory_pressure, memory_allocated, sockets_allocated, and sysctl_mem to acessor macros. Those macros can either receive a socket argument, or a mem_cgroup argument, depending on the context they live in. Since we're only doing a macro wrapping here, no performance impact at all is expected in the case where we don't have cgroups disabled. Signed-off-by: NGlauber Costa <glommer@parallels.com> Reviewed-by: NHiroyouki Kamezawa <kamezawa.hiroyu@jp.fujitsu.com> CC: David S. Miller <davem@davemloft.net> CC: Eric W. Biederman <ebiederm@xmission.com> CC: Eric Dumazet <eric.dumazet@gmail.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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