| Commit message (Collapse) | Author | Age | Files | Lines |
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We use the idr subsystem and always ask for an id
at or above 1. This results in a id reuse when one
association is terminated while another is created.
To prevent re-use, we keep track of the last id returned
and ask for that id + 1 as a base for each query. We let
the idr spin lock protect this base id as well.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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When setting the autoclose timeout in jiffies there is a possible
integer overflow if the value in seconds is very large
(e.g. for 2^22 s with HZ=1024). The problem appears even on
64-bit due to the integer promotion rules. The fix is just a cast
to unsigned long.
Signed-off-by: Andrei Pelinescu-Onciul <andrei@iptel.org>
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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To avoid overflowing the maximum timer interval when transforming
the autoclose interval from seconds to jiffies, limit the maximum
autoclose value to MAX_SCHEDULE_TIMEOUT/HZ.
Signed-off-by: Andrei Pelinescu-Onciul <andrei@iptel.org>
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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Recently had a bug reported to me, in which the user was sending
packets with a payload containing a sequence number. The packets
were getting delivered in order according the chunk TSN values, but
the sequence values in the payload were arriving out of order. At
first I thought it must be an application error, but we eventually
found it to be a problem on the transmit side in the sctp stack.
The conditions for the error are that multihoming must be in use,
and it helps if each transport has a different pmtu. The problem
occurs in sctp_outq_flush. Basically we dequeue packets from the
data queue, and attempt to append them to the orrered packet for a
given transport. After we append a data chunk we add the trasport
to the end of a list of transports to have their packets sent at
the end of sctp_outq_flush. The problem occurs when a data chunks
fills up a offered packet on a transport. The function that does
the appending (sctp_packet_transmit_chunk), will try to call
sctp_packet_transmit on the full packet, and then append the chunk
to a new packet. This call to sctp_packet_transmit, sends that
packet ahead of the others that may be queued in the transport_list
in sctp_outq_flush. The result is that frames that were sent in one
order from the user space sending application get re-ordered prior
to tsn assignment in sctp_packet_transmit, resulting in mis-sequencing
of data payloads, even though tsn ordering is correct.
The fix is to change where we assign a tsn. By doing this earlier,
we are then free to place chunks in packets, whatever way we
see fit and the protocol will make sure to do all the appropriate
re-ordering on receive as is needed.
Signed-off-by: Neil Horman <nhorman@tuxdriver.com>
Reported-by: William Reich <reich@ulticom.com>
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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Current implementation of max.burst ends up limiting new
data during cwnd decay period. The decay is happening becuase
the connection is idle and we are allowed to fill the congestion
window. The point of max.burst is to limit micro-bursts in response
to large acks. This still happens, as max.burst is still applied
to each transmit opportunity. It will also apply if a very large
send is made (greater then allowed by burst).
Tested-by: Florian Niederbacher <florian.niederbacher@student.uibk.ac.at>
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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The transport last_time_used variable is rather useless.
It was only used when determining if CWND needs to be updated
due to idle transport. However, idle transport detection was
based on a Heartbeat timer and last_time_used was not incremented
when sending Heartbeats. As a result the check for cwnd reduction
was always true. We can get rid of the variable and just base
our cwnd manipulation on the HB timer (like the code comment sais).
We also have to call into the cwnd manipulation function regardless
of whether HBs are enabled or not. That way we will detect idle
transports if the user has disabled Heartbeats.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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SCTP_GET_*_OLD stuffs are schedlued to be removed.
Cc: Vlad Yasevich <vladislav.yasevich@hp.com>
Signed-off-by: WANG Cong <amwang@redhat.com>
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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Since draft-ietf-tsvwg-sctpsocket-15.txt, setting the
SPP_MTUD_ENABLE flag when changing pathmaxrxt via the
SCTP_PEER_ADDR_PARAMS setsockopt is not required any
longer.
Signed-off-by: Andrei Pelinescu-Onciul <andrei@iptel.org>
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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We currently send window update SACKs every time we free up 1 PMTU
worth of data. That a lot more SACKs then necessary. Instead, we'll
now send back the actuall window every time we send a sack, and do
window-update SACKs when a fraction of the receive buffer has been
opened. The fraction is controlled with a sysctl.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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When sctp_connectx() is used, we pick the first address as
primary, even though it may not have worked. This results
in excessive retransmits and poor performance. We should
select the address that the association was established with.
Reported-by: Thomas Dreibholz <dreibh@iem.uni-due.de>
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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The "Invalid Stream Identifier" error has a 16 bit reserved
field at the end, thus making the parameter length be 8 bytes.
We've never supplied that reserved field making wireshark
tag the packet as malformed.
Reported-by: Chris Dischino <cdischino@sonusnet.com>
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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This patch implement the sender side for SACK-IMMEDIATELY
extension.
Section 4.1. Sender Side Considerations
Whenever the sender of a DATA chunk can benefit from the
corresponding SACK chunk being sent back without delay, the sender
MAY set the I-bit in the DATA chunk header.
Reasons for setting the I-bit include
o The sender is in the SHUTDOWN-PENDING state.
o The application requests to set the I-bit of the last DATA chunk
of a user message when providing the user message to the SCTP
implementation.
Signed-off-by: Wei Yongjun <yjwei@cn.fujitsu.com>
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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This patch implement the receiver side for SACK-IMMEDIATELY
extension:
Section 4.2. Receiver Side Considerations
On reception of an SCTP packet containing a DATA chunk with the I-bit
set, the receiver SHOULD NOT delay the sending of the corresponding
SACK chunk and SHOULD send it back immediately.
Signed-off-by: Wei Yongjun <yjwei@cn.fujitsu.com>
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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master.kernel.org:/pub/scm/linux/kernel/git/davem/net-2.6
Conflicts:
drivers/net/can/Kconfig
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Recent commits
sctp: Get rid of an extra routing lookup when adding a transport
and
sctp: Set source addresses on the association before adding transports
changed when routes are added to the sctp transports. As such,
we didn't set the socket source address correctly when adding the first
transport. The first transport is always the primary/active one, so
when adding it, set the socket source address. This was causing
regression failures in SCTP tests.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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A new (unrealeased to the user) sctp_connectx api
c6ba68a26645dbc5029a9faa5687ebe6fcfc53e4
sctp: support non-blocking version of the new sctp_connectx() API
introduced a regression cought by the user regression test
suite. In particular, the API requires the user library to
re-allocate the buffer and could potentially trigger a SIGFAULT.
This change corrects that regression by passing the original
address buffer to the kernel unmodified, but still allows for
a returned association id.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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Recent commit 8da645e101a8c20c6073efda3c7cc74eec01b87f
sctp: Get rid of an extra routing lookup when adding a transport
introduced a regression in the connection setup. The behavior was
different between IPv4 and IPv6. IPv4 case ended up working because the
route lookup routing returned a NULL route, which triggered another
route lookup later in the output patch that succeeded. In the IPv6 case,
a valid route was returned for first call, but we could not find a valid
source address at the time since the source addresses were not set on the
association yet. Thus resulted in a hung connection.
The solution is to set the source addresses on the association prior to
adding peers.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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Avoid touching device refcount in sctp/ipv6, thanks to RCU
Signed-off-by: Eric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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struct can_proto had a capability field which wasn't ever used. It is
dropped entirely.
struct inet_protosw had a capability field which can be more clearly
expressed in the code by just checking if sock->type = SOCK_RAW.
Signed-off-by: Eric Paris <eparis@redhat.com>
Acked-by: Arnaldo Carvalho de Melo <acme@redhat.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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Adds RCU management to the list of netdevices.
Convert some for_each_netdev() users to RCU version, if
it can avoid read_lock-ing dev_base_lock
Ie:
read_lock(&dev_base_loack);
for_each_netdev(net, dev)
some_action();
read_unlock(&dev_base_lock);
becomes :
rcu_read_lock();
for_each_netdev_rcu(net, dev)
some_action();
rcu_read_unlock();
Signed-off-by: Eric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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In order to have better cache layouts of struct sock (separate zones
for rx/tx paths), we need this preliminary patch.
Goal is to transfert fields used at lookup time in the first
read-mostly cache line (inside struct sock_common) and move sk_refcnt
to a separate cache line (only written by rx path)
This patch adds inet_ prefix to daddr, rcv_saddr, dport, num, saddr,
sport and id fields. This allows a future patch to define these
fields as macros, like sk_refcnt, without name clashes.
Signed-off-by: Eric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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Create a new socket level option to report number of queue overflows
Recently I augmented the AF_PACKET protocol to report the number of frames lost
on the socket receive queue between any two enqueued frames. This value was
exported via a SOL_PACKET level cmsg. AFter I completed that work it was
requested that this feature be generalized so that any datagram oriented socket
could make use of this option. As such I've created this patch, It creates a
new SOL_SOCKET level option called SO_RXQ_OVFL, which when enabled exports a
SOL_SOCKET level cmsg that reports the nubmer of times the sk_receive_queue
overflowed between any two given frames. It also augments the AF_PACKET
protocol to take advantage of this new feature (as it previously did not touch
sk->sk_drops, which this patch uses to record the overflow count). Tested
successfully by me.
Notes:
1) Unlike my previous patch, this patch simply records the sk_drops value, which
is not a number of drops between packets, but rather a total number of drops.
Deltas must be computed in user space.
2) While this patch currently works with datagram oriented protocols, it will
also be accepted by non-datagram oriented protocols. I'm not sure if thats
agreeable to everyone, but my argument in favor of doing so is that, for those
protocols which aren't applicable to this option, sk_drops will always be zero,
and reporting no drops on a receive queue that isn't used for those
non-participating protocols seems reasonable to me. This also saves us having
to code in a per-protocol opt in mechanism.
3) This applies cleanly to net-next assuming that commit
977750076d98c7ff6cbda51858bb5a5894a9d9ab (my af packet cmsg patch) is reverted
Signed-off-by: Neil Horman <nhorman@tuxdriver.com>
Signed-off-by: Eric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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This provides safety against negative optlen at the type
level instead of depending upon (sometimes non-trivial)
checks against this sprinkled all over the the place, in
each and every implementation.
Based upon work done by Arjan van de Ven and feedback
from Linus Torvalds.
Signed-off-by: David S. Miller <davem@davemloft.net>
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Sizing of memory allocations shouldn't depend on the number of physical
pages found in a system, as that generally includes (perhaps a huge amount
of) non-RAM pages. The amount of what actually is usable as storage
should instead be used as a basis here.
Some of the calculations (i.e. those not intending to use high memory)
should likely even use (totalram_pages - totalhigh_pages).
Signed-off-by: Jan Beulich <jbeulich@novell.com>
Acked-by: Rusty Russell <rusty@rustcorp.com.au>
Acked-by: Ingo Molnar <mingo@elte.hu>
Cc: Dave Airlie <airlied@linux.ie>
Cc: Kyle McMartin <kyle@mcmartin.ca>
Cc: Jeremy Fitzhardinge <jeremy@goop.org>
Cc: Pekka Enberg <penberg@cs.helsinki.fi>
Cc: Hugh Dickins <hugh.dickins@tiscali.co.uk>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: Patrick McHardy <kaber@trash.net>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
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Signed-off-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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Remove long removed "inet_protocol_base" declaration.
Signed-off-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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Since our TSN map is capable of holding at most a 4K chunk gap,
there is no way that during this gap, a stream sequence number
(unsigned short) can wrap such that the new number is smaller
then the next expected one. If such a case is encountered,
this is a protocol violation.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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Use sctp_packet_reset() instead of dup code.
Signed-off-by: Wei Yongjun <yjwei@cn.fujitsu.com>
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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This patch introduces a new sysctl option to make IPv4 Address Scoping
configurable <draft-stewart-tsvwg-sctp-ipv4-00.txt>.
In networking environments where DNAT rules in iptables prerouting
chains convert destination IP's to link-local/private IP addresses,
SCTP connections fail to establish as the INIT chunk is dropped by the
kernel due to address scope match failure.
For example to support overlapping IP addresses (same IP address with
different vlan id) a Layer-5 application listens on link local IP's,
and there is a DNAT rule that maps the destination IP to a link local
IP. Such applications never get the SCTP INIT if the address-scoping
draft is strictly followed.
This sysctl configuration allows SCTP to function in such
unconventional networking environments.
Sysctl options:
0 - Disable IPv4 address scoping draft altogether
1 - Enable IPv4 address scoping (default, current behavior)
2 - Enable address scoping but allow IPv4 private addresses in init/init-ack
3 - Enable address scoping but allow IPv4 link local address in init/init-ack
Signed-off-by: Bhaskar Dutta <bhaskar.dutta@globallogic.com>
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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We used to perform 2 routing lookups for a new transport: one
just for path mtu detection, and one to actually route to destination
and path mtu update when sending a packet. There is no point in doing
both of them, especially since the first one just for path mtu doesn't
take into account source address and sometimes gives the wrong route,
causing path mtu updates anyway.
We now do just the one call to do both route to destination and get
path mtu updates.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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We currently track if AUTH has been bundled using the 'auth'
pointer to the chunk. However, AUTH is disallowed after DATA
is already in the packet, so we need to instead use the
'has_auth' field.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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The packet information does not reset after packet transmit, this
may cause some problems such as following DATA chunk be sent without
AUTH chunk, even if the authentication of DATA chunk has been
requested by the peer.
Signed-off-by: Wei Yongjun <yjwei@cn.fujitsu.com>
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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Add-IP feature allows users to delete an active transport. If that
transport has chunks in flight, those chunks need to be moved to another
transport or association may get into unrecoverable state.
Reported-by: Rafael Laufer <rlaufer@cisco.com>
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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We had a bug that we never stored the user-defined value for
MAXSEG when setting the value on an association. Thus future
PMTU events ended up re-writing the frag point and increasing
it past user limit. Additionally, when setting the option on
the socket/endpoint, we effect all current associations, which
is against spec.
Now, we store the user 'maxseg' value along with the computed
'frag_point'. We inherit 'maxseg' from the socket at association
creation and use it as an upper limit for 'frag_point' when its
set.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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SCTP will delay the last part of a large write due to NAGLE, if that
part is smaller then MTU. Since we are doing large writes, we might
as well send the last portion now instead of waiting untill the next
large write happens. The small portion will be sent as is regardless,
so it's better to not delay it.
This is a result of much discussions with Wei Yongjun <yjwei@cn.fujitsu.com>
and Doug Graham <dgraham@nortel.com>. Many thanks go out to them.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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The decision to delay due to Nagle should be based on the path mtu
and future packet size. We currently incorrectly base it on
'frag_point' which is the SCTP DATA segment size, and also we do
not count DATA chunk header overhead in the computation. This
actuall allows situations where a user can set low 'frag_point',
and then send small messages without delay.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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We currently set a_rwnd to 0 when faking a SACK from SHUTDOWN.
This results in an hung association if the remote only uses
SHUTDOWNs (which it's allowed to do) to acknowlege DATA when
closing. The reason for that is that we simply honor the a_rwnd
from the sack, but since we faked it to be 0, we enter 0-window
probing. The fix is to use the peers old rwnd and add our flight
size to it.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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SCTP has a problem that when small chunks are used, it is possible
to exhaust the receiver buffer without fully closing receive window.
This happens due to all overhead that we have account for with small
messages. To fix this, when receive buffer is exceeded, we'll drop
the window to 0 and save the 'drop' portion. When application starts
reading data and freeing up recevie buffer space, we'll wait until
we've reached the 'drop' window and then add back this 'drop' one
mtu at a time. This worked well in testing and under stress produced
rather even recovery.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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If T3 timer expires, we are retransmitting data due to timeout any
any fast recovery is null and void. We can clear the fast recovery
flag.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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SCTP RFC 4960 states that unacknowledged HEARTBEATS count as
errors agains a given transport or endpoint. As such, we
should increment the error counts for only for unacknowledged
HB, otherwise we detect failure too soon. This goes for both
the overall error count and the path error count.
Now, there is a difference in how the detection is done
between the two. The path error detection is done after
the increment, so to detect it properly, we actually need
to exceed the path threshold. The overall error detection
is done _BEFORE_ the increment. Thus to detect the failure,
it's enough for the error count to match the threshold.
This is why all the state functions use '>=' to detect failure,
while path detection uses '>'.
Thanks goes to Chunbo Luo <chunbo.luo@windriver.com> who first
proposed patches to fix this issue and made me re-read the spec
and the code to figure out how this cruft really works.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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create_proc_entry() is deprecated (not formally, though).
Signed-off-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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The receiver of the HEARTBEAT should respond with a HEARTBEAT ACK
that contains the Heartbeat Information field copied from the
received HEARTBEAT chunk. So the received HEARTBEAT-ACK chunk
must have a length of:
sizeof(sctp_chunkhdr_t) + sizeof(sctp_sender_hb_info_t)
A badly formatted HB-ACK chunk, it is possible that we may access
invalid memory. We should really make sure that the chunk format
is what we expect, before attempting to touch the data.
Signed-off-by: Wei Yongjun <yjwei@cn.fujitsu.com>
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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If Cumulative TSN Ack field of SHUTDOWN chunk is less than the
Cumulative TSN Ack Point then drop the SHUTDOWN chunk.
Signed-off-by: Wei Yongjun <yjwei@cn.fujitsu.com>
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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Currenlty, sctp breaks up user messages into fragments and
sends each fragment to the lower layer by itself. This means
that for each fragment we go all the way down the stack
and back up. This also discourages bundling of multiple
fragments when they can fit into a sigle packet (ex: due
to user setting a low fragmentation threashold).
We introduce a new command SCTP_CMD_SND_MSG and hand the
whole message down state machine. The state machine and
the side-effect parser will cork the queue, add all chunks
from the message to the queue, and then un-cork the queue
thus causing the chunks to get transmitted.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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If the association has a SACK timer pending and now DATA queued
to be send, we'll try to bundle the SACK with the next application send.
As such, try encourage bundling by accounting for SACK in the size
of the first chunk fragment.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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We are now trying to bundle SACKs when we have outbound
DATA to send. However, there are situations where this
outbound DATA will not be sent (due to congestion or
available window). In such cases it's ok to wait for the
timer to expire. This patch refactors the sending code
so that betfore attempting to bundle the SACK we check
to see if the DATA will actually be transmitted.
Based on eirlier works for Doug Graham <dgraham@nortel.com> and
Wei Youngjun <yjwei@cn.fujitsu.com>.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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Since an application may specify the maximum SCTP fragment size
that all data should be fragmented to, we need to fix how
we do segmentation. Right now, if a user specifies a small
fragment size, the segment size can go negative in the presence
of AUTH or COOKIE_ECHO bundling.
What we need to do is track the largest possbile DATA chunk that
can fit into the mtu. Then if the fragment size specified is
bigger then this maximum length, we'll shrink it down. Otherwise,
we just use the smaller segment size without changing it further.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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If a socket has a lot of association that are in the process of
of being closed/aborted, it is possible for a remote to establish
new associations during the time period that the old ones are shutting
down. If this was a result of a close() call, there will be no socket
and will cause a memory leak. We'll prevent this by setting the
socket state to CLOSING and disallow new associations when in this state.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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This patch corrects the conditions under which a SACK will be piggybacked
on a DATA packet. The previous condition was incorrect due to a
misinterpretation of RFC 4960 and/or RFC 2960. Specifically, the
following paragraph from section 6.2 had not been implemented correctly:
Before an endpoint transmits a DATA chunk, if any received DATA
chunks have not been acknowledged (e.g., due to delayed ack), the
sender should create a SACK and bundle it with the outbound DATA
chunk, as long as the size of the final SCTP packet does not exceed
the current MTU. See Section 6.2.
When about to send a DATA chunk, the code now checks to see if the SACK
timer is running. If it is, we know we have a SACK to send to the
peer, so we append the SACK (assuming available space in the packet)
and turn off the timer. For a simple request-response scenario, this
will result in the SACK being bundled with the response, meaning the
the SACK is received quickly by the client, and also meaning that no
separate SACK packet needs to be sent by the server to acknowledge the
request. Prior to this patch, a separate SACK packet would have been
sent by the server SCTP only after its delayed-ACK timer had expired
(usually 200ms). This is wasteful of bandwidth, and can also have a
major negative impact on performance due the interaction of delayed ACKs
with the Nagle algorithm.
Signed-off-by: Doug Graham <dgraham@nortel.com>
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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When the sctp transport is marked down, we can release the
cached route and force a new lookup when attempting to use
this transport for anything. This way, if a better route
or source address is available, we'll try to use it.
Signed-off-by: Vlad Yasevich <vladislav.yasevich@hp.com>
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