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Closes #2241: Use Keccak-256 from golang.org/x/crypto/sha3 and mention explicitly
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As we aren't really using the standarized SHA-3
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On Windows, UDPConn.ReadFrom returns an error for packets larger
than the receive buffer. The error is not marked temporary, causing
our loop to exit when the first oversized packet arrived. The fix
is to treat this particular error as temporary.
Fixes: #1579, #2087
Updates: #2082
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The test expected the timeout to fire after a matcher for the response
was added, but the timeout is random and fired sooner sometimes.
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This change simplifies the dial scheduling logic because it
no longer needs to track whether the discovery table has been
bootstrapped.
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thanks to Felix Lange (fjl) for help with design & impl
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The strict matching can get in the way of protocol upgrades.
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nodeDB.querySeeds was not safe for concurrent use but could be called
concurrenty on multiple goroutines in the following case:
- the table was empty
- a timed refresh started
- a lookup was started and initiated refresh
These conditions are unlikely to coincide during normal use, but are
much more likely to occur all at once when the user's machine just woke
from sleep. The root cause of the issue is that querySeeds reused the
same leveldb iterator until it was exhausted.
This commit moves the refresh scheduling logic into its own goroutine
(so only one refresh is ever active) and changes querySeeds to not use
a persistent iterator. The seed node selection is now more random and
ignores nodes that have not been contacted in the last 5 days.
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fdtrack: hide message
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This reverts commit 5c949d3b3ba81ea0563575b19a7b148aeac4bf61.
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PR #1621 changed Table locking so the mutex is not held while a
contested node is being pinged. If multiple nodes ping the local node
during this time window, multiple ping packets will be sent to the
contested node. The changes in this commit prevent multiple packets by
tracking whether the node is being replaced.
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Might solve #1579
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If the timeout fired (even just nanoseconds) before the deadline of the
next pending reply, the timer was not rescheduled. The timer would've
been rescheduled anyway once the next packet was sent, but there were
cases where no next packet could ever be sent due to the locking issue
fixed in the previous commit.
As timing-related bugs go, this issue had been present for a long time
and I could never reproduce it. The test added in this commit did
reproduce the issue on about one out of 15 runs.
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Table.mutex was being held while waiting for a reply packet, which
effectively made many parts of the whole stack block on that packet,
including the net_peerCount RPC call.
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Not closing the table used to be fine, but now the table has a database.
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Package fdtrack logs statistics about open file descriptors.
This should help identify the source of #1549.
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I forgot to update one instance of "go-ethereum" in commit 3f047be5a.
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All code outside of cmd/ is licensed as LGPL. The headers
now reflect this by calling the whole work "the go-ethereum library".
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rand.Source isn't safe for concurrent use.
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This should increase the speed a bit because all findnode
results (up to 16) can be verified at the same time.
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The code assumed that Table.closest always returns at least 13 nodes.
This is not true for small tables (e.g. during bootstrap).
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neighbours packets.
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We don't have a UDP which specifies any messages that will be 4KB. Aside from being implemented for months and a necessity for encryption and piggy-backing packets, 1280bytes is ideal, and, means this TODO can be completed!
Why 1280 bytes?
* It's less than the default MTU for most WAN/LAN networks. That means fewer fragmented datagrams (esp on well-connected networks).
* Fragmented datagrams and dropped packets suck and add latency while OS waits for a dropped fragment to never arrive (blocking readLoop())
* Most of our packets are < 1280 bytes.
* 1280 bytes is minimum datagram size and MTU for IPv6 -- on IPv6, a datagram < 1280bytes will *never* be fragmented.
UDP datagrams are dropped. A lot! And fragmented datagrams are worse. If a datagram has a 30% chance of being dropped, then a fragmented datagram has a 60% chance of being dropped. More importantly, we have signed packets and can't do anything with a packet unless we receive the entire datagram because the signature can't be verified. The same is true when we have encrypted packets.
So the solution here to picking an ideal buffer size for receiving datagrams is a number under 1400bytes. And the lower-bound value for IPv6 of 1280 bytes make's it a non-decision. On IPv4 most ISPs and 3g/4g/let networks have an MTU just over 1400 -- and *never* over 1500. Never -- that means packets over 1500 (in reality: ~1450) bytes are fragmented. And probably dropped a lot.
Just to prove the point, here are pings sending non-fragmented packets over wifi/ISP, and a second set of pings via cell-phone tethering. It's important to note that, if *any* router between my system and the EC2 node has a lower MTU, the message would not go through:
On wifi w/normal ISP:
localhost:Debug $ ping -D -s 1450 52.6.250.242
PING 52.6.250.242 (52.6.250.242): 1450 data bytes
1458 bytes from 52.6.250.242: icmp_seq=0 ttl=42 time=104.831 ms
1458 bytes from 52.6.250.242: icmp_seq=1 ttl=42 time=119.004 ms
^C
--- 52.6.250.242 ping statistics ---
2 packets transmitted, 2 packets received, 0.0% packet loss
round-trip min/avg/max/stddev = 104.831/111.918/119.004/7.087 ms
localhost:Debug $ ping -D -s 1480 52.6.250.242
PING 52.6.250.242 (52.6.250.242): 1480 data bytes
ping: sendto: Message too long
ping: sendto: Message too long
Request timeout for icmp_seq 0
ping: sendto: Message too long
Request timeout for icmp_seq 1
Tethering to O2:
localhost:Debug $ ping -D -s 1480 52.6.250.242
PING 52.6.250.242 (52.6.250.242): 1480 data bytes
ping: sendto: Message too long
ping: sendto: Message too long
Request timeout for icmp_seq 0
^C
--- 52.6.250.242 ping statistics ---
2 packets transmitted, 0 packets received, 100.0% packet loss
localhost:Debug $ ping -D -s 1450 52.6.250.242
PING 52.6.250.242 (52.6.250.242): 1450 data bytes
1458 bytes from 52.6.250.242: icmp_seq=0 ttl=42 time=107.844 ms
1458 bytes from 52.6.250.242: icmp_seq=1 ttl=42 time=105.127 ms
1458 bytes from 52.6.250.242: icmp_seq=2 ttl=42 time=120.483 ms
1458 bytes from 52.6.250.242: icmp_seq=3 ttl=42 time=102.136 ms
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The previous metric was pubkey1^pubkey2, as specified in the Kademlia
paper. We missed that EC public keys are not uniformly distributed.
Using the hash of the public keys addresses that. It also makes it
a bit harder to generate node IDs that are close to a particular node.
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This commit changes the discovery protocol to use the new "v4" endpoint
format, which allows for separate UDP and TCP ports and makes it
possible to discover the UDP address after NAT.
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This a fix for an attack vector where the discovery protocol could be
used to amplify traffic in a DDOS attack. A malicious actor would send a
findnode request with the IP address and UDP port of the target as the
source address. The recipient of the findnode packet would then send a
neighbors packet (which is 16x the size of findnode) to the victim.
Our solution is to require a 'bond' with the sender of findnode. If no
bond exists, the findnode packet is not processed. A bond between nodes
α and β is created when α replies to a ping from β.
This (initial) version of the bonding implementation might still be
vulnerable against replay attacks during the expiration time window.
We will add stricter source address validation later.
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The primary motivation for doing this right now is that old PoC 8
nodes and newer PoC 9 nodes keep discovering each other, causing
handshake failures.
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Range expressions capture the length of the slice once before the first
iteration. A range expression cannot be used here since the loop
modifies the slice variable (including length changes).
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udp.Table was assigned after the readLoop started, so
packets could arrive and be processed before the Table was there.
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The discovery RPC protocol does not yet distinguish TCP and UDP ports.
But it can't hurt to do so in our internal model.
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