Lookup calls would spin out of control when network connectivity was
lost. The throttling that was in place only took effect when the table
returned zero results, which doesn't happen very often.
The new throttling should not have a negative impact when the host is
online. Lookups against the network take some time and dials for all
results must complete or hit the cache before a new one is started. This
usually takes longer than four seconds, leaving online lookups
unaffected.
Fixes#1296
As of this commit, we no longer rely on the protocol handler to report
write errors in a timely fashion. When a write fails, shutdown is
initiated immediately and no new writes can start. This will also
prevent new writes from starting after Server.Stop has been called.
The most visible change is event-based dialing, which should be an
improvement over the timer-based system that we have at the moment.
The dialer gets a chance to compute new tasks whenever peers change or
dials complete. This is better than checking peers on a timer because
dials happen faster. The dialer can now make more precise decisions
about whom to dial based on the peer set and we can test those
decisions without actually opening any sockets.
Peer management is easier to test because the tests can inject
connections at checkpoints (after enc handshake, after protocol
handshake).
Most of the handshake stuff is now part of the RLPx code. It could be
exported or move to its own package because it is no longer entangled
with Server logic.
The previous limit was 10MB which is unacceptable for all kinds
of reasons, the most important one being that we don't want to
allow the remote side to make us allocate 10MB at handshake time.
The returned reason is currently not used except for the log
message. This change makes the log messages a bit more useful.
The handshake code also returns the remote reason.
People stil get confused about the messages. This commit changes
the levels so that the only thing printed at the default level (info)
is a successful mapping.
The test listens for multicast UDP packets on the default interface
because I couldn't get it to work reliably on loopback without massive
changes to goupnp. This means that the test might fail when there is a
UPnP-enabled router attached on that interface. I checked that locally
by looping the test and it passes reliably because the local SSDP server
always responds faster.
Concurrent calls to Interface methods on autodisc could return a "not
discovered" error if the discovery did not finish before the call.
autodisc.wait expected the done channel to carry the found Interface
but it was closed instead.
The fix is to use sync.Once for now, which is easier to get right.
And there is a test. Finally.
This will have to change again when we introduce re-discovery.
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
