As long as I know, when a connection request SYN segment arrives, TCP will response with an SYN&ACK segment and mark the connection incomplete(SYN_RCVD) and put it into an incomplete queue, waiting for the ACK to complete the connection.
But I think it won't always stay in the queue, is there a timeout on it?
On my Ubuntu system:
# cat /proc/sys/net/ipv4/tcp_synack_retries
5
A tcp_synack_retries variable is responsible for controlling the number of retransmissions in Linux operating system. Its default value is set to 5 for most Linux operating systems, which causes the half-open connection to be removed after 3 minutes. In the below table there are calculations for other values.
Value Time of retransmission Total time to keep half-open connections in the backlog queue
1 in 3rd second 9 seconds
2 in 3rd and 9th second 21 seconds
3 in 3rd , 9th and 21st second 45 seconds
Related
I know this is a very basic question but i was not aware since i'm new to SNMP stuff.So, is there any minimum timeout for SNMP based on version if so can you please specify the version and the timeout in seconds.
As #LexLi mentioned, You need to be more specific about which timeouts are You asking. And, he is right, there is no dependency between version and timeout.
snmpcmds like snmpwalk\snmpget\snmpset... have timeout
((retries+1) x timeout_between_retries) seconds, which is (5+1) x 1 = 6 seconds
by default. Both parameters could be changed by either command line
or snmp.conf.
Example:
snmpwalk -v2c -cpublic -r1 -t10 host OID
The maximum timeout here will be 20 seconds because we will send 2 queries with the timeout between them is 10 seconds.
Also there is agentXTimeout and agentXRetries, which are defining timeout for AgentX requests between masterAgent and subAgent. Values are the same: 5 times and 1 second ,respectively. Those values could be changed by the command line or in the snmpd.conf.
While using the UNIX traceroute command, in order to calculate average latency for each loop (from one hop to the next one, e.g.: hop 8 to 9) what procedure should we take?
8 146.97.33.6 2.150 ms 2.159 ms 2.133 ms
9 146.97.33.61 1.580 ms 1.543 ms 1.552 ms
10 146.97.35.170 1.544 ms 1.535 ms 1.526 ms
I am aware, for instance, the average latency for hop 9, is 1.5583:
1.580 ms+1.543 ms+1.552 ms / 3
However, is this the average time it takes from the local host to that particular hop, or is it the time it takes for data packets to travel from previous hop to the particular hop?
The latency is the round-trip latency from the originating host to the hop where it times out and back to the originating host, but it includes the time it takes for the timeout hop to get around to generating an ICMP message back to the originating host.
The primary purpose of a router is to route packets as fast as it can. Generating ICMP messages is a much lower priority for the router. If the router is busy routing packets, it will get around to generating the ICMP message when it has some spare time.
That is why you can see the times for some intermediate hops to be much longer than it is for the full path.
According to Wikipedia it looks like it is the former
the route is recorded as the round-trip times of the packets received
from each successive host (remote node) in the route (path); the sum
of the mean times in each hop is a measure of the total time spent to
establish the connection.
The answer is "from the local host to that particular hop"
I use jmeter to test a tomcat web application running behind f5's load balancer.
From the pcap file I captured in jmeter node I see that there is a duplicate ack and then
16 seconds later the client retransmit the packet and do nothing within 20 seconds.
As tomcat's default socket timeout is set as 20 seconds the client received Http error.
My question is:
What could be the reason that make client's TCP stack retransmit after 16 seconds and later do nothing within 20 seconds? Is it too busy? Is there a way to find out the reason?
From the HAProxy documentation on client timeouts:
It is a good practice to cover one or several TCP packet losses by
specifying timeouts that are slightly above multiples of 3 seconds
(eg: 4 or 5 seconds).
That seems like an arbitrary number. What is the significance of the 3 second figure?
It appears this is the default TCP retransmission timeout. From this Microsoft KB article:
TCP starts a re-transmission timer when each outbound segment is
handed down to IP. If no acknowledgment has been received for the data
in a given segment before the timer expires, then the segment is
retransmitted, up to the TcpMaxDataRetransmissions times. The default
value for this parameter is 5.
The re-transmission timer is initialized to 3 seconds when a TCP
connection is established; however it is adjusted "on the fly" to
match the characteristics of the connection using Smoothed Round Trip
Time (SRTT) calculations as described in RFC793. The timer for a given
segment is doubled after each re-transmission of that segment. Using
this algorithm, TCP tunes itself to the "normal" delay of a
connection. TCP connections over high-delay links will take much
longer to time out than those over low- delay links.
I am reading 《Internetworking with TCP/IP》 by Douglas Comer,and when talking about creating a tcp connection ,there is a problem:
Suppose an implementation of TCP use
initial sequence number 1 when it
creates a connection,Explain how a
system crash and restart can confuse a
remote system into believing that the
old connection remained open.
I can't figure out why,please help me,Thanks.
Consider why a connection may get duplicate sequence numbers normally.
Then consider how the receiving system would handle a packet with a "duplicate" sequence number (because the transmitting system started reusing sequence numbers in packets it is using try to re-establish a connection).
*Edit: *
OP says:
but when re-establish the connection,the transmitting system will send a segment with SYN code bit set(and the sequence number be set 1 of course),won't that(SYN code bit set) inform the receiving system it is a new connection trying to be established ?see wiki for Transmission_Control_Protocol,it says that "Only the first packet sent from each end should have this flag(SYN) set."
But packets get lost and delayed and arrive out of order. You can't simply say everything arriving after the packet with the SYN flag is new. Lets say some of the old packets are delayed and arrive after establishment of a new connection. How do you distinguish whether a packet with sequence number #10 is from the old connection or new one? The worse case scenario is that it's from the old connection and the receiving system accepts it as from the new connection. When the real new connection packet #10 arrives, it's ignored as an unnecessary retranmission. The stream is corrupted without any indication of it.
http://www.tcpipguide.com/free/t_TCPConnectionEstablishmentSequenceNumberSynchroniz.htm
... The problem with starting off each connection with a sequence number of 1 is that it introduces the possibility of segments from different connections getting mixed up. Suppose we established a TCP connection and sent a segment containing bytes 1 through 30. However, there was a problem with the internetwork that caused this segment to be delayed, and eventually, the TCP connection itself to be terminated. We then started up a new connection and again used a starting sequence number of 1. As soon as this new connection was started, however, the old segment with bytes labeled 1 to 30 showed up. The other device would erroneously think those bytes were part of the new connection.
... This is but one of several similar problems that can occur. ...
The other issue with a predictable initial sequence number, such as starting at 1 every time, is that the predictability presents a vulnerability:
A malicious person could write code to analyze ISNs and then predict the ISN of a subsequent TCP connection based on the ISNs used in earlier ones. This represents a security risk, which has been exploited in the past (such as in the case of the famous Mitnick attack). To defeat this, implementations now use a random number in their ISN selection process.
Mitnick attack - http://www.cas.mcmaster.ca/wiki/index.php/The_Mitnick_attack
It's far worse than that though anyway - being predictable with sequence numbers makes spoofing and injecting an order of magnitude easier
After restart, if the first TCP connection is towards the same remote system, and since the sequence number will again be 1 - consider what that will cause.