1 Introduction to Analog delay transferNetem and Tc:netem are a network emulation function module provided by Linux 2.6 and above kernel versions. The function module can be used to simulate complex Internet transmission performance in a well-performing LAN, such as low bandwidth, transmission delay, packet loss and so on. Many distributions of Linux with Linux 2
1 Introduction to Analog delay transmission
Netem and Tc:netem are a network emulation module provided by the Linux kernel version 2.6 and above. This function module can be used to simulate complex Internet transmission performance in a well performing LAN, such as low bandwidth, transmission delay, packet loss, and so on. Many distributions Linux that use the L
lower layer. If pending is returned, release our mypacket in ptsendcomplete; otherwise, the mypacket will be released immediately in this function.
3. When the lower-level miniport driver sends mypacket, it will call ndismsendcomplete
4. NDIS then calls passthru's ptsendcomplete. In this function, we should release mypacket and notify the upper-layer protocol driver to release their packet.
Packet
Bandwidth or flow rate unit:
Kbps
Kbyte/sec
Kilobytes per second
Mbps
Mbyte/sec
Megabytes per second
Kbit
Kbits/sec
Mbit
Mbits/sec
1 , Analog Delay Transfer# TC Qdisc Add dev eth0 root netem delay 100msThis command sets the transmission of the eth0 NIC to a delay of 100 milliseconds to send.In a more realistic case, the delay value will not be so accurate, there will be a certain flu
UDP (packet length, packet collection capability, packet loss and process structure selection)UDP Packet Length: the theoretical length of a UDP packet
What is the theoretical length of udp data packets and what is the proper udp
good.Internet programming is different because routers on the Internet may set the MTU to a different value. If we assume that the MTU is sending data to 1500来, and the MTU value of a network passing through is less than 1500 bytes, then the system will use a series of mechanisms to adjust the MTU value so that the datagram can reach its destination smoothly. Because the standard MTU value on the Internet is 576 bytes, it is best to use UDP data-length controls within 548 bytes (576-8-20) for U
between two routers, regardless of packet length or link transfer rate.
queue delay and packet loss
Queue Delay
ratio LA/R is called flow intensityIf la/r >1, the average rate at which the bit arrives in the queue exceeds the rate at which the queue is transferred, the increase in the queue tends to be unbounded, and the queuing delay tends to be infinite. The
in front of the Libpcap capture packet, especially in the gigabit network conditions, a large number of drops, online search for a long time, probably all Pf_packet +mmap,napi,pf_ring and other methods, I pf_ring+libpcap experiment, The detection of gigabit network conditions, capturing the performance of the packet is very good, almost no packet
For an online chat window provided by slice, use httpanalyzer to analyze the packet sent by the window, and then simulate the packet to send the message.
The headers obtained through the software are as follows:
We can get the post address through the header:/chats/sendmessage. ashx page, and then cookie.
The data of post is as follows:
For example, the
expected ID, it is determined that packet loss occurs. The packet loss ID is sent back to the server. When the server receives a packet loss response, it resends the lost packet.
It i
This is a BDP test in rhca, which is also a very common simulation of latency and packet loss. I will share it with you now.
We also use the application software to test TCP/UDP comparison. When we test the impact of BDP on TCP/IP, we all need some network latency and packet loss simulation, many commercial software ca
I don't know how to say it. In short, the boat, from the mouth, I can not see HUANGFA and impoverished! I'm not going to say anything except cursing!Prior to BBR, there are two kinds of congestion control algorithms, based on packet loss and delay-based, regardless of which is based on detection, in other words, packet loss
Transferred from: http://cizixs.com/2018/01/13/linux-udp-packet-drop-debug?hmsr=toutiao.ioutm_medium=toutiao.ioutm_ Source=toutiao.ioRecent work encountered a server application UDP packet loss, in the process of reviewing a lot of information, summed up this article, for more people to refer to.Before we get started, we'll use a graph to explain the process of r
Simulate the network packet sending and receiving delay for the specified IP port on Linux.
When compiling network applications, we usually test and debug the network on a LAN or even a local machine. Is there a way to simulate a complex Internet environment-especially network latency-for applications without modifying their own code in an Intranet environment wi
this This paper mainly introduces the realization of WEBRTC in Nack, Weizhenwei, the article was first published in the Wind network , Id:befoioSupport original, reprint must indicate the source, welcome attention to my public number blacker (Id:blackerteam or WEBRTCORGCN).In WEBRTC, forward error correction (FEC) and packet loss retransmission (NACK) are important methods to resist network errors. FEC adds
Routing RingNetwork Packet Loss
This is an actual case of analyzing the causes of a large number of packet loss on the network. The user's network packet loss is very serious, causing a lot of trouble to the user, we try to analy
How should I avoid packet loss when a large amount of data is continuously transmitted over TCP?
For example, sending a file. I remember someone mentioned the possible stack overflow. How can this problem be avoided? Can I send a confirmation packet after receiving the data? After receiving the confirmation packet
The network port uses the 1000M rate time to appear the network communication loses the packet +idc the computer room managed server communication is not smooth.
Network failure:
Switch port 1000M, network card is 1000M, NIC configuration is normal. You lose the packet at the interval of ping.
The performance is packet los
Let's start by recognizing what a packet loss is, and what kind of phenomenon is being lost to the network:
Data is transmitted on the Internet on a packet-by-unit basis, with a packet of NK, no more, no less. That is to say, no matter how good the network line is, how strong the network equipment, your da
Link: http://blog.chinaunix.net/u3/105477/showart_2087878.html
Key points:Learn how to compile the sampling process, how to obtain the required parameters, how to use individual files for recording, understand the specific physical meanings of throughput, packet loss rate, and end-to-end latency, and learn more about the parameter interfaces provided by NS2! [Scenario description]: four of the eight nodes
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