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34

As others have pointed out, Google DNS uses IP anycast, which allows multiple servers in multiple locations to effectively share an IP address. Google (like many others) has many servers around the world that will respond to 8.8.8.8. So when you ping that address, the server that responds is the one closest to you, perhaps one in your country. Note that ...


27

Network latency is how long it takes for something sent from a source host to reach a destination host. There are many components to latency, and the latency can actually be different A to B and B to A. The round trip time is how long it takes for a request sent from a source to a destination, and for the response to get back to the original source. ...


19

Google in particular uses distributed datacenters around the globe. They announce the same IP network at various places and due to the way routing protocols work, you reach the nearest one. This is called anycast bbc.co.uk points to an IP address that belongs to Fastly, Inc, a content delivery network, that also has points of presence around the world, ...


16

No, they won't for typical internet connectivity. You won't even be able to measure it using typical tools. For DC environments, we're fighting for every microsecond, but for typical ethernet switches, you get from 5-7 microseconds to 30-50 microseconds per hop (switch). Even 30-50 microseconds is too low to notice; typically latency is problematic for ...


16

You're overthinking. The number of lanes used doesn't really matter. Whether you transport 50 Gbit/s over 1, 2, or 5 lanes, the serialization delay is 20 ps/bit. So, you'd get 5 bits every 100 ps, regardless of the lanes used. Splitting data into lanes and recombining it takes place in the PCS sublayer and is invisible even on top of the physical layer. ...


14

The part that does the division to multiple lanes is called Physical Coding Sublayer in IEEE 802.3ba standard. This presentation by Gary Nicholl gives a good overview of it. The short explanation is that the data is divided to multiple lanes in blocks of 64 bits each (encoded on wire as 66 bits for clock recovery). Therefore as soon as packet size exceeds N*...


13

Pre-ACKing is only possible because the near accelerator keeps the segment data in queue. Therefore, it can ACK in stead of the far destination, prompting the sender to advance the send window and send out more segments than would be possible by delay-bandwidth alone. Of course, the near accelerator needs to keep track of its internal, larger send window ...


12

On Cisco devices, you can use Cisco IP SLA. You need to first configure and enable it, and then monitor the results. Steps: 1. ip sla monitor operation-number 2. type echo protocol ipIcmpEcho {destination-ip-address | destination-hostname} [source-ipaddr {ip-address | hostname} | source-interface interface-name] 3. frequency seconds 4. ip sla monitor ...


12

Hi and welcome to Network Engineering. As for "delay" vs "latency": The terms are not always used consistently. Some hints may be found here. I think generally, the term latency is used when looking at end-to-end times for one direction, which essentially are composed of the sum of all propagation, serialization, buffering (and possibly processing) delays ...


11

"Latency" can mean different things. Generally, it's a delay of some sort - application latency is the reaction time of an application (from input to output), network latency the delay for getting a packet from point A to B and so on. "Round-trip time" is more or less well defined as the network delay from point A to B and back . This is the sum of all ...


9

Latency is the amount of time it takes a packet of data to leave your computer and receive a response back from the end point. That is why this is measure in time. This is key for time sensitive applications like VoIP and video conferencing. Bandwidth/data rate is the amount of data (bits) you can upload or download in a given time (seconds). This is key ...


9

One way to do this is ICMP Timestamp, which is milliseconds from midnight UTC. It has the added benefit that you don't necessarily need to control both ends, as long as the far-end is not firewalled, there is good chance it'll work. However, to have reliable one-way measurements, you need reliably same time in both ends. As ICMP timestamp only have ...


9

Does increasing the bandwidth on a link from lets say 1mb to 30mb reduce the RTT? In short, yes; you are changing serialization delay; at 1Mbps the serialization delay is non-trivial. Compare the serialization delay for a 1500 Byte packet at 1Mbps and 30Mbps: 1500 Bytes * 8 bits/Byte / 1,000,000 bits/second = 12 milliseconds (at 1Mbps) 1500 Bytes * 8 ...


8

The question mentions that this is not hardware specific, however different platforms have different switching latencies and switching modes. Cut through switching is the fastest but lets through fragments. Fragment free is the second fastest and makes sure that frames are at least 64 bytes meaning they are not runts which is one of the signs of collisions. ...


8

Twisted pair cable used for the local loop has a velocity factor of about .58 - each km of cable takes ca. 6 µs to travel, adding ca. 12 µs to latency. The rest of the Internet probably uses fiber with a VF of .67, resulting in ca. 10 µs or .01 ms latency per km. ADSL has a basic encoding latency of around 10 ms. However, your ISP may likely interleave DSL ...


8

Yes, a bridge / switch adds some delay to a frame - in the order of 1 to 20 µs. For switches you usually speak of latency - the delay between receiving a frame and forwarding it out another port. A switch requires some time to receive the destination address and make the forwarding decision. Store-and-forward switches (the common kind) need to receive the ...


8

A cable half-way around the globe has a minimum latency of 100 ms, 200 ms round-trip (20,000 km distance / 200,000 km/s signal speed) - that's the physical limit.[1] In reality, links aren't as the crow flies (at all) and there are additional, active components in between, adding to latency - a more realistic figure is 150 or more one-way. Anything with ...


8

IP geolocation databases essentially represent a geolocation company's guess as to where an IP is located. Often they are based on the addresses in whois registrations, sometimes supplemented with the geolocation company's own research. For end-user IPs they are usually accurate to country level but for infrastructure IPs even that is frequently off. When ...


6

How can I measure how this latency is divided between outbound and inbound leg? You can find the congested ping direction with hping on linux / cygwin. If it turns out to be generated equally or mostly on inbound traffic, can I affect that (i.e. ingress throttling) or do I have to contact my ISP? You can do it either way, but the ISP method is better ...


6

Does increasing the bandwidth on a link from lets say 1mb to 30mb reduce the RTT? No, increasing the bandwidth does not reduce RTT strictly speaking. I say "strictly speaking" because it depends on what your are measuring! Scenario 1: The physical layer Given the following simple topology which is easy to follow, a copper Ethernet connection running at ...


6

On juniper hardware you can use the RPM service to get those measurements. The service can be configured to monitor specific interfaces which will help with the 'per-link' requirement.


5

There are a number of possibilities here, not limited to, but including: The CPE's CPU is maxing out. (CPE=Customer Premesis Device) Check the specifications for the router you are using to make sure it can support the level of traffic you're trying to push. Try and graph the CPE's CPU if you are unsure. Bufferbloat in your ISP If your ISP has configured ...


5

First, the linked article contains several poor approximation and is not IMHO a good source. Back to the question, insufficient bandwidth leads to link congestion, that means that the equipment buffers will be full and so some packets will be delayed, waiting for their turn to be sent from those buffers, thus increasing latency.


5

Round-trip time (RTT) is the time it takes for a packet to go from the sending endpoint to the receiving endpoint and back. There are many factors that affect RTT, including propagation delay, processing delay, queuing delay, and encoding delay. These factors are generally constant for a given pair of communicating endpoints. In addition, network congestion ...


4

Etherate enables you to measure throughput, latency and packet (frame) loss down at layer 2 directly over Ethernet (which sounds like what you are looking for). It doesn’t measure jitter at present but it will do in the future. Using it you can generate layer 2 Ethernet traffic in a controlled manner which you can measure. You can specify traffic flow ...


4

Check out Y.1731 protocol (or IEEE 802.1ag, or OAM). It's fairly easy to implement in software (I'm guessing that's what you're aiming at), and it's supported by all the physical network elements. There is a very naive open-source Y.1731 implementation in C, which you can look at. Please note that Y.1731 is not designed to cross physical network boundary (...


4

The thing most directly affecting RTT is signalling speed. Look at the progression of ethernet over the eons: 10M, 100M, 1G, 10G, 40G, and 100G. Each following version (except 40G) is 10x faster than the previous; the time to transmit a single bit is 1/10th as long. The time to transmit a full (1500B) frame drops by a factor of 10. So, the answer to your ...


4

Well, every link transmits only either 0 or 1 at any specific moment. So if your router connected to 2Mbit/s link receives first 250 bytes packet, and assuming that it doesn't have anything at the moment to send, it will pass the 250 bytes into the interface driver, and then serialize it (meaning put 0s and 1s) onto a link. The next packet, even if it's ...


4

Technically yes there will be a small difference and notability will depend on your devices/configuration. This is because of the different paths the packets have to take, but like i said, depends on how your setup is designed - there are plenty of variables. 1) If you are on the LAN then your path to the webserver via its private IP is merely just switched....


4

I'm not sure what you mean by buffer. There no buffering like when you want to watch a video. Besides the inherent latency of the bits traveling the various media in a path, there are delays due to serialization/deserialization to get bits on/off a link, frames must be stripped so that routers can inspect the packet, delays for routing lookups, packets must ...


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