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Transcription:

Network Management & Monitoring Network Delay These materials are licensed under the Creative Commons Attribution-Noncommercial 3.0 Unported license (http://creativecommons.org/licenses/by-nc/3.0/)

End-to-end Delay The time required to transmit a packet along its entire path Created by an application, handed over to the OS, passed to a network card (NIC), encoded, transmitted over a physical medium (copper, fibre, air), received by an intermediate device (switch, router), analyzed, retransmitted over another medium, etc. The most common measurement uses ping for total round-trip-time (RTT).

Historical Measurement of RTT What is this telling us? We need to understand the sources of delay

Causes of Delay Processing delays Queuing delays Transmission delays Propagation delays

1. Processing Delay Time required by intermediate routers to decide where to forward the packet, update TTL, perform header checksum calculations (Note: most modern routers handle packet forwarding in hardware at full line rate) plus: Time for the far end to process the ICMP echo request and generate a response

2. Queuing Delay Queuing Delay The time a packet is enqueued while the link is busy sending other packets This is a statistical function and depends on the arrival times of other packets QoS configurations may prioritize some types of traffic over others (In practice, that means multiple queues, and different packets are assigned to different queues)

3. Transmission Delay Transmission Delay The time required to push all the bits in a packet on the transmission medium in use For N=Number of bits in packet, R=transmission rate (bits per second) t = N/R For example, to transmit 1500 bytes (12000 bits) using Fast Ethernet (100Mbps): t = 12000/1x10 8 = 0.12 milliseconds

4. Propagation Delay Once a bit is 'pushed' on to the transmission medium, the time required for the bit to propagate to the other end of its physical path For a given medium, the velocity of propagation is usually constant (some fraction of the speed of light) The longer the path, the longer the delay For x = distance, v = propagation velocity t = x/v

Transmission vs. Propagation Can be confusing at first Consider this example: Two 100 Mbps circuits - 1 km of optic fiber - Via satellite with a distance of 35,000 km between the base and the satellite For two packets of the same size which will have the larger transmission delay? Propagation delay?

Jitter

Questions about Jitter Queuing Delay We've seen four causes of delay. Which are constant for a given path and packet size, and which are variable? What applications are particularly sensitive to jitter? Those applications may apply extra buffering to smooth out jitter why is that additional delay a problem?

Questions??

Packet Loss Causes of packet loss: Transmission errors Queue overflow (congestion)

1. Transmission errors "1" received as "0", or vice versa - e.g. due to excess noise, poor connections,... Can be measured in terms of "bit error rate" (BER) If one or more bits in a packet is corrupted, the whole packet is discarded Retransmission of lost packets is the responsibility of higher layers (transport or application)

2. Queue overflow Queues do not have infinite size If a packet arrives when queue already full, it is dropped Ultimately caused by insufficient capacity However, packet loss starts to occur before the link is 100% utilized, because of random distribution of arrival times Retransmissions cause further demand and could lead to network collapse!

TCP and Congestion Control Flow Control and Congestion TCP limits sending rate by means of a "congestion window" The congestion window starts small, and increases gradually while there is no packet loss Any detected packet loss causes the congestion window to shrink rapidly, so the sender sends more slowly

Different TCP Congestion Control Algorithms

Effects of TCP congestion control Flow Control and Congestion Network collapse is prevented "Fair sharing" When there are multiple TCP streams, each one uses an approximately equal share of available bandwidth TCP detects congestion by observing packet loss Newer TCP stacks also respond to "Explicit Congestion Notification" signals from routers: packets are marked when queues nearly full

TCP and transmission errors Flow Control and Congestion TCP cannot tell the difference between transmission errors and queue overflows! Hence transmission errors cause TCP to slow down too Formula for maximum throughput of TCP in the presence of packet loss: MSS RTT. p loss

Example calculation: LAN Flow Control and Congestion MSS = 1460 bytes RTT = 1ms = 0.001 seconds Packet loss = 2% = 0.02 1460 / (0.001 * 0.02) 10.3MB/sec = 82 Mbps Short RTT means packet loss does not have a huge impact on local transfers

Example calculation: WAN Flow Control and Congestion MSS = 1460 bytes RTT = 150ms = 0.15 seconds Packet loss = 0.02% = 0.0002 1460 / (0.15 * 0.0002) 690KB/sec = 5.5 Mbps Loss of just 1 packet in 5,000 causes severe reduction of throughput when transferring across the Internet!

Measurement of packet loss Flow Control and Congestion Smokeping gives a coarse measurement (20 packets every 5 minutes => 5% loss detectable, but bursts may be missed) For more accurate measurement you need a tool like perfsonar / owamp Standard configuration sends 10 packets per second continuously Can detect packet loss of 0.17% over one minute, or 0.0028% over one hour Separate measurements in each direction

Questions??