Common Network Performance Issues
Bursty data transfer is when the sequence in which the packets are delivered and the variable intervals between packets arriving do not materially affect the application.
- used in file transfer
Quality-of-Service (QoS) protocols and appliances are designed to support real-time services.
- voice and video applications carry real-time data
- have high bandwidth requirements
- typically less of a problem than packet loss, latency, and jitter
- have high bandwidth requirements
Bandwidth
Bandwidth is the amount of information that can be transmitted.
- measured in bits per second (bps), or some multiple thereof
- expresses the available capacity of the link
- when monitoring, need to:
- distinguish between the nominal data link/Ethernet bit rate
- throughput of a link at layer 3
- the goodput available to an application
- bandwidth for audio depends on:
- sampling frequency (Hertz)
- bit depth of each sample
- e.g., early digital telecommunications links were based on 64 Kbps channels
- derived from calculating:
- voice frequency range is 4,000 Hz
- must be sampled at twice the rate to ensure an accurate representation of the original analog waveform
- sample size is 1 byte (8 bits)
- 8 KHz x 8 bits = 64 Kbps
- voice frequency range is 4,000 Hz
- derived from calculating:
- VoIP bandwidth requirements for voice calling can vary
- allowing 100 Kbps per call upstream and downstream is sufficient in most cases
- Video bandwidth is determined by:
- image resolution (number of pixels)
- color depth
- frame rate (fps)
Bottlenecks
A bottleneck is a point of poor performance that reduces the productivity of the whole network.
- may occur because:
- a device is underpowered or faulty
- user or application behavior
- identifying cause of bottleneck:
- identify where and when the network overutilization or excessive errors occur
- if problem is continual,
- likely to be device related
- if problem occurs at certain times,
- likely to be user or application related
Packet Loss
- packet loss is expected
- but only to a degree
- larger network = more packet loss during heavy traffic
- identifying packet loss:
- run a packet sniffer on affected segment
- high numbers of TCP retransmission and duplicate acknowledgements are strong indicators
- know where and when packet loss occurs to find device that is dropping frames
- reasons for dropped packets:
- server, router, or switch is overloaded
- power outage
- firewall is blocking packets from a known destination
- malicious actor is interfering with network transmissions
- faulty firmware is causing packet processing errors
Latency and Jitter
Latency is the time it takes for a transmission to reach the recipient.
- measured in milliseconds (ms)
- can test latency of a link with
pingpathpingmtr
Jitter is variation in the time it takes for a signal to reach the recipient.
- manifests as an inconsistent rate of packet delivery
- measured in milliseconds (ms)
- using an algorithm to calculate the value from a sample of transmit times
- can test jitter with
mtr
Latency and jitter deal with problems of timing and sequence of packet delivery.
- not significant problems when data transfer is bursty
- real-time applications are more sensitive
- manifests as echo, delay and video slow down
- if packets are delayed, arrive out of sequence, or are lost
- receiving host must buffer received packets until the delayed packets are received
- if packet loss is so excessive that buffer is exhausted,
- then noticeable audio or video problems (artifacts) are experienced
- when assessing latency, consider Round Trip Time (RTT)
- VoIP requires
- RTT < 300 ms
- jitter < 30 ms
- packet loss < 1%
- VoIP requires
- with Wireshark
- gives each packet a timestamp relevant to when the last frame was sent from the very first transmitted frame
- can find delays in TCP conversation during a session
- can plot a sequence graph to visually represent how this delay behaves
- line should have a gradual, steady increase upward to the right
- optimal network should have small gaps between each transmission
- the more longer and jagged the graph, the more latency is introduced
- gives each packet a timestamp relevant to when the last frame was sent from the very first transmitted frame