Basic network requirements for remote working (Teleworker)

Due to an increase in the amount of users working from home (or otherwise remotely), users are experiencing more network issues that lead to degraded or interrupted service. This article is meant to provide basic network requirements for remote working along with an explanation of the different factors that can impact user experience.
Environment
Home network
Procedure

Conditions that can result in voice quality issues include:

  • High latency
  • Packet Loss
  • High jitter
  • Lack of Bandwidth
 Latency and packet loss:

Latency in a call is the amount of time it takes for the caller’s voice to reach the other end of the connection. As latency increases in a conversation, it becomes increasingly difficult for users to sustain a normal two-way conversation. The conversation rapidly deteriorates from an interactive exchange to an "over-to-you" radio-style form of communication. Severe latency in the network can result in dropped calls.

Latency becomes noticeable in a call at 80 ms to 200 ms delay and is radio-style at 400 ms delay. Assuming that jitter and packet loss (explained later on) are not an issue, end-to-end delay is the most likely cause of voice quality issues on VoIP calls. Ideally, this should not exceed 80 ms.

Packet loss occurs when a network packet fails to reach its destination. If this happens a lot, you will notice a decrease in voice quality. Depending on the amount of packet loss, you might find words or parts of sentences cut out, making it hard to hold a conversation. Severe cases can cause a call to drop entirely.
 

Jitter:

Jitter is the variation in the packet delay. The major cause of jitter is network congestion. This occurs when the amount of data arriving at any node along the network path (including the source, destination, routers, and switches) exceeds the capacity of that node to forward the data. In this case, data is stored in a buffer queue until the node is able to forward the data. The time that the data spends in this buffer before being forwarded is the major source of jitter.

The following table shows how latency, jitter and packet loss define good, borderline, or unacceptable call quality:
 

RATING

END-TO-END DELAY

PING DELAY

PACKET LOSS

JITTER

Good

<50 ms

<100 ms

<0.5%

<20 ms

Borderline

<80 ms

<160 ms

<2%

<60 ms

Unacceptable

>80ms

>160 ms

>2%

>60 ms

 

Bandwidth:
Bandwidth is often referred to as internet speed and is a measure of how quickly you can send/receive data over the internet. If there is insufficient bandwidth, symptoms experienced can include degraded voice quality, slow response, service interruption, or in extreme cases complete loss of service.

The following table shows bandwidth requirements for common activities:

Voice

If compression (G.729a) enabled: 24 Kbps (bi-directional)
If compression not enabled (): 80 Kbps (bi-directional)

MiCollab Audio, Web and

Video Conferencing

192 Kbps (bi-directional)

MiCollab Client Video

256 Kbps – 1600 Kbps (bi-directional)

MiVoice Video Unit

512 Kbps – 1500 Kbps (bi-directional)

MiTeam Meetings Audio

54 Kbps (bi-directional)

MiTeam Meetings Video 1600 Kbps upload
1600 + (( n-2) * 400) Kbps download (n is number of participants)
MiTeam Meetings Screen Share 1200 kbps upload or download (upload if you're sharing, download if someone else is)

Note that this is in Kbps, Kilobits per second. Traffic listed as bi-directional needs the listed amount in terms of both upload and download bandwidth.

Testing your connection:
A very quick and easy (but not very thorough) test to run is a speedtest. This test will connect to the nearest speedtest server and test your jitter, ping, and upload & download bandwidth. It does however not show packet loss.
Please note that although your results might be good at some point, network conditions can change and intermittent issues are common, so it's a good idea to run this test while you're experiencing issues as well.

A more appropriate (and still easy) test can be performed using the Teleworker Network Analyser. This program comes with the MBG, and will be attached to the bottom of this article for convenience. To use it, download the installer and install. Once it's installed you can open it up, enter the MBG's FQDN or Public IP address (1), go to the 'Load Test' tab (2), press 'Start Load Test' (3):



Optionally, you can log the results to a file (4), e.g. for analysis by your administrator.

If you notice your results are unexpected (e.g. the results look good, but you're experiencing issues), a wider variety of tests may need to be performed. If you're unsure how to go about this, contact your support.

Additional Notes

SIP ALG

SIP Application Level Gateway is a common component of commercial routers. It is supposed to help improve SIP connections, but due to it being implemented slightly differently everywhere (and sometimes poorly), it can cause interoperability and other issues. It's therefore recommended to disable this feature on your home router. Most home routers will have a guide available online, so you can search for something along the lines of "disable SIP ALG <your router model>". Alternatively, ask your Internet Service Provider for assistance.

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Voor troubleshooting kun je de TNA tool hier downloaden : https://dl.businesscom.nl/?dir=Mitel\Micollab 

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