how to allocate bandwidth for users and prioritize for voip/video calls?

learning_network

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Mar 29, 2011
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Even though we have 2 mbps connection for 25 users, we always have an issue of staff complaining they could not hear properly during video/voip calls for business purpose.

Our Network graph shows however we are utilizing only 1.5 to 1.6 mbps on an average and the remaining are wasted.

I initially thought it could be our ISP who are not providing full bandwidth to us. However I confirmed we are getting 2 mbps by streaming many videos on the computer.

So basically it seems even though we are not utilizing full bandwidth, during video/voips calls they bandwidth is not allotted properly by our TP-Link router to utilize it completely.

Right now we allotted 1 mbps for 17 users and another 1 mbps for another 8 users(important users).

We don't use the connection for browsing much. It is used mainly for company email and Skype/other video calls.

Can you guide me to utilize our bandwidth fully? And if need any suggestions for new hardware/software to control it.
 
Solution
If this is a dedicated network, ie not internet, you should be able to work with the provider to give you the quality of service you need. What you want to do is allocate a certain amount for voice traffic. This traffic you want to use what is called low latency queuing on. You must be careful to correctly size this because the packets are always sent before any other packets. Then for your video traffic you want a guaranteed bandwidth, this is slightly different than the voice traffic. It is allowed to be delay a tiny bit as long as the average throughput meet the requirement. The key here will marking your traffic so the ISP can apply the proper traffic rules to it. Most end equipment can marks its own traffic.

The reason you...
If your router supports QoS that's where you would set this up. You find the port number range of your priority traffic and give it a higher priority so that in the event of bandwidth shortages it knows which packets to send first as they pile up.

VOIP is actually pretty bandwidth lean but extremely latency sensitive. However, you mentioned video calls. While for purely audio you probably only want to provision for about 10KB/s (about 100kbps . . . it will actually use less, more like half, but you want the headroom to keep latency down). However, per user the video call you probably want to provision more like 500kbps - 1.5Mbps per user depending on your settings. The connection clearly isn't good enough for more than a single user or possibly 2 video calls on low quality/high compression settings.

This all also assumes the connection is strictly used for ONLY VOIP/Communication. If any other traffic is going on the line then this connection is not adequate. I believe you can set the ports for voice data in the skype settings so that you can prioritize that traffic, but the best bet would be to get a connection that's more suited to the traffic, or a second connection strictly for communication.
 
If this is a dedicated network, ie not internet, you should be able to work with the provider to give you the quality of service you need. What you want to do is allocate a certain amount for voice traffic. This traffic you want to use what is called low latency queuing on. You must be careful to correctly size this because the packets are always sent before any other packets. Then for your video traffic you want a guaranteed bandwidth, this is slightly different than the voice traffic. It is allowed to be delay a tiny bit as long as the average throughput meet the requirement. The key here will marking your traffic so the ISP can apply the proper traffic rules to it. Most end equipment can marks its own traffic.

The reason you are seeing issues even though you are not at 100% utilization is because the numbers you look at are averaged over too long a period of time. If you would get a graph that showed 1 minute average utilization you would see lots of small spikes to 100%

Now if this is internent connection or the ISP will not work with you the only way to avoid this is to buy more bandwidth to try to cut the number of spikes in the data stream. Even then you will always have some and voice is extremely picky...or the users are since it does not actually get bad enough to drop the call just some of the speech
 
Solution