Friday, December 27, 2019

How do Cloud VoIP Providers guarantee qos and call quality if you access their services over the Internet?

As we all know as network professionals there is no qos honored on the Internet between different carriers. Dscp is usually stripped off or at least ignored. We also know as network professionals that VoIP cannot work without qos.

If you send 10 udp packets from your location to another location on the Internet chances are all 10 packets will each take a completely different path, hitting different routers and even different autonomous system numbers. This is just how the Internet is designed, and if at any hop your packet meets a loaded interface your packet will be buffered and transmitted best efforts after any carrier grade traffic is given priority.

This means two big things.

  1. The time between the packet being sent will not match the time between the packet arriving. This is important because RTP sends a steady stream of packets each packet sent at exact time intervals.

  2. The packets may not arrive at the same order they were sent. This is important because each packet has a small sample of audio data

My question is how do Cloud VoIP providers guarantee good call quality and qos on their product if you are using a best effort medium to reach them?

If you have got a tier 2 isp for example your VoIP might go through 3-4 differ transit provider before it reaches your provider.

I am just wondering how businesses are able to use Cloud VoIP and the users do not notice any problem? How is that working so good? Many businesses are using this Cloud VoIP so I’m wondering if there is something going on where they found a way to protect this traffic and give it qos?



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