EKK:LoFi Sounds in HiFi Spaces/Making connection/Skype: Difference between revisions

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[[File:Skype Tech info.png]]
[[File:Skype Tech info.png]]
Here are some explanations for the terms used in the display.
Packet Loss: Packet Loss measures the reliability of a connection. A set of known data is sent to the server and then the server is supposed to deliver the same data back without any changes. If 100 packets were sent out, but only 80 made it back, then it would be 20% packet loss. In order to maintain a good Skype communication session, the packet loss should not be higher than 5%.
Roundtrip: Roundtrip is the latency of the network connection. Roundtrip times between 150ms to 200ms represent an excellent condition in which none of the call participants need to pause and wait for other people to talk. If the number reaches the 350ms, the need to pause becomes significant.
Relays: When a Skype call is being relay transferred, it means the direct connection between the caller and the receiver could not be established. The data going back and forth between two parties would be routed through several nodes. The idea situation is to have zero relay to achieve direct connection. The lower the relay number, the better the call quality will be.
Jitter: Jitter stands for the variations between consecutive data packets arriving at the user's side. The lower this number, the better the voice quality will be. Skype has implemented a jitter buffer to ensure the uninterrupted communication.
CPU usage: CPU usage over 80% will degrade the call voice quality.

Revision as of 16:42, 22 April 2013

I found an option on Skype that gives the connection quality data. To get this information the user needs to go to skype preferences > advanced > click on the "Display technical call info during calls". Once it is done the user makes a skype call and while it is active the user presses cmd 5 or goes to window>Display technical call info. A window like this will appear.

File:Skype Tech info.png

Here are some explanations for the terms used in the display.

Packet Loss: Packet Loss measures the reliability of a connection. A set of known data is sent to the server and then the server is supposed to deliver the same data back without any changes. If 100 packets were sent out, but only 80 made it back, then it would be 20% packet loss. In order to maintain a good Skype communication session, the packet loss should not be higher than 5%.

Roundtrip: Roundtrip is the latency of the network connection. Roundtrip times between 150ms to 200ms represent an excellent condition in which none of the call participants need to pause and wait for other people to talk. If the number reaches the 350ms, the need to pause becomes significant.

Relays: When a Skype call is being relay transferred, it means the direct connection between the caller and the receiver could not be established. The data going back and forth between two parties would be routed through several nodes. The idea situation is to have zero relay to achieve direct connection. The lower the relay number, the better the call quality will be.

Jitter: Jitter stands for the variations between consecutive data packets arriving at the user's side. The lower this number, the better the voice quality will be. Skype has implemented a jitter buffer to ensure the uninterrupted communication.

CPU usage: CPU usage over 80% will degrade the call voice quality.