VoIP Network Quality Tester: Difference between revisions
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Contact: John Palombo, [[Metaswitch Networks]] <John.Palombo@metaswitch.com> | |||
VoIP and Video calls are very sensitive to latency, jitter and packet loss in IP networks, but there aren’t many good end-user tools available to determine how good a network is. This project creates a probe that users can run on a standard end user device (e.g. desktop, tablet or smart phone) that communicates with a remote server and provides an indication of how good the network is and predicts the quality of calls through it. Ease of use is essential – both to install/run and presenting the results in a user friendly manner. Idea could be extended for example to crowd-sourcing across many users in order to build a much wider picture of the network, or to predict quality of a gaming connection. | VoIP and Video calls are very sensitive to latency, jitter and packet loss in IP networks, but there aren’t many good end-user tools available to determine how good a network is. This project creates a probe that users can run on a standard end user device (e.g. desktop, tablet or smart phone) that communicates with a remote server and provides an indication of how good the network is and predicts the quality of calls through it. Ease of use is essential – both to install/run and presenting the results in a user friendly manner. Idea could be extended for example to crowd-sourcing across many users in order to build a much wider picture of the network, or to predict quality of a gaming connection. | ||
Feedback: Very timely. However, students may not have suitable signal processing skills. Would this use separate audio hardware to inject and capture controlled signals, or rely on intercept of system audio? Would capture and analysis of video quality be an alternative? | |||
Response: The project can actually be much simpler than you are thinking and wouldn't need to sniff the network. You'd have two end point boxes, each sending a stream of packets to the other at a fixed rate (like a VoIP call, but you can think of it like a fast ping). The remote end would then gather stats about the packet arrival times, specifically packet delay (latency), the variation in packet delay (jitter) and lost packets. That data alone would be very useful to characterise how good a connection you have. | |||
You are right that the next step to interpret how that affects a call from knowledge of the codecs might be a bit tougher, but they could make an empirical judgement by actually making a VoIP call over the same network connection and judging the voice quality and correlating that with the packet stats. In order to have a suitably poor network connection to get some good data, they might need to get a bit creative by connecting over a poor wifi or mobile network, but that should be quite doable. | |||
Latest revision as of 17:57, 29 September 2014
Contact: John Palombo, Metaswitch Networks <John.Palombo@metaswitch.com>
VoIP and Video calls are very sensitive to latency, jitter and packet loss in IP networks, but there aren’t many good end-user tools available to determine how good a network is. This project creates a probe that users can run on a standard end user device (e.g. desktop, tablet or smart phone) that communicates with a remote server and provides an indication of how good the network is and predicts the quality of calls through it. Ease of use is essential – both to install/run and presenting the results in a user friendly manner. Idea could be extended for example to crowd-sourcing across many users in order to build a much wider picture of the network, or to predict quality of a gaming connection.
Feedback: Very timely. However, students may not have suitable signal processing skills. Would this use separate audio hardware to inject and capture controlled signals, or rely on intercept of system audio? Would capture and analysis of video quality be an alternative?
Response: The project can actually be much simpler than you are thinking and wouldn't need to sniff the network. You'd have two end point boxes, each sending a stream of packets to the other at a fixed rate (like a VoIP call, but you can think of it like a fast ping). The remote end would then gather stats about the packet arrival times, specifically packet delay (latency), the variation in packet delay (jitter) and lost packets. That data alone would be very useful to characterise how good a connection you have.
You are right that the next step to interpret how that affects a call from knowledge of the codecs might be a bit tougher, but they could make an empirical judgement by actually making a VoIP call over the same network connection and judging the voice quality and correlating that with the packet stats. In order to have a suitably poor network connection to get some good data, they might need to get a bit creative by connecting over a poor wifi or mobile network, but that should be quite doable.
