MCS MyVoIP simulates voice traffic over a UDP connection and measures the key factors affecting VoIP call quality – jitter, packet loss, and loss distribution, and reports a MOS rating for the connection.
The MyVoIP test includes an option to test SIP response to help determine if a remote client can establish a VoIP connection. Response times are reported for Register, Invite and Bye.
MyVoIP supports measuring and reporting of the VoIP codec protocols (such as the G7xx codec) deployed by most VoIP service providers, non-standard codec protocols are also supported.
View Connection Test Method Comparison
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Most streaming video is transmitted using TCP and the Real Time Streaming protocol, which means jitter is the key performance metric, as opposed to packet loss. Any lost packets will be recovered by the TCP stack, resulting in little to no packet loss, but in turn causing significant jitter spikes which degrade video picture and sound quality.
MyVideo simulates video traffic over a TCP socket connection using the Real Time Streaming protocol. The test process measures and identifies TCP delays that cause jitter and clearly shows the impact to the data flow and resulting jitter over time for both audio and video streams. The response times for Setup, Describe and Play requests are reported, along with Trip Time. The test payload can be customized by packet size and rate for audio and video.
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