Today’s distributed teams depend on consistently high-quality VoIP, video conferencing, and application performance. Yet, network bottlenecks and latency issues frequently disrupt productivity, frustrating users and draining IT resources.
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Instantly visualize your network’s true performance across every location, from local offices to global data centers. MyConnection Server (MCS) delivers intuitive visualizations of crucial performance metrics, including latency, jitter, packet loss, and bandwidth availability, pinpointing exactly where network issues occur and enabling proactive action before productivity suffers.
With MCS’s advanced alerting and diagnostic capabilities, you'll swiftly resolve network bottlenecks and ISP problems, ensuring seamless VoIP and video conferencing experiences. By continuously simulating real-world conditions, MCS helps guarantee exceptional digital interactions, every time.
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Jitter time will increase proportionate to the number of packets lost and audio content will be missing. The distribution of the packets lost is a critical measure.
Most VoIP tests report packet loss as an average, for example dropping 25 packets out of a thousand would be expressed as 0.25%. Sounds like a very small number and not worth worrying about.
However, consider the problem if the 25 packets were dropped contiguously. At 50 packets per second that represents 50% of the time window.
Jitter is measure in milliseconds and a good result is 0 (zero) milliseconds and a bad result is any number other than zero. The higher the number the bigger the threat to audio quality.
As a practical human example consider two cars that leave 5 minutes apart heading to the airport. If the cars arrive 5 minutes apart then the jitter value is zero i.e. no change in time. If the trailing cars arrive 6 minutes apart the reported jitter is 1 minute.
If the jitter time is large then the second car arrives too late and misses the flight. In other words, to deliver a good experience the car must arrive in time to catch the flight.
For example on 1Gbps workstation the smallest data payload sent from a browser is about 48 packets. An connection at 12Mbps will consume 1 millisecond of time for each packet. Why is this and why does it matter?
12 million bits per second is 12,000 bits per 1ms. A data packet is normally 1500 bytes which is 12,000 bits. Therefore 48 data packets will consume 48 milliseconds of time on the network. If your VoIP packets are behind the data stream then on a 12Mbps connect your jitter peaks will be at least 48ms or higher.
Remember 48 packets is the smallest payload of data, plus there are many other workstations that send data on the network. If jitter is consistently high then the jitter buffer will drain and cause the application audio quality to diminish. Most Jitter buffers are around 80 – 100ms and increasing the jitter buffer is not a practical solution because it increases the talk-over lag time which also affects call quality.