Can Your Network Successfully Deploy Video Conferencing Sessions?
by June 1, 2016

Filed under: Networking Technology, Performance Monitoring

Video conferencing is increasingly a part of daily office life. Ad-hoc meeting software like Google Hangouts, HipChat, Slack and Skype have popularized virtual face-to-face meetings. But formal meetings with entry codes like GoToMeeting and WebEx are often relied on for marketing webinars or sales demos. Regardless of purpose, video conferencing has evolved over the years from multi-screen hardware deployments to built-in laptop webcams.

But it only works if the network environment and available bandwidth can support the demands of video conferencing. For large deployments, the time and money devoted to equipment, conference room modifications (in the case of audio or camera mounting), training for users, etc. will not yield expected business returns if the network cannot handle the demands placed on it by video traffic. For smaller deployments, productivity is lost when conferencing fails.

All video conferencing traffic is real-time traffic and demands adequate QoS support across both LANs and WANs. And we have all learned the hard way that video applications are unforgiving in terms of network quality. Each link in the network must have adequate “clean bandwidth” (that is, bandwidth plus high performance on the upload and the download) to handle not only the expected voice and video traffic, but also the myriad other IP-based applications running on it. That’s everything from SaaS applications to IP storage to virtual desktop infrastructure.

Video Conferencing Experience By the Numbers

In terms of performance metrics, even the most forgiving video conferencing applications can handle only minimal latency, jitter and packet loss:

  • Excessive latency threatens to de-synchronize the audio and video portions of the conference. When round-trip latency between endpoints exceeds 200 to 300 milliseconds, participants start to notice delays between the movement of speakers’ lips and the corresponding audio.
  • Packet loss greater than 0.1% to 2% causes the video presentation to be “blocky” or jerky, and causes the audio to drop out.
  • Jitter exceeding 15 to 30 milliseconds can give the video a frozen or stuttering appearance.

Maintaining end-to-end network performance for all video conference participants within these narrow parameters is vital to successful remote interactions, and ultimately essential to realizing the expected ROI for video conferencing deployments. The shift from traditional, room-based videoconferencing systems to desktop and mobile solutions has greatly increased not only the number of endpoints but also the magnitude of the network management challenge.

Another growing challenge with providing network QoS for video conferencing involves ensuring acceptable experience when interfacing with partners, customers and others outside the enterprise network. Whenever video conference data travels across network boundaries, service quality can fluctuate.

To assure successful video conferences, you need to know how much bandwidth you have available and how the network is performing—not just on your LAN but end-to-end across all the networks involved in supporting the video conference. Simply tracking total, available and utilized capacity over networks you own isn’t sufficient, because it leaves you simply hoping for success across a web of service providers and best-effort public networks. You need visibility across these networks as well, both to pinpoint where problems are occurring and to know if service providers are meeting their SLAs.

AppNeta Performance Manager offers the end-to-end visibility required to ensure network performance quality for video conferencing—and maximize the value of these investments.