nanoStream Webcaster Client Release 5.16.0.

We have released the nanoStream Webcaster Client 5.16.0.

Release Notes

This release introduces additional quality indication statistics on the client’s side. These new statistics will allow you to show a “traffic light” to your webcasters which indicates the current quality of the ingest. This is in regards to the upstream network conditions (from your customers’ browsers to our webcast servers).

A bad network is the major reason for bad end-to-end user experience.  In case of bad network conditions you will now be able to warn your users and they will be able to take further steps in order to improve the situation.

For monitoring ingest quality, we added statistics on the client’s side for:

  • Packet Loss (percentage)
  • Round Trip Time (milliseconds)

The values will be emitted with the WebRTCStatsEvent as “data.stats.packetLoss” and “data.stats.roundTripTime”. Along with the stats for videoSendDelay, the additional values of packet loss and round trip time will give you a better picture of current ingest conditions. Please be aware that videoSendDelay is only visible on Chromium based browsers, like Chrome and Microsoft Edge browsers.

Please find an updated documentation on “Stats and Metrics” here.

Further Explanation of The Featured Statistics

Packet Loss

This will show the percentage of lost packages within the last 10 seconds of streaming, so your users can observe if the network is good or bad at the moment. For low latency streaming it is important to have a low amount of packet loss, high packet loss will impact/decrease stream bitrate and introduce additional latency.

Round Trip Time (RTT)

This shows the amount of milliseconds that data takes from the client to the webcaster server and back. It is an important indicator
for possible additional latency being introduced on the network between your customers and the nanocosmos infrastructure.
Reasons for higher round trip times are:

  • Streaming from wifi networks
  • Having additional software or hardware running that consumes bandwith

Recommendations For User Feedback

Based upon our analysis, we have gathered recommendations for indicating bad streaming conditions to the end users.

Round Trip Time: We concider an rtt of 150 or less as acceptable. We observed RTT values going up to 150, with still good playback experience.
Above 150 and below 250 the viewer experience will degrade slighly. Above 250 milliseconds users should check their network and
see if there are things that can be improved.

Packet Loss: A packet loss of less than 5% still results good playback experience. Greater than 10% of lost packages will degrade viewer experience,
especially for streams with higher bitrates (2Mb and above). Above 10% of packet loss, we observed that streams can get chunky and your users should take action then.

Recommendations For User Actions

If the indicators keep staying obove the upper threshold recommendations, users should:

  • High round trip time:
    – check their network with a speed test
  • High packet loss:
    – reconnect
    – reduce resolution
    – reduce bitrate

Additional Release Improvements

Additionally, this release includes a minor improvement of the javascript file size. The release file was reduzed in size, which allows slighly faster loading times of your webpages.

Please find an updated documentation on “Stats and Metrics” here.



  • Emit additional stats with the WebRTCStatsEvent: packetLoss and roundTripTime.


  • Reduced release js file size



Token Helper

Documentation about secure ingest

5.16.0 Release Packages Download

Complete Bundle Package with Demo Page







Please reach out to our team for further information or any questions: