Monitoring adaptive bitrate (ABR)

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davidsw...@gmail.com

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Mar 25, 2019, 6:55:46 PM3/25/19
to discuss-webrtc
If one wants to monitor connection quality from the client side, I think the major factor other than disconnections is "adaptive bitrate (ABR)". I'm pretty sure that the getStats browser api is one of the best ways to do this, but the word "adaptive" doesn't appear in https://www.w3.org/TR/webrtc-stats/

I can see in Chrome72/Win10 that "candidate-pair" events have property "availableOutgoingBitrate" as well as "bytesReceived" and "bytesSent", and that event "outbound-rtp" has properties "bytesSent" and "packetsSent", and that event "transport" has "bytesSent" and "bytesReceived". I'd guess that "candidate-pair" events appear only during call setup.

Has a best practice emerged about how to use these events and properties over the duration of a call (assuming we probe at an interval of 150msec to 1sec) to monitor adaptive bitrate?

David P

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Mar 27, 2019, 2:04:09 PM3/27/19
to discuss-webrtc
The best option I've found so far is to use callstats.io but they may not support your clientside js library (they don't in our case).

The second-best is to use the rtcstats client, which only works with the rtcstats server, whose installation is mostly undocumented and is said to be pretty difficult:  https://github.com/lifeonairteam/rtcstats-server/issues/73

So it seems rolling one's own is actually the only option in many cases.

Philipp Hancke

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Mar 28, 2019, 12:18:14 PM3/28/19
to discuss...@googlegroups.com
if you merely want to calculate the bitrate something along the lines of
should be sufficient. Typically you'll want bytesSent/bytesReceived on the outbound-rtp/inbound-rtp.
The bytesSent/bytesReceived on the candidate-pair includes a number of non-rtp things such as stun packets.


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