Intent to Experiment: Configurable WebRTC jitter buffer max size

[jonas...@chromium.org]

Contact emails

kud...@chromium.org, jonas...@chromium.org

Spec

https://docs.google.com/document/d/1stYIZhEmDZ7NJF9gjjsM66eLFJUdc-14a3QutrFbIwI/edit?usp=sharing

Summary

There's a field in the native WebRTC configuration that controls the size of the audio jitter buffer. We intend to create an origin trial that enables users to set this value, and read a stat counting how many times the buffer gets flushed.

Goals for experimentation

We want to investigate how good the current default value is. If changing it turns out to be useful, we might either tweak the default or try to standardize some way to control it.

We'll base these judgments on the new buffer flush stat as well as some preexisting performance metrics.

Experimental timeline

We plan to add the experiment to M72, and run it until March 2019.

Any risks when the experiment finishes?

Not really. When it finishes we'll automatically revert to using the current default value for the buffer size.

Ongoing technical constraints

None

Will this feature be supported on all five Blink platforms supported by Origin Trials (Windows, Mac, Linux, Chrome OS, and Android)?

Yes.

[Philip Jägenstedt]

LGTM!

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[Nitin Gode]

Hello,

Would there be any plans to integrate this experiment?

[Jonas Olsson]

Hello,
there's no plans to launch this experiment in its current state. kuddai@, do you know if there's any future plans for a jitter buffer API?