From various examples, sites, forums, discussions, and source codes, I've compiled this list of experimental google RTCPeerConnection constraints. What do each of them do in the context of a WebRTC RTCPeerConnection?
peerConnectionConstraints:
    optional: [
        #goog experimental
        { googIPv6: true }
        { googImprovedWifiBwe: true }
        { googDscp: true }
        { googSuspendBelowMinBitrate: true }
        { googScreencastMinBitrate:400 }
        { googCombinedAudioVideoBwe: true }
        { googCpuOveruseDetection: true }
        { googCpuOveruseEncodeUsage: true }
        { googCpuUnderuseThreshold: 55 }
        { googCpuOverUseThreshold: 85 }
    ]
Most of these options are non-documented. Some of them used for Google's products such as Hangouts.
googIPv6 - enables support of IPv6 for Chrome (should also work in latest Firefox)
googImprovedWifiBwe - Chrome has bandwidth estimation algorithm, and this flag enables improved (but experimental) version of the algorithm. In the latest version of Chrome this flag is on by default, - the experiment was successful.
googDscp - enables DSCP
googScreencastMinBitrate - Used by Hangouts application. Sets the minimum bitrate for screensharing.
googCpuOveruseDetection - if enabled, Chrome will lower outgoing video quality and video resolution
googCpuUnderuseThreshold - the min CPU load (percents), used in pair with googCpuOveruseDetection
googCpuOverUseThreshold - the max CPU (percents), used in pair with googCpuOveruseDetection
googCpuOveruseEncodeUsage - don't know
googSuspendBelowMinBitrate - the minimal bitrate for considering session is alive (not sure)
googCombinedAudioVideoBwe - don't know
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