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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