- From: Djalma Araújo via GitHub <sysbot+gh@w3.org>
- Date: Thu, 29 Jun 2017 14:13:53 +0000
- To: public-webrtc-logs@w3.org
@vr000m Thanks for your answer. In Chrome 58+ with the new Spec implementation, I can only find jitter in the inbound, unfortunately. Thanks for your attention on this. These are the options available in chrome 58+: ```text "RTCCodec_InboundAudio_0" "RTCCodec_InboundAudio_103" "RTCCodec_InboundAudio_104" "RTCCodec_InboundAudio_105" "RTCCodec_InboundAudio_106" "RTCCodec_InboundAudio_110" "RTCCodec_InboundAudio_111" "RTCCodec_InboundAudio_112" "RTCCodec_InboundAudio_113" "RTCCodec_InboundAudio_126" "RTCCodec_InboundAudio_13" "RTCCodec_InboundAudio_8" "RTCCodec_InboundAudio_9" "RTCCodec_OutboundAudio_110" "RTCCodec_OutboundAudio_111" "RTCIceCandidatePair_PPgO5Gu9_4lcBLYP7" "RTCIceCandidate_4lcBLYP7" "RTCIceCandidate_PPgO5Gu9" "RTCInboundRTPAudioStream_2241787686" "RTCMediaStreamTrack_local_audio_e5f39725- "RTCMediaStreamTrack_remote_audio_a0_2241787686" "RTCMediaStream_local_ePzpEYoCX3TjptkSjzUUH51zzByZFz4ieQt4" "RTCMediaStream_remote_N8yj0WGDixhoP1cMPMBS8IDWuJtEO1zs" "RTCOutboundRTPAudioStream_262070451" "RTCPeerConnection" "RTCTransport_audio_1" ``` -- GitHub Notification of comment by djalmaaraujo Please view or discuss this issue at https://github.com/w3c/webrtc-stats/issues/228#issuecomment-311979448 using your GitHub account
Received on Thursday, 29 June 2017 14:13:59 UTC