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Re: [webrtc-stats] Why no Jitter for RTCOutboundRTPAudioStream?

From: Djalma Araújo via GitHub <sysbot+gh@w3.org>
Date: Thu, 29 Jun 2017 14:13:53 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issue_comment.created-311979448-1498745631-sysbot+gh@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"
```

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Received on Thursday, 29 June 2017 14:13:59 UTC

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