files/en-us/web/api/rtcinboundrtpstreamstats/index.md
{{APIRef("WebRTC")}}
The RTCInboundRtpStreamStats dictionary of the WebRTC API is used to report statistics related to the receiving end of an RTP stream on the local end of the {{domxref("RTCPeerConnection")}}.
The statistics can be obtained by iterating the {{domxref("RTCStatsReport")}} returned by {{domxref("RTCPeerConnection.getStats()")}} or {{domxref("RTCRtpReceiver.getStats()")}} until you find a report with the type of inbound-rtp.
timestamp property, on the other hand, indicates the time at which the statistics object was generated.getStats().MediaStreamTrack associated with the inbound stream.These properties are computed locally, and are only available to the device receiving the media stream. Their primary purpose is to examine the error resiliency of the connection, as they provide information about lost packets, lost frames, and how heavily compressed the data is.
These statistics are measured at the receiving end of an RTP stream, regardless of whether it's local or remote.
The following properties are common to all WebRTC statistics objects.
<!-- RTCStats -->"inbound-rtp", which indicates the type of statistics that the object contains.{{Specifications}}
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