RTCRtpReceiver
E24809
RTCRtpReceiver is a WebRTC API interface that represents the receiving end of an RTP media stream, providing access to incoming audio or video tracks and related reception statistics.
All labels observed (1)
| Label | Occurrences |
|---|---|
| RTCRtpReceiver canonical | 3 |
How this entity was disambiguated
This entity first appeared as the object of triple T193067 — resolving that mention is where its identity was fixed. The disambiguator weighed these candidate entities and picked the highlighted one (or “None”, minting a new entity). This is how homonymy is resolved: the same surface form can point to different entities.
NED1
Entity disambiguation (via context triple)
gpt-5-mini-2025-08-07
Target entity: RTCRtpReceiver Context triple: [WebRTC, hasComponent, RTCRtpReceiver]
-
A.
RTCIceCandidate
RTCIceCandidate is a WebRTC interface representing a potential network route (ICE candidate) used during peer-to-peer connection negotiation.
-
B.
WebRTC
WebRTC is an open web technology that enables real-time audio, video, and data communication directly between browsers and devices without requiring plugins.
-
C.
SRTP for secure media transport
SRTP for secure media transport is a security protocol that encrypts and authenticates real-time audio and video streams, commonly used in VoIP and WebRTC communications.
-
D.
ORTC
ORTC (Object Real-Time Communications) is a JavaScript API and architecture for enabling real-time audio, video, and data communications on the web, serving as an alternative and evolutionary step beyond the traditional WebRTC 1.0 model.
-
E.
QUIC
QUIC is a modern, multiplexed transport protocol developed by Google and standardized by the IETF that runs over UDP to provide faster, more secure, and reliable web connections than traditional TCP-based HTTPS.
- F. None of above. chosen
- G. Unsure - the case is ambiguous/there is not enough information to decide.
NED2
Entity disambiguation (via description)
gpt-5-mini-2025-08-07
Target entity: RTCRtpReceiver Target entity description: RTCRtpReceiver is a WebRTC API interface that represents the receiving end of an RTP media stream, providing access to incoming audio or video tracks and related reception statistics.
-
A.
RTCIceCandidate
RTCIceCandidate is a WebRTC interface representing a potential network route (ICE candidate) used during peer-to-peer connection negotiation.
-
B.
WebRTC
WebRTC is an open web technology that enables real-time audio, video, and data communication directly between browsers and devices without requiring plugins.
-
C.
SRTP for secure media transport
SRTP for secure media transport is a security protocol that encrypts and authenticates real-time audio and video streams, commonly used in VoIP and WebRTC communications.
-
D.
ORTC
ORTC (Object Real-Time Communications) is a JavaScript API and architecture for enabling real-time audio, video, and data communications on the web, serving as an alternative and evolutionary step beyond the traditional WebRTC 1.0 model.
-
E.
QUIC
QUIC is a modern, multiplexed transport protocol developed by Google and standardized by the IETF that runs over UDP to provide faster, more secure, and reliable web connections than traditional TCP-based HTTPS.
- F. None of above. chosen
Statements (47)
| Predicate | Object |
|---|---|
| instanceOf |
RTP receiver
ⓘ
WebRTC API interface ⓘ |
| associatedWith |
MediaStreamTrack
ⓘ
RTCRtpSender ⓘ RTCRtpTransceiver ⓘ |
| canBeObtainedFrom |
RTCPeerConnection.getTransceivers
ⓘ
RTCRtpTransceiver.receiver ⓘ |
| canProvide |
RTP reception statistics
ⓘ
jitter information ⓘ packet loss information ⓘ round-trip time estimates via RTCP stats ⓘ |
| createdByMethod | RTCPeerConnection.getReceivers ⓘ |
| definedIn |
W3C WebRTC specification
ⓘ
linked to:
WebRTC
WebRTC 1.0 specification ⓘ
linked to:
WebRTC
|
| exposes |
incoming media track
ⓘ
reception statistics ⓘ |
| handlesMediaType |
audio
ⓘ
video ⓘ |
| hasAttribute |
receiverId (implementation-specific in some browsers)
ⓘ
rtcpTransport ⓘ track ⓘ transport ⓘ |
| hasMethod |
getContributingSources()
ⓘ
getParameters() ⓘ getStats() ⓘ getSynchronizationSources() ⓘ playoutDelayHint (attribute or parameter in some specs/implementations) ⓘ |
| hasPurpose | represent the receiving end of an RTP media stream ⓘ |
| hasSecurityConsideration | exposes network and quality metrics via getStats() ⓘ |
| hasSpecificationStatus | Recommendation (via WebRTC 1.0) ⓘ |
| implementedIn |
JavaScript runtime environments with WebRTC support
ⓘ
web browsers ⓘ |
| mediaDirection | receive-only (from the receiver perspective) ⓘ |
| namespace | global WindowOrWorkerGlobalScope in browsers ⓘ |
| partOf | WebRTC ⓘ |
| relatedTo |
RTCRtpEncodingParameters
ⓘ
RTCRtpReceiveParameters ⓘ |
| returnsFromMethod | RTCPeerConnection.addTrack ⓘ |
| standardizedBy |
W3C
ⓘ
linked to:
World Wide Web Consortium
|
| subjectTo | same-origin policy for JavaScript access ⓘ |
| supportsProtocol |
RTCP
ⓘ
RTP ⓘ |
| usedFor |
receiving remote audio in WebRTC
ⓘ
receiving remote video in WebRTC ⓘ |
| usedWith | RTCPeerConnection ⓘ |
| worksOver | ICE-selected network path ⓘ |
| worksWith | DTLS-SRTP encrypted media ⓘ |
How these facts were elicited
The pipeline generated the facts above by prompting gpt-5.1 with this entity's name + description and the instruction below.
Instruction
You are a knowledge base construction expert. Given a subject entity and a description of it, return factual statements that you know for the subject as a JSON list of dictionaries(triples), where keys must be "subject", "predicate" and "object". The number of facts may be very high, between 25 to 50 or more, for very popular subjects. For less popular subjects, the number of facts can be very low, like 5 or 10. # Requirements - If you don't know the subject at all, return an empty list. - If the subject is not a named entity, return an empty list. - Include at least one triple where predicate is "instanceOf". - Do not get too wordy. - Separate several objects into multiple triples with one object.
Input
Subject: RTCRtpReceiver Description of subject: RTCRtpReceiver is a WebRTC API interface that represents the receiving end of an RTP media stream, providing access to incoming audio or video tracks and related reception statistics.
Referenced by (3)
Full triples — surface form annotated when it differs from this entity's canonical label.