RTCPeerConnection
E127263
RTCPeerConnection is a WebRTC API interface that enables real-time audio, video, and data communication between web browsers or devices by managing the underlying peer-to-peer connection.
All labels observed (1)
| Label | Occurrences |
|---|---|
| RTCPeerConnection canonical | 5 |
How this entity was disambiguated
This entity first appeared as the object of triple T1096710 — 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: RTCPeerConnection Context triple: [RTCIceCandidate, createdBy, RTCPeerConnection]
-
A.
WebRTC
WebRTC is an open web technology that enables real-time audio, video, and data communication directly between browsers and devices without requiring plugins.
-
B.
RTCIceCandidate
RTCIceCandidate is a WebRTC interface representing a potential network route (ICE candidate) used during peer-to-peer connection negotiation.
-
C.
RTCRtpReceiver
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.
-
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.
IETF RTCWEB Working Group
The IETF RTCWEB Working Group is a standards body group within the Internet Engineering Task Force responsible for defining the protocols and architecture that enable real-time communication capabilities in web browsers and applications.
- 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: RTCPeerConnection Target entity description: RTCPeerConnection is a WebRTC API interface that enables real-time audio, video, and data communication between web browsers or devices by managing the underlying peer-to-peer connection.
-
A.
WebRTC
WebRTC is an open web technology that enables real-time audio, video, and data communication directly between browsers and devices without requiring plugins.
-
B.
RTCIceCandidate
RTCIceCandidate is a WebRTC interface representing a potential network route (ICE candidate) used during peer-to-peer connection negotiation.
-
C.
RTCRtpReceiver
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.
-
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.
IETF RTCWEB Working Group
The IETF RTCWEB Working Group is a standards body group within the Internet Engineering Task Force responsible for defining the protocols and architecture that enable real-time communication capabilities in web browsers and applications.
- F. None of above. chosen
Statements (75)
| Predicate | Object |
|---|---|
| instanceOf |
JavaScript API
ⓘ
WebRTC interface ⓘ browser API ⓘ |
| availableIn |
Chrome
ⓘ
linked to:
Google Chrome
Edge ⓘ Firefox ⓘ
linked to:
Mozilla Firefox
Safari ⓘ |
| definedIn |
WebRTC 1.0 specification
ⓘ
linked to:
WebRTC
|
| enables |
end-to-end media encryption
ⓘ
low-latency communication ⓘ |
| hasConstructor | RTCPeerConnection() ⓘ |
| hasEvent |
connectionstatechange
ⓘ
datachannel ⓘ icecandidate ⓘ icecandidateerror ⓘ iceconnectionstatechange ⓘ icegatheringstatechange ⓘ negotiationneeded ⓘ signalingstatechange ⓘ track ⓘ |
| hasMethod |
addIceCandidate
ⓘ
addTrack ⓘ addTransceiver ⓘ close ⓘ createAnswer ⓘ createDataChannel ⓘ createOffer ⓘ getReceivers ⓘ getSenders ⓘ getStats ⓘ getTransceivers ⓘ removeTrack ⓘ setLocalDescription ⓘ setRemoteDescription ⓘ |
| hasProperty |
canTrickleIceCandidates
ⓘ
connectionState ⓘ currentLocalDescription ⓘ currentRemoteDescription ⓘ iceConnectionState ⓘ iceGatheringState ⓘ localDescription ⓘ pendingLocalDescription ⓘ pendingRemoteDescription ⓘ remoteDescription ⓘ signalingState ⓘ |
| partOf | WebRTC ⓘ |
| relatedTo |
MediaStream
ⓘ
MediaStreamTrack ⓘ RTCDataChannel ⓘ RTCRtpReceiver ⓘ RTCRtpSender ⓘ RTCRtpTransceiver ⓘ |
| runsIn |
web browser
ⓘ
worker context ⓘ |
| securityProperty | requires HTTPS origin in browsers ⓘ |
| standardizedBy |
W3C
ⓘ
linked to:
World Wide Web Consortium
|
| supportsProtocol |
RTCP
ⓘ
RTP ⓘ |
| supportsTransport |
DTLS
ⓘ
SRTP ⓘ
linked to:
SRTP for secure media transport
TCP ⓘ
linked to:
Transmission Control Protocol
UDP ⓘ |
| usedBy |
getUserMedia-based applications
ⓘ
real-time multiplayer games ⓘ video conferencing applications ⓘ voice over IP applications ⓘ |
| usedFor |
audio streaming
ⓘ
data communication ⓘ peer-to-peer communication ⓘ real-time communication ⓘ video streaming ⓘ |
| usesMechanism |
ICE
ⓘ
SDP ⓘ STUN ⓘ TURN ⓘ |
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: RTCPeerConnection Description of subject: RTCPeerConnection is a WebRTC API interface that enables real-time audio, video, and data communication between web browsers or devices by managing the underlying peer-to-peer connection.
Referenced by (5)
Full triples — surface form annotated when it differs from this entity's canonical label.