Triple
T1166455
| Position | Surface form | Disambiguated ID | Type / Status |
|---|---|---|---|
| Subject | RTCRtpReceiver |
E24809
|
entity |
| Predicate | relatedTo |
P37
|
FINISHED |
| Object |
RTCRtpEncodingParameters
RTCRtpEncodingParameters is a WebRTC configuration object that defines how a particular RTP stream is encoded, including settings like bitrate, scalability, and redundancy for real-time audio or video transmission.
|
E134377
|
NE FINISHED |
How this triple was built (4 steps)
Every LLM step that produced this triple, in pipeline order — named-entity classification, the disambiguation choices (the exact options shown, with the pick highlighted), and the generated description. The batch + timestamp of each is in the Provenance table below.
NER
Named-entity recognition
gpt-5-mini
Instruction
Given a phrase, classify it is english named entity (e.g., persons, organizations, works of art) in Latin script, or not (e.g., literals, dates, URLs, verbose phrases). For disambiguation, the statement where the phrase occurs as object is also given. Please return a JSON object with `phrase` (string, the phrase being analyzed) and `is_ne` (boolean, indicating whether the phrase is a Named Entity).
Input
Phrase: RTCRtpEncodingParameters | Statement: [RTCRtpReceiver, relatedTo, RTCRtpEncodingParameters]
NED1
Entity disambiguation (via context triple)
gpt-5-mini-2025-08-07
Target entity: RTCRtpEncodingParameters Context triple: [RTCRtpReceiver, relatedTo, RTCRtpEncodingParameters]
-
A.
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.
-
B.
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.
-
C.
G.722
G.722 is a wideband audio codec standard that provides higher-quality voice transmission than traditional narrowband codecs, commonly used in VoIP and teleconferencing applications.
-
D.
G.729
G.729 is a widely used ITU-T audio codec standard that compresses voice for bandwidth-efficient transmission in VoIP and other telephony applications.
-
E.
G.711
G.711 is a widely used ITU-T audio codec standard for pulse code modulation (PCM) of voice in traditional telephony and VoIP systems.
- F. None of above. chosen
- G. Unsure - the case is ambiguous/there is not enough information to decide.
NEDg
Description generation
gpt-5.1
Instruction
Generate a one-sentence description of the target entity. You are given a context triple in the form (subject, predicate, object), where the object is the target entity. # Instructions Use the triple to infer relevant information about the entity. Describe the entity based on what is most defining, well-known. Avoid repeating the information from the triple, unless really essential. # Response Format Return only the sentence: "Description: [one-sentence description of the target entity]"
Input
Entity: RTCRtpEncodingParameters Triple: [RTCRtpReceiver, relatedTo, RTCRtpEncodingParameters]
Generated description
RTCRtpEncodingParameters is a WebRTC configuration object that defines how a particular RTP stream is encoded, including settings like bitrate, scalability, and redundancy for real-time audio or video transmission.
NED2
Entity disambiguation (via description)
gpt-5-mini-2025-08-07
Target entity: RTCRtpEncodingParameters Target entity description: RTCRtpEncodingParameters is a WebRTC configuration object that defines how a particular RTP stream is encoded, including settings like bitrate, scalability, and redundancy for real-time audio or video transmission.
-
A.
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.
-
B.
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.
-
C.
G.722
G.722 is a wideband audio codec standard that provides higher-quality voice transmission than traditional narrowband codecs, commonly used in VoIP and teleconferencing applications.
-
D.
G.729
G.729 is a widely used ITU-T audio codec standard that compresses voice for bandwidth-efficient transmission in VoIP and other telephony applications.
-
E.
G.711
G.711 is a widely used ITU-T audio codec standard for pulse code modulation (PCM) of voice in traditional telephony and VoIP systems.
- F. None of above. chosen
Provenance (5 batches)
The batch behind each pipeline step, in order, with when it ran. Timestamps are batch-level — stages were processed in waves, so the object chain (NER → NED1 → NEDg → NED2) reads in order, but predicate / elicitation batches can sit in a different wave.
| Step | Stage | Batch ID | Status | When |
|---|---|---|---|---|
| creating | Elicitation | batch_69a494082a7c819095004f423f294a64 |
completed | March 1, 2026, 7:31 p.m. |
| NER | Named-entity recognition | batch_69a4bccc62a88190882d8801908015a4 |
completed | March 1, 2026, 10:25 p.m. |
| NED1 | Entity disambiguation (via context triple) | batch_69ac668290008190a5e175c0a2e5feb0 |
completed | March 7, 2026, 5:55 p.m. |
| NEDg | Description generation | batch_69ac6aaa84fc81908130543b14938e41 |
completed | March 7, 2026, 6:12 p.m. |
| NED2 | Entity disambiguation (via description) | batch_69ac6b24987c819080e00b49df8f6edb |
completed | March 7, 2026, 6:15 p.m. |
Created at: March 1, 2026, 7:45 p.m.