vr000m Goto Github PK

rmcat-cc-app-interaction

Interactive real-time media applications that use the Real-time Transport Protocol (RTP) over the User Datagram Protocol (UDP) must use congestion control techniques above the UDP layer since it provides none. This memo describes the interactions and interfaces necessary between the application components that relate to congestion control, including the RTP layer, the higher-level media codec control layer, and the lower-level transport interface and the congestion control layer.

rmcat-eval-test

The Real-time Transport Protocol (RTP) is used to transmit media in multimedia telephony applications, these applications are typically required to implement congestion control. The RMCAT working group is currently working on candidate algorithms for such interactive real-time multimedia applications. This document describes the test cases to be used in the performance evaluation of those candidate algorithms.

rmcat-feedback-message

This repo contains the draft on a generic feedback message for congestion control for real-time interactivce media applications. This is a outcome of RMCAT desing team. For more information about IETF RMCAT working group visit : https://datatracker.ietf.org/wg/rmcat/charter/

rtvi-client-web

rtvi-web-demo

samples

WebRTC Web demos and samples

sensortrafficgenerator

The code generates random traffic for an IoT simulator

snapper

Command Line Utility to send TCP RST packets to endpoints in your WLAN (subnet)

taps-transports

Services provided by IETF transport protocols and congestion control mechanisms (IETF TAPS WG document)

ucx

Unified Communication X (mailing list - https://elist.ornl.gov/mailman/listinfo/ucx-group)

visionllm

VisionLLM: Large Language Model is also an Open-Ended Decoder for Vision-Centric Tasks

webrtc-pc

WebRTC 1.0 API

webrtc-rtptransport

Repository for the RTPTransport specification of the WebRTC Working Group

wt-demo

WebTransport Sample Code

xrblock-independent-burst-gap-discard

This document defines an RTP Control Protocol (RTCP) Extended Report (XR) block that allows the reporting of burst and gap discard metrics independently of the burst and gap loss metrics for use in a range of RTP applications.

xrblock-rtcweb-rtcp-xr-metrics

This document describes monitoring features related to media streams in WebRTC. It provides a list of RTCP XR metrics, which are useful and may need to be supported in some RTCWEB implementations. It also describes a list of identifiers for WebRTC's statistics API. These identifiers are a set of RTCP SR, RR, and XR metrics related to the transport of multimedia flows.