Search Results for "c webrtc"
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Showing 6 open source projects for "c webrtc"
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Forever Free Full-Stack Observability | Grafana CloudBuilt on open standards like Prometheus and OpenTelemetry, Grafana Cloud includes Kubernetes Monitoring, Application Observability, Incident Response, plus the AI-powered Grafana Assistant. Get started with our generous free tier today.
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Go From AI Idea to AI App FastAccess Gemini 3 and 200+ models. Build chatbots, agents, or custom models with built-in monitoring and scaling.
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SRS
SRS is a simple, high efficiency and realtime video server
SRS is a simple, high-efficiency and real-time video server that supports RTMP, WebRTC, HLS, HTTP-FLV and SRT. SRS is licenced under MIT or MulanPSL-2.0, and note that MulanPSL-2.0 is compatible with Apache-2.0, but some third-party libraries are distributed using their own licenses. Besides of FFmpeg or OBS, it's also able to publish by H5 if WebRTC is enabled, please remember to set the CANDIDATE for WebRTC. Highly recommend that directly run SRS by docker, Cloud Virtual Machine, or K8s,... -
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ZLMediaKit
WebRTC/RTSP/RTMP/HTTP/HLS/HTTP-FLV/WebSocket
ZLMediaKit is a high-performance C++11 media server and streaming SDK for real-time audio and video applications. It supports a wide range of protocols, including RTSP, RTMP, HLS, HTTP-FLV, WebSocket-FLV, GB28181, HTTP-TS, HTTP-fMP4, MP4, and WebRTC. The project is designed for protocol conversion, live streaming, low-latency playback, recording, forwarding, and large-scale client access. -
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AnalysisAVP
Encode decode, rgb yuv h264 aac flv mp4 rtmp
AnalysisAVP is a comprehensive educational repository focused on audio and video technology concepts, providing structured knowledge across multimedia systems and processing pipelines. It covers foundational topics such as encoding, decoding, color formats like RGB and YUV, and widely used codecs including H.264 and AAC. The project also explores media container formats like MP4 and FLV, along with streaming protocols such as RTMP and WebRTC, offering a broad understanding of media... -
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Telegram WebRTC (VoIP)
Voice chats, private incoming and outgoing calls in Telegram
Telegram WebRTC (VoIP) is a Python and C++ library that enables real-time voice and video communication features for Telegram bots and clients. It provides an interface for joining, managing, and streaming audio or video in Telegram group calls and voice chats. The library is built on top of low-level communication protocols, ensuring efficient handling of real-time media streams. -
Stop Storing Third-Party Tokens in Your DatabaseRolling your own OAuth token storage can be a security liability. Token Vault securely stores access and refresh tokens from federated providers and handles exchange and renewal automatically. Connected accounts, refresh exchange, and privileged worker flows included.
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anyLive
RTMP streamer, RTMP (HLS) second-on player, live streaming on demand
anyLive is a cross-platform live streaming framework that provides a complete end-to-end solution for audio and video capture, encoding, transmission, decoding, and rendering. Built on a C++ core and leveraging WebRTC and RTMP technologies, it supports multiple operating systems including Android, iOS, Windows, and Linux using a unified codebase. The project is designed for real-time communication and live broadcasting applications, offering low-latency streaming and compatibility with standard RTMP servers and CDNs. ... -
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MCU Media Server
SIP Video Multiconference Media Server with WebRTC support.
REPOSITORY MOVED TO GITHUB!! https://github.com/medooze/media-server Video Multiconference Media Server with WebRTC support. Provide Multiconference and video broadcasting services to any SIP service. Supports VP8, H264, MP4V-ES, H263 and H263P, continuous presence, RTMP flash broadcasting, adhoc conferences, load balancing and administrative WEB interface. JSR309 driver implementation under development. .
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