Ant Media Server Cloud

Best Self Hosted Alternatives to Ant Media Server Cloud

A curated collection of the 8 best self hosted alternatives to Ant Media Server Cloud.

Cloud-hosted real-time media server providing ultra-low-latency live streaming and WebRTC for interactive video applications. Supports HLS, RTMP, RTSP, SRT, Zixi and CMAF; offers auto-scaling, transcoding, SDKs and integration APIs.

Alternatives List

#1
SRS (Simple Realtime Server)

SRS (Simple Realtime Server)

SRS (Simple Realtime Server) is a high-efficiency media server supporting RTMP, WebRTC, HLS, HTTP-FLV, SRT, MPEG-DASH, and GB28181 for real-time streaming.

SRS (Simple Realtime Server) screenshot

SRS (Simple Realtime Server) is an open-source, high-performance real-time media server designed for building live streaming and real-time communication (RTC) services. It acts as a streaming gateway that ingests and delivers media across multiple protocols with an emphasis on low latency and efficiency.

Key Features

  • Multi-protocol streaming support including RTMP, WebRTC, HLS, HTTP-FLV, SRT, MPEG-DASH, and GB28181
  • Broad codec compatibility, including H.264, H.265/HEVC, AV1, VP9, AAC, Opus, and G.711
  • Designed for high throughput and low-latency delivery for live streaming and RTC scenarios
  • Docker-friendly deployments and support for cloud-native workflows
  • Built-in observability support (commonly used with Prometheus exporters)

Use Cases

  • Live streaming platforms that need RTMP ingest with HLS/HTTP-FLV/WebRTC playback
  • Low-latency streaming for interactive events, gaming, and real-time broadcasts
  • Video gateway services bridging traditional streaming protocols with WebRTC

Limitations and Considerations

  • Primarily focused on media transport/gateway capabilities; a complete video platform typically requires additional components (players, authentication, recording/transcoding pipelines)

SRS is a strong fit when you need a reliable, efficient media server with broad protocol support and production-oriented performance characteristics. It is commonly used as a core building block in custom live streaming and WebRTC solutions.

28.4kstars
5.6kforks
#2
Owncast

Owncast

Owncast is a self-hosted live streaming server that supports RTMP ingest, HLS playback, and built-in web chat, with optional Fediverse (ActivityPub) integration.

Owncast screenshot

Owncast is a free and open source live video streaming and web chat server that you run on your own infrastructure. It is designed for single-channel creators and works with common broadcasting tools by ingesting RTMP and delivering streams to viewers via the web.

Key Features

  • RTMP ingest compatibility with popular broadcasters (for example, OBS and similar tools)
  • HLS-based web playback with an integrated viewer page
  • Built-in live chat, including support for custom emotes and community interaction
  • Admin interface for stream configuration, moderation, and managing the viewer experience
  • Optional Fediverse integration via ActivityPub so people can follow and share streams across compatible networks
  • Embeddable player and chat components for integrating into other sites

Use Cases

  • Run an independent live stream for a community, club, or small organization
  • Host creator live streams without relying on centralized streaming platforms
  • Add live video and chat to an existing website using embeds

Limitations and Considerations

  • Primarily designed for a single streamer/channel rather than multi-tenant streaming platforms
  • Does not natively support running as a Windows server (typically run on Linux; WSL2 may be used on Windows)

Owncast provides a straightforward way to control your live content, audience experience, and chat community from a server you manage. It is well-suited for creators who want a lightweight, independent alternative with familiar broadcasting workflows.

10.9kstars
1.2kforks
#3
Janus WebRTC Server

Janus WebRTC Server

Janus WebRTC Server is a lightweight, general-purpose WebRTC gateway for building real-time audio/video applications via plugins and a JSON-based API.

Janus WebRTC Server screenshot

Janus WebRTC Server is a general-purpose WebRTC server (gateway) designed to establish and manage real-time media sessions with browsers and other WebRTC endpoints. It focuses on handling WebRTC signaling and media transport, while application logic is provided through a modular plugin architecture.

Key Features

  • Pluggable server-side modules to implement features like echo tests, conferencing, recording, streaming, and SIP gatewaying
  • JSON-based signaling and control API for interacting with plugins
  • WebRTC media relay handling RTP/RTCP between peers and server-side components
  • Optional API transports such as HTTP REST and WebSockets (build-time optional)
  • Designed for a small footprint and deployment flexibility on Linux (and can be compiled on macOS)

Use Cases

  • Building custom WebRTC conferencing, audio-bridge, or streaming applications
  • Adding WebRTC access to SIP/VoIP infrastructures via gateway plugins
  • Implementing media relays, recorders, and test utilities for real-time communications

Limitations and Considerations

  • Functionality depends on plugins; the core server does not provide a full application by itself
  • Windows is not officially supported (commonly run via Linux environments such as WSL)

Janus is best suited for developers and operators who need a high-performance WebRTC gateway with flexible, modular capabilities. Its plugin-based design makes it adaptable to many RTC architectures while keeping the core lightweight.

9kstars
2.6kforks
#4
datarhei Restreamer

datarhei Restreamer

Self-hosted live streaming server to ingest RTMP/SRT/RTSP, transcode with FFmpeg, and restream to platforms like YouTube Live and Twitch with an easy web UI.

datarhei Restreamer screenshot

datarhei Restreamer is a self-hosted live streaming server that ingests video from sources like OBS or IP cameras, processes streams via FFmpeg, and publishes them to your website and/or external platforms. It combines a browser-based UI with streaming protocols and a documented API to manage inputs, outputs, and monitoring.

Key Features

  • Ingest and publish using common streaming protocols (including RTMP, SRT, and HLS)
  • Restream a single input to multiple outputs (e.g., social platforms or other streaming servers)
  • Web-based setup wizard and streamlined administration interface
  • Built-in embeddable web player and optional publication page
  • FFmpeg-based transcoding and processing, including muxing separate audio
  • Hardware-accelerated encoding support (e.g., NVIDIA CUDA), depending on deployment
  • Bandwidth/viewer monitoring and optional limiting
  • REST API with OpenAPI/Swagger documentation
  • TLS automation with Let’s Encrypt for HTTPS (deployment-dependent)

Use Cases

  • Publish a live stream to your own website while simultaneously restreaming to major platforms
  • Centralize ingest from OBS and distribute to multiple RTMP/SRT endpoints
  • Create a lightweight streaming gateway for events using commodity hardware (including SBCs)

Limitations and Considerations

  • Reliable transcoding and multi-output restreaming can require significant CPU/GPU resources
  • External publishing depends on the capabilities and limits of the destination platforms

Restreamer is a practical choice for organizations and creators that want to control their streaming stack while keeping setup approachable. Its UI-driven configuration, multi-protocol support, and API make it suitable for both simple live embeds and more advanced distribution workflows.

4.8kstars
516forks
#5
FreeSWITCH

FreeSWITCH

FreeSWITCH is an open-source software-defined telecom stack for building VoIP and real-time communication services with SIP and WebRTC support.

FreeSWITCH screenshot

FreeSWITCH is an open-source, software-defined telecom stack used to build and run voice, video, and real-time communications infrastructure on commodity hardware. It is commonly used as a softswitch and media server for SIP-based telephony and can also provide browser-based calling via WebRTC.

Key Features

  • SIP signaling and media handling suitable for softswitch and PBX-style deployments
  • Native WebRTC capabilities for browser-based real-time communications
  • Modular architecture with loadable modules for telephony features and integrations
  • Built-in conferencing and real-time media services (including mixing and related functions)
  • Common voice features such as call routing, IVR building blocks, call recording, and voicemail support

Use Cases

  • Build and operate a SIP softswitch or VoIP application server for carriers and enterprises
  • Implement IVR, call routing, and contact-center style call flows
  • Provide WebRTC calling to web applications without requiring a separate gateway

Limitations and Considerations

  • Configuration and operation can be complex for newcomers due to the breadth of telecom concepts and module options
  • Certain advanced capabilities may be available only via commercial modules or enterprise distributions

FreeSWITCH is a mature foundation for telecom and real-time media systems, from embedded devices to large-scale deployments. Its modular design and protocol support make it a flexible core for custom telephony platforms and communication products.

4.6kstars
1.7kforks
#6
OvenMediaEngine

OvenMediaEngine

OvenMediaEngine (OME) is a sub-second latency live streaming server that ingests multiple protocols, transcodes to ABR, and delivers streams via WebRTC and Low-Latency HLS.

OvenMediaEngine screenshot

OvenMediaEngine (OME) is a low-latency live streaming server designed for large-scale, high-definition delivery. It can ingest live inputs via multiple broadcast protocols, optionally transcode them, and deliver streams to viewers using WebRTC and Low-Latency HLS.

Key Features

  • Multi-protocol ingest and pull, including WebRTC, SRT, RTMP, RTSP, and MPEG-2 TS
  • Sub-second playback via WebRTC and low-latency delivery via LL-HLS
  • Embedded live transcoder with adaptive bitrate (ABR) output
  • Origin-edge clustering model for scalable deployments
  • DVR (live rewind), file recording, and dump-to-VOD workflows
  • WebRTC signaling over WebSocket and support for WebRTC over TCP with embedded TURN
  • Access control features including signed policies and admission webhooks
  • Monitoring and REST API for automation and operational integration

Use Cases

  • Low-latency interactive live events and auctions using WebRTC playback
  • Large-scale live broadcasting with ABR output and edge distribution
  • Streaming platform backends that need to accept RTMP/SRT and deliver LL-HLS/WebRTC

Limitations and Considerations

  • Some advanced capabilities (for example certain DRM workflows) may require careful client/player compatibility and licensing considerations
  • Operational tuning (ports, UDP reachability, TURN behavior, origin-edge topology) is important to achieve consistent sub-second latency

OvenMediaEngine is well-suited for teams building their own live streaming infrastructure where ultra-low latency and protocol flexibility are key. It combines ingest, transcoding, and delivery in one server to simplify building scalable real-time streaming services.

3kstars
1.1kforks
#7
La Suite Meet

La Suite Meet

Self-hostable web video conferencing app built on LiveKit, with large-meeting performance, screen sharing, chat, recordings, and transcription features.

La Suite Meet is an open source video conferencing application designed for browser-based meetings with strong performance and modern collaboration features. It is built on top of LiveKit to provide reliable real-time audio/video and scalable large meetings.

Key Features

  • Browser-based video meetings with LiveKit-powered real-time media
  • Optimized for stability in large meetings (100+ participants)
  • Multiple simultaneous screen-sharing streams
  • Non-persistent secure meeting chat
  • Meeting recording support
  • Transcription and meeting summaries (beta)
  • Telephony integration
  • Authentication and access control for secure participation
  • Customizable frontend styling

Use Cases

  • Internal meetings for organizations that need a self-controlled conferencing stack
  • Large webinars or all-hands meetings requiring stable performance at scale
  • Secure video calls with access control and optional recording/transcription

Limitations and Considerations

  • Some advanced features (such as recording and transcription) may have limited installation/operations documentation depending on your deployment
  • End-to-end encryption is listed as a planned feature rather than generally available

La Suite Meet is a practical choice for teams that want a modern video conferencing experience with a web client and a scalable media backend. It fits especially well in environments needing tighter control over authentication, access, and deployment options.

981stars
108forks
#8
MistServer

MistServer

MistServer is an open-source streaming media toolkit that supports HLS, DASH, RTMP, RTSP, SRT and WebRTC for low-latency live and VOD workflows.

MistServer screenshot

MistServer is a full-featured open-source streaming media toolkit for OTT, live and VOD workflows. It provides a modular controller-based architecture, a web management interface and an API for automation and integration.

Key Features

  • Broad protocol support for ingest and egress including HLS (CMAF/TS), MPEG-DASH, RTMP, RTSP, SRT, RIST and WebRTC for low-latency delivery.
  • Wide container and codec compatibility (MP4, MKV, TS, FLV; H.264, H.265/HEVC, AV1, VP8/VP9 and common audio codecs) with configurable transmuxing and live MP4/recording options.
  • Low-latency capabilities via WebRTC (including WHIP/WHEP variants), SRT and LL-HLS plus options for segmenting/transmuxing for different player targets.
  • Modular runtime: MistController discovers and runs Mist* binaries, web UI listens on port 4242 and a programmable API and trigger system enable automation and integration.
  • Built for building from source with Meson/Ninja; optional ffmpeg integration for encoding/transcoding processes and optional libsrt/librist support; official Docker assets and prebuilt binaries are provided.

(mistserver.org)

Use Cases

  • OTT streaming platform: multi-protocol delivery (HLS/DASH) for adaptive bitrate delivery to browsers, mobile apps and set-top boxes.
  • Ultra/low-latency streaming and preview: WebRTC, SRT or RIST for real-time monitoring, remote production and interactive streams.
  • VOD hosting and live-to-VOD workflows: on-the-fly transmuxing, recording to MP4/MKV/TS and integration with storage and analytics pipelines.

Limitations and Considerations

  • Feature variance between editions: the open-source edition omits several Pro features (DRM, some access-control features, certain recording/analytics/process tools), so production needs requiring DRM or enterprise support should verify edition capabilities.

(mistserver.org)

MistServer is practical for developers and integrators who need a flexible, protocol-rich media server with programmatic control and multiple output formats. It is optimized for Linux-based deployments and provides tooling for compilation, container deployment and integration.

472stars
144forks

Why choose an open source alternative?

  • Data ownership: Keep your data on your own servers
  • No vendor lock-in: Freedom to switch or modify at any time
  • Cost savings: Reduce or eliminate subscription fees
  • Transparency: Audit the code and know exactly what's running