Best Self-hosted Alternatives to Amazon Chime

Mattermost is an open-source, self-hosted collaboration platform designed for secure team communication and real-time collaboration in high-stakes environments. It supports flexible deployment options, including on-premises or private cloud setups, with enterprise-grade security, governance, and integration capabilities.

Key Features

• Self-hosted deployment with high availability for mission-critical use
• Native audio calls and screen sharing within channels
• Interoperability with external tools (MS Teams, GitLab, Atlassian) and native playbooks
• Open-source Team Edition licensed under MIT
• Data control, granular admin and deployment flexibility (air-gapped networks, on-prem/private cloud)
• Real-time DevSecOps collaboration with integrated playbooks and automation

Use Cases

• Integrated Security Operations and incident response coordination
• Real-time DevSecOps collaboration to accelerate software delivery
• Self-sovereign collaboration and data governance for regulated environments

Limitations and Considerations

• Team Edition is intended for small teams and is not recommended for government or sensitive workloads

Conclusion: Mattermost offers a secure, self-hosted collaboration platform with deployment flexibility and strong integration capabilities, making it suitable for organizations needing data control and mission-critical workflows.

Jitsi Meet is an open source video conferencing application built on WebRTC, designed for secure, scalable meetings in the browser and on mobile devices. It can be used as a standalone meeting app or embedded into other products via SDKs.

Key Features

• Browser-based video meetings with modern WebRTC support
• Mobile applications for Android and iOS
• Screen/content sharing for presentations and collaboration
• In-meeting chat including private conversations
• Meeting controls such as raise hand, reactions, and polls
• Virtual backgrounds and common conferencing UX features
• Integration options via web and native SDKs

Use Cases

• Team video meetings with screen sharing and chat
• Hosting community calls, workshops, and webinars
• Embedding video meetings into an existing web or mobile application

Limitations and Considerations

• End-to-end encryption is available but may limit some features depending on configuration and client support

Jitsi Meet is a strong choice for organizations that want full control over video conferencing while keeping an easy, browser-first user experience. Its ecosystem and SDK support also make it well-suited for product teams building custom video experiences.

Zulip is an open-source team chat platform designed for both real-time and asynchronous communication. Its distinctive topic-based threading keeps conversations organized, making it easier to follow multiple discussions without losing context.

Key Features

• Topic-based threading within channels (streams) to keep discussions focused
• Inbox-style view to prioritize unread conversations
• Real-time messaging with searchable history
• Powerful integrations and bots, including webhook-based workflows
• Granular permissions and administration for organizations and communities
• Multi-platform clients, including web, desktop, and mobile apps

Use Cases

• Team communication for engineering, product, and operations groups
• Open source or community collaboration with many parallel discussions
• Async-first coordination across time zones while preserving context

Zulip is a strong fit for teams that want the immediacy of chat without sacrificing long-term clarity. Its conversation organization model scales well as the number of channels and active threads grows.

Screego is an open-source, self-hosted screen sharing tool designed for developers. It delivers high-quality, low-latency screen sharing via WebRTC and runs as a Docker container or a single binary.

Key Features

• Multi-user screen sharing
• Secure transfer via WebRTC
• Low latency and high-resolution streaming
• Easy deployment via Docker or a single binary
• Integrated TURN server for NAT traversal
• GPL-3.0 open-source and actively maintained
• Browser-based client (no dedicated desktop client)

Use Cases

• Developer code reviews, pair programming, and technical demos
• Remote debugging and collaborative troubleshooting
• Live training and technical presentations

Conclusion

Screego provides a focused, self-hosted alternative for teams needing private, high-quality screen sharing. It is built with Go and TypeScript, and a public demo instance is available for quick evaluation.

BigBlueButton is an open-source virtual classroom and web conferencing system designed primarily for online teaching and learning. It provides real-time collaboration tools for instructors and learners, with features tailored to classroom workflows.

Key Features

• Real-time audio and webcam video with selectable quality levels
• Screen sharing for presenters
• Slide and document presentation with whiteboard annotations
• Multi-user whiteboards for collaborative activities
• Public and private chat
• Breakout rooms for small-group work
• Polling and in-session feedback tools (such as hand raise)
• Shared notes for group collaboration
• Session recording and playback
• Learning analytics visible to moderators via a Learning Analytics Dashboard

Use Cases

• Live online classes, lectures, and seminars
• One-to-one tutoring and office hours
• Group collaboration sessions using breakout rooms and shared whiteboards

BigBlueButton is a feature-rich, education-focused conferencing platform that goes beyond general-purpose video meetings by emphasizing engagement, facilitation, and classroom-oriented tooling. It is widely used and commonly integrated into learning management systems for delivering virtual instruction at scale.

Mumble is an open-source voice chat platform focused on low-latency, high-quality audio for groups and communities. It consists of a desktop client (Mumble) and a server component (mumble-server, formerly Murmur) to host your own voice infrastructure.

Key Features

• Low-latency, high-quality voice communication using the Opus codec
• Encrypted communication with certificate-based (public/private key) authentication support
• Channels and an extensive permission system (ACL) for complex community setups
• In-game overlay and positional audio features for supported games and setups
• Extensibility via plugins and server integration APIs (including Ice-based tooling)
• Cross-platform client support (Windows, macOS, and Linux; server runs on many platforms)

Use Cases

• Private voice servers for gaming groups, clans, and large online communities
• Team voice communication for organizations that need control over privacy and access
• Audio collaboration and recording scenarios such as podcasts and multi-user sessions

Mumble is a mature, widely used VoIP solution that prioritizes responsiveness, sound quality, and administrative control. It is well-suited for anyone needing reliable hosted voice chat with fine-grained permissions and strong encryption.

Spacebar is a free and open source communication platform that reimplements Discord’s core user experience as a self-hostable, full-stack system. It aims for compatibility with Discord clients and bots while providing control over configuration, theming, and deployment.

Key Features

• Discord-compatible API behavior to reuse many existing clients and bots with minimal changes
• Real-time text chat with gateway-style events
• Voice and video calling via WebRTC
• Built-in content delivery (CDN-style) for media and attachments
• Admin dashboard and configurable limits for communities and deployments
• Theme system and design customization
• Plugin/extension approach for adding functionality
• Decentralization-oriented design goals to reduce single points of failure

Use Cases

• Host a Discord-like community server for a company, school, or gaming group
• Run a private collaboration chat with full control over data retention and policies
• Provide a compatible backend for custom clients, integrations, or bots

Limitations and Considerations

• Compatibility with all Discord features and third-party clients/bots may vary due to ongoing reimplementation work

Spacebar is a strong option for teams and communities that want a familiar Discord-style experience while retaining control over hosting, customization, and platform behavior. It is especially appealing when Discord API compatibility and real-time communication features are key requirements.

Jitsi Videobridge is an open-source WebRTC Selective Forwarding Unit (SFU) that routes media streams between participants to enable scalable multiparty video conferencing. It is a core backend component of the Jitsi Meet stack and is designed for high scalability and low CPU overhead.

Key Features

• Selective Forwarding Unit (SFU) architecture: forwards participant streams rather than mixing, improving quality and scalability for many-participant conferences.
• WebRTC-native media support: handles RTP/RTCP streams, common codecs, and NAT traversal (ICE/STUN/TURN) with secure transport via DTLS/SRTP.
• Control APIs: supports Colibri XMPP control as well as HTTPS/REST control endpoints for orchestration, load balancing, and automation.
• Observability: exports operational statistics and Prometheus-format metrics (private HTTP /metrics endpoint) plus JSON and XMPP-stat reports for monitoring and autoscaling.
• Implementation & build: primarily Kotlin and Java codebase, built with Maven and running on the JVM.
• Deployment options: packaged for Debian/Ubuntu, runnable locally via Maven, and commonly deployed in containerized or multi-node topologies for capacity.

Use Cases

• Powering large-scale multiparty video conferencing (used by Jitsi Meet) where many participants join without per-stream server mixing.
• Backend media routing for webinar platforms, remote education, telehealth, and enterprise meeting services requiring scalable, low-latency forwarding.
• Integrating into custom WebRTC applications that need a flexible SFU with XMPP or REST control and observability hooks.

Limitations and Considerations

• No server-side mixing/transcoding: Videobridge forwards streams (SFU) rather than acting as an MCU; server-side transcoding requires separate components or external transcoders.
• Prometheus coverage is substantial but not every internal metric is exposed in Prometheus format; some monitoring workflows rely on auxiliary exporters or the JSON/stats endpoints.
• Performance and capacity depend on network, codec choices, and server sizing; achieving maximum concurrent streams requires tuning (heap, GC, networking) and appropriate deployment topologies.

Jitsi Videobridge is a mature, open-source SFU for teams and developers who need scalable WebRTC media routing with XMPP/REST control and production monitoring. It is widely used in the Jitsi ecosystem and can be integrated into custom conferencing solutions or deployed as part of a larger real-time communications stack.

Chitchatter is an open-source, browser-first communication tool that enables private, peer-to-peer chat and media streaming directly between participants without requiring an API server. It emphasizes end-to-end encryption, ephemeral messaging, and decentralized connectivity for private conversations.

Key Features

• Peer-to-peer browser communication with TURN fallback for reliable connectivity.
• End-to-end encrypted text, audio, and video using WebRTC-based peer connections.
• Ephemeral message handling: message content is not persisted to disk and is cleared when leaving a room.
• File sharing with client-side encryption; transfers are encrypted and the room name functions as the key.
• Screen sharing, multi-peer rooms (limited by browser peer capacity), direct messaging, and conversation backfilling for late joiners.
• Embeddable via iframe and implemented as a modern web app (Vite + TypeScript + React tooling indicated in the repository).

Use Cases

• Secure ad-hoc conversations between individuals or small groups where no server-side message history is desired (private calls, sensitive discussions).
• Temporary collaboration with screen sharing and file exchange without creating accounts or storing data long-term.
• Embedding private chat functionality into other web apps via iframe for ephemeral support or short-lived group sessions.

Limitations and Considerations

• Multi-peer scalability is constrained by browser peer-connection limits; performance and reliability may degrade with many simultaneous direct connections.
• Connectivity relies on public WebTorrent and TURN relay servers when direct peer connections fail; availability or trust in those relays can affect reliability.
• There is no built-in persistent message history or centralized moderation/identity system; features like persistent archives, long-term user identity, or enterprise-grade moderation are not provided by default.

Chitchatter is a lightweight option for users seeking browser-native, ephemeral, and encrypted peer communication with media and file-transfer capabilities, designed for privacy-focused ad-hoc use. It is distributed as a client-side web application with optional server components for enhanced connectivity and integration.

Nextcloud Talk is a privacy-respecting, self-hosted communication platform that combines chat, video conferencing, and webinars in a single Nextcloud app. It supports on-premises deployment and federated collaboration across Nextcloud servers, with end-to-end encrypted calls and deep integration with other Nextcloud apps.

Key Features

• Threaded conversations with AI chat and call summaries
• End-to-end encrypted calls for private communication
• Federated chat and video calls across Nextcloud hubs and servers
• Webinars and presentations with screen sharing and participant controls
• Private, group and public calls with external participants via shareable links
• Tight integration with Nextcloud Files, Calendar and other apps for unified workflows
• Bridging with other networks via Nextcloud Bridging (e.g., IRC, Slack, Teams, Matrix)

Use Cases

• Hybrid teams needing private, on-premises collaboration across chat, calls and webinars
• Environments requiring GDPR/compliance and air-gap/self-hosted deployment
• Organizations collaborating across multiple Nextcloud servers or bridged networks

Limitations and Considerations

• NAT traversal often requires a TURN server for reliable media in restricted networks; behind symmetric NATs this becomes more critical
• Large-scale deployments usually need High Performance Backend (HPB) for thousands of participants; private setups typically handle dozens of calls depending on capacity
• Some advanced features depend on specific server components or configurations and may require additional infrastructure

Conclusion: Nextcloud Talk offers a privacy-first, self-hosted alternative for integrated chat, calls and webinars within the Nextcloud ecosystem. It emphasizes data control, security and federated collaboration, with deployment paths suitable for small teams up to large enterprises.

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.

Briefing is a privacy-focused video group chat application built on open web technologies. It enables secure, direct video conferencing in modern browsers without requiring additional software installation.

Key Features

• Browser-based video group chat using WebRTC
• End-to-end encrypted communication (E2EE)
• Simple invitation links to create and join rooms
• Text chat alongside video calls
• Desktop/screen sharing
• Includes core components needed to run (UI, signaling, and STUN support)
• Designed for customization and white-label deployments

Use Cases

• Private team or community video calls where minimizing data exposure matters
• Embedded video chat rooms for websites, events, or customer sessions
• Self-hosted video meetings for organizations with stricter privacy requirements

Limitations and Considerations

• Real-world reliability and media quality can depend heavily on network conditions and NAT traversal; TURN infrastructure may be required for some environments

Briefing provides a practical, secure foundation for WebRTC conferencing with an emphasis on privacy and simplicity. It is well-suited for both quick ad-hoc meetings and teams looking to customize or embed a video chat experience.

Databag is a fast, lightweight self-hosted messenger designed to run on minimal hardware while still supporting modern messaging features. It is federated, allowing users on different independently hosted nodes to communicate directly with strong cryptographic identities and end-to-end encryption.

Key Features

• Federated messaging between accounts on different nodes
• Public/private key based identity that is not tied to a hosting domain
• End-to-end encryption for sealed topics so the server admin cannot read content
• Topic-based threads to organize conversations by subject
• Unlimited participants in group threads
• Low-latency push using WebSockets (avoids polling)
• Optional audio and video calls using WebRTC (requires STUN/TURN for NAT traversal)
• Multi-factor authentication via TOTP
• Mobile push notifications (UnifiedPush, FCM, APN)

Use Cases

• Private family or small-community messaging on a personal server or low-power device
• Federated messaging between multiple organizations without a central provider
• Secure group threads for projects where message privacy is required

Limitations and Considerations

• WebRTC calls typically require configuring a STUN/TURN relay to work reliably behind NAT

Databag focuses on efficiency and decentralization while keeping the deployment simple enough for home servers. It fits best for users who want federated communication, end-to-end encryption, and optional calling without heavy infrastructure.

Galene is a self-hosted WebRTC videoconferencing system designed to be easy to deploy while using moderate server resources. It is well-suited to both one-to-many sessions like lectures and traditional team meetings, using an SFU-style architecture for scalable media forwarding.

Key Features

• Multi-party audio and video with arbitrary numbers of streams
• Text chat, user status indicators (for example “raise hand”), and moderation tools
• Screen and window sharing, including sharing multiple windows
• Recording to disk and media streaming from local files
• Built-in TURN server and robust ICE handling with automatic flow restarts
• Codec support including VP8/VP9 (with SVC and simulcast), partial H.264 support, and preliminary AV1 support
• Bandwidth estimation and congestion control for low-latency sessions
• Administrative HTTP API for managing groups and users
• Password-based and token-based (OAuth2-style) authorization options
• Support for WHIP protocol

Use Cases

• University lectures, seminars, and tutorials with large audiences
• Team meetings with screen sharing and moderation controls
• Hosting community or academic conferences with multiple sessions

Limitations and Considerations

• No end-to-end encryption: media is decrypted and re-encrypted by the server, so the server must be trusted
• Many-to-many meetings scale quadratically with participant count, so capacity planning matters for larger interactive groups

Galene provides a practical WebRTC conferencing stack that prioritizes deployability, performance, and operational simplicity. It is especially strong for lecture-style sessions and lightweight deployments that still need modern conferencing features.

Apache OpenMeetings is an open source web conferencing and groupware platform for running online meetings in the browser. It combines real-time audio/video, chat, whiteboards, and file collaboration with administrative controls for users, rooms, and organizations.

Key Features

• Web-based audio/video conferencing with multiple modes (audio+video, audio-only, video-only, picture-only)
• WebRTC screen sharing with selectable desktop/window and quality options
• Meeting recording with downloadable MP4 output
• Multi-whiteboard with drawing tools, document import, and save/restore of whiteboards
• In-room file explorer with drag-and-drop, private and room-scoped public storage
• User, group, and room management, including multi-organization support
• Integrated messaging and contacts, plus meeting planning with calendar invites and secure room links
• Polls and voting with result visualization
• Backup and restore of user-generated content

Use Cases

• Hosting internal team meetings with screen sharing and recordings
• Running online classes, workshops, or interviews using whiteboards and document sharing
• Providing a self-managed web meeting solution integrated with existing user/group structures

Limitations and Considerations

• Requires an external media server component (Kurento) for real-time media streaming features

Apache OpenMeetings is a strong fit for organizations that need a full-featured, browser-based conferencing system plus collaboration tools like whiteboards, file management, and scheduling. It is especially useful when you want centralized control over rooms, users, and recorded content.

PlugNMeet is an open-source, self-hosted web conferencing system that provides scalable, customizable WebRTC meetings powered by a Go backend. It combines real-time audio/video, collaboration tools, recording and broadcasting, and an AI meeting agent for transcripts, translations, and summaries.

Key Features

• High-performance Go backend designed for horizontal scaling and low resource usage
• WebRTC-based real-time audio/video with simulcast and dynacast for adaptive streaming and support for H264, VP8, VP9, and AV1 codecs
• AI Meeting Agent offering real-time transcription, live spoken translations, automated summaries and action items, exposed via an artifacts API
• Collaboration tools including a shared whiteboard, collaborative notepad, polls, breakout rooms, and screen sharing
• Recording and broadcasting: reliable MP4 recording, RTMP/RTMPS broadcasting and RTMP/WHIP ingress for production tools
• Flexible integration via APIs and SDKs (JavaScript, PHP) and plugins for popular platforms for white-label deployment
• Deployable with Docker and docker-compose; integrates with LiveKit, Redis, NATS and MariaDB/MySQL for production environments

Use Cases

• Host virtual classrooms and remote learning with recordings, whiteboard collaboration, and LTI integration for LMS
• Run webinars or hybrid events with live broadcasting, MP4 archive recording, and adaptive streaming for diverse network conditions
• Embed white-label video meetings and AI-driven meeting intelligence into SaaS products or corporate intranets

Limitations and Considerations

• Requires a separately deployed LiveKit instance and supporting services (Redis, NATS, MariaDB/MySQL) for full production functionality
• Advanced AI features may need additional service configuration or external models/providers and can increase resource usage
• Office file rendering in the whiteboard relies on optional tools (libreoffice, mupdf-tools) and extra system dependencies

In summary, plugNmeet is a full-featured, extensible web conferencing stack suitable for organizations that need a self-hosted, AI-enabled meeting platform with recording and broadcasting capabilities.

Tigase XMPP Server is a highly optimized, modular XMPP (Jabber) server implemented in Java. It provides core XMPP services for real-time messaging, presence, and federation and is designed for high performance and large-scale deployments.

Key Features

• Implements core XMPP standards and many XEP extensions including stream management, message archiving, message carbons, MUC, publish-subscribe and HTTP file upload
• Supports client connections over TCP, BOSH (HTTP long-polling), and WebSockets, plus server-to-server federation and component connections
• Modular architecture with optional components and connectors for features like MUC, PubSub, STUN, SOCKS5 proxy and database connectors
• HTTP API and high-performance Jetty-based components for integration and management
• Push notifications support and tooling for large-scale, low-latency deployments
• Designed for scalability and optimization with monitoring and management tools available as companion projects

Use Cases

• Powering real-time chat and presence for consumer or enterprise messaging applications
• Implementing group chat, pub/sub systems, and message archiving for collaboration platforms
• Backend for IoT messaging and presence use cases that require XMPP interoperability

Limitations and Considerations

• Advanced configuration and tuning requires familiarity with XMPP concepts and Java-based deployments; many capabilities are provided via optional components rather than a single monolithic UI

Tigase is suited for operators needing a standards-compliant, extensible XMPP server capable of handling large user bases and custom integrations.

Flexisip is a modular, scalable SIP server suite written in modern C++. It provides proxying, presence, conferencing, push gateway and B2BUA capabilities to build VoIP, messaging and real-time communication services.

Key Features

• Modular server suite: proxy, presence server, conference server, B2BUA and an account manager as separate components.
• Push Gateway: routes SIP notifications to mobile platforms and supports RFC 8599 and current APNs/Firebase requirements.
• Media and NAT handling: built-in media relay with ICE/STUN/TURN support and RTP/RTCP/SRTP handling for NAT traversal and secure media.
• Authentication and security: Digest, TLS client certificate authentication, TLS (OpenSSL) and optional OpenID Connect/OAuth support.
• Account management and remote config: REST API (FlexiAPI) for user/account lifecycle, dynamic per-user configuration, and a web admin platform for service management.
• Scalability and deployment: designed for high-availability cluster deployments, load balancing and low-footprint embedded targets (Raspberry Pi, IoT).
• Interoperability: compatible with other SIP systems and extensions; supports SIP transport over UDP/TCP/TLS and various SIP RFC extensions.

Use Cases

• Deploy a hosted SIP service with calling, messaging and conferencing features for mobile and desktop clients.
• Add push-notification support in front of legacy SIP platforms to deliver calls/messages reliably to smartphones.
• Embed a lightweight SIP proxy in edge/IoT devices (intercoms, access control) to add audio/video calling features.

Limitations and Considerations

• PSTN interconnection requires a separate trunk or gateway; Flexisip provides B2BUA functionality but does not supply PSTN lines or virtual numbers.
• Some optional features require additional native dependencies (for example MySQL for account DB clustering, Redis for registrar/cluster communication, XercesC for presence features), so build-time configuration impacts feature set.
• Dual-licensing (AGPLv3 or proprietary) may impose AGPL obligations for some deployments; evaluate licensing before commercial use.

Flexisip combines a full set of SIP server components with mobile-friendly features and small-footprint deployment options. It is targeted at teams building unified communication services, embedded SIP-enabled devices, and operators needing push-enabled SIP gateways.

Prosody IM is a modern XMPP (Jabber) communication server focused on being lightweight, easy to configure, and efficient with system resources. It provides a flexible, modular platform for running private chat infrastructure and extending XMPP with custom functionality.

Key Features

• Standards-based XMPP server supporting a wide range of desktop and mobile XMPP clients
• Modular architecture for enabling features and adding custom extensions via modules
• Federation with other XMPP servers to participate in the open XMPP network
• Designed for low resource usage and straightforward configuration
• Developer-friendly platform for rapidly prototyping new protocol features

Use Cases

• Host private chat for a company, community, family, or homelab
• Run a federated messaging service that connects to other XMPP-compatible servers
• Build custom XMPP functionality (bots, integrations, experimental protocol extensions)

Prosody IM is a strong fit when you want control over messaging data and a mature, open protocol with broad client support. Its Lua-based modular design makes it especially suitable for operators and developers who value simplicity and extensibility.

Wazo Platform is an open-source, API-first project for building carrier-grade IP communication infrastructures. It provides microservices, APIs and SDKs to deliver VoIP, WebRTC, messaging, conferencing, call center and PBX features for custom and scalable deployments.

Key Features

• API-first microservices implemented primarily in Python, exposing REST APIs, WebSockets and Webhooks.
• Call-control and telephony services (wazo-calld) for creating and managing calls, voicemail, transfers and switchboards.
• WebRTC-enabled softphone SDKs and demos for embedding browser-based voice/video clients.
• Engine integration with telecom components (Asterisk, Kamailio, RTPEngine) and a technical stack using Nginx, RabbitMQ and PostgreSQL.
• Container and packaging support (Docker / docker-compose) and OpenAPI-described endpoints for easier integration.

Use Cases

• Build a white-label UCaaS or MSP offering with programmable VoIP, chat and conferencing.
• Integrate an embedded softphone or add telephony features into web and mobile apps.
• Deploy SIP routing, session border controller or contact center/call-center services.

Limitations and Considerations

• Wazo relies on third-party telecom components (Asterisk, Kamailio, RTPEngine); deploying and operating production telecom stacks requires telephony and infrastructure expertise.
• The community maintains most components and some container tooling is marked experimental; CI/packaging and deployment workflows may need adaptation for production.

Wazo Platform provides a modular, extensible foundation for building programmable telephony and UC solutions. It targets operators, MSPs and developers who need deep customization and API-level control over telecommunication features.