Best Self Hosted Alternatives to Cisco Jabber

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.

Tinode is a full-stack instant messaging platform designed for building modern chat applications. It provides a Go-based server, official clients for web and mobile, and APIs that support custom integrations such as chatbots and automated agents.

Key Features

• One-to-one messaging, group chats, and broadcast-style channels with read-only subscribers
• Real-time delivery, read receipts, typing indicators, and presence notifications
• Rich message content: markdown-style formatting, inline media, and file attachments
• Voice and video calls, plus voice messages
• Granular per-topic access control and server-side blocking controls
• Extensible architecture with plugins for features like moderation and chatbots
• JSON over WebSocket (with optional long polling) and Protobuf over gRPC
• Pluggable database layer with support for PostgreSQL, MySQL/MariaDB, and MongoDB
• Media/file handling via local filesystem or S3-compatible storage backends

Use Cases

• Building a custom consumer or community chat app (mobile + web)
• Adding in-app messaging, support chat, or anonymous chat to a product
• Developing bot-driven workflows and integrations using the messaging APIs

Limitations and Considerations

• Federation and end-to-end encryption are listed as planned features and may not be available
• Some advanced capabilities (for example, full-text message search) are planned rather than included by default

Tinode fits teams that want an embeddable messaging backend with strong real-time features and multiple official clients. Its API options and extensibility make it suitable for both standalone chat products and messaging features inside existing applications.

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.

Asterisk is an open-source telephony engine and PBX toolkit implemented primarily in C and developed for GNU/Linux. It exposes traditional PBX features and low-level telephony primitives so developers and operators can build SIP, WebRTC and PSTN-connected communications applications.

Key Features

• Modular, channel-based architecture with pluggable modules for SIP (chan_pjsip / chan_sip), media, codecs and hardware interfaces
• WebRTC support (WSS/DTLS-SRTP), RTP/RTCP handling, and modern codec support including Opus for browser and realtime audio
• ARI (Asterisk REST Interface) exposing REST + WebSocket events for building custom programmable call applications
• AMI and AGI interfaces for management, automation and traditional dialplan scripting; full CLI and menuselect build configuration
• PSTN and telephony hardware integration (traditional telephony cards and drivers) alongside VoIP gateway capability
• Source-driven build system using autoconf/Autotools and GNU Make; extensive documentation, community forum and release advisories

Use Cases

• Deploying an enterprise or branch PBX providing calls, voicemail, conferencing, queues and call routing
• Acting as a VoIP gateway or SBC to bridge SIP/WebRTC clients with PSTN trunks and telephony hardware
• Building programmable communications services (IVR, voicebots, conferencing, call recording) using ARI or AMI

Limitations and Considerations

• Nontrivial operational complexity: requires careful configuration, dependency management and familiarity with telephony concepts
• Requires proactive security and performance tuning (file-descriptor limits, TLS/DTLS configuration); security advisories are periodically published for critical fixes
• Feature surface is large and modularity means some functionality requires enabling/building specific modules or external libraries

Asterisk is a mature, widely adopted telephony engine suited for operators and developers who need deep control over call handling and media. It is maintained by a large community and is intended for production PBX and programmable-telephony deployments.

Openfire is an open source real-time collaboration server that implements the XMPP (Jabber) protocol for instant messaging and presence. It is designed to be straightforward to deploy and administer while supporting scalable, standards-based messaging.

Key Features

• XMPP server for messaging, presence, and roster management
• Web-based administration console for configuration and user management
• Multi-User Chat (MUC) support for group chatrooms
• Extensible plugin architecture to add features and integrations
• TLS support and configurable authentication options for secure deployments

Use Cases

• Self-hosted team chat infrastructure based on open standards (XMPP)
• Embedding XMPP messaging in custom applications and products
• Running private group chat services for communities, schools, or organizations

Openfire is a mature XMPP server with an active ecosystem, making it a solid choice for standards-based messaging deployments that need extensibility and administrative control.

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. (wazo-platform.org)

Key Features

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

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. (wazo-platform.org)
• The community maintains most components and some container tooling is marked experimental; CI/packaging and deployment workflows may need adaptation for production. (github.com)

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. (wazo-platform.org)

Conduit is a lightweight, open-source Matrix homeserver implemented in Rust. It targets easy setup and low system requirements by shipping as a single binary with an embedded database (RocksDB by default) while also supporting a SQLite backend for smaller installations. (conduit.rs)

Key Features

• Single self-contained binary for simple deployment and low overhead.
• Implemented in Rust and built on Ruma and the axum web framework for modularity and performance. (gitlab.com)
• Default RocksDB storage engine with an optional SQLite backend; configurable cache and DB tuning options. (famedly.gitlab.io)
• Docker and systemd packaging examples and deployment guides; configuration via a conduit.toml file (TOML-based). (famedly.gitlab.io)
• Focus on core Matrix features with ongoing improvements and a changelog documenting protocol/version updates and fixes. (conduit.rs)

Use Cases

• Small personal or family Matrix homeserver on low-resource hardware (Raspberry Pi, single-board computers).
• Lightweight team or community chat server that needs easy setup and minimal maintenance overhead.
• Developers and administrators who prefer a single-binary Rust implementation for embedding or custom deployments.

Limitations and Considerations

• Project is marked beta: some Matrix features and federation edge-cases are incomplete (examples historically include certain E2EE federation behaviors and outgoing presence/read-receipt handling). Users should evaluate feature gaps against their needs before production use. (github.com)

Conduit provides a compact, Rust-native alternative to heavier Matrix homeservers, prioritizing simplicity, performance, and low resource usage while continuing to close feature gaps and improve federation behavior.