Vonage Business Communications

Best Self Hosted Alternatives to Vonage Business Communications

A curated collection of the 9 best self hosted alternatives to Vonage Business Communications.

Cloud-based business communications platform providing VoIP phone service, SMS/MMS messaging, video meetings, team messaging, and contact-center features. Includes PBX/call routing, conferencing, and integrations with CRM and other business systems.

Alternatives List

#1
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
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#2
Jitsi Videobridge

Jitsi Videobridge

An open-source WebRTC SFU that routes media for scalable multiparty video conferencing; supports Colibri XMPP/REST control, DTLS/SRTP, and Prometheus metrics.

Jitsi Videobridge screenshot

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. (jitsi.org)

Key Features

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

Use Cases

  • Powering large-scale multiparty video conferencing (used by Jitsi Meet) where many participants join without per-stream server mixing. (jitsi.github.io)
  • 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. (jitsi.org)

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. (jitsi.org)
  • 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. (forger.sitiv.fr)
  • Performance and capacity depend on network, codec choices, and server sizing; achieving maximum concurrent streams requires tuning (heap, GC, networking) and appropriate deployment topologies. (github.com)

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. (jitsi.org)

3kstars
1kforks
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#3
Asterisk

Asterisk

Open-source PBX and telephony toolkit for building communications applications; modular C-based engine with SIP, WebRTC, RTP, ARI/AMI APIs and hardware support.

Asterisk screenshot

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.

3kstars
1.2kforks
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#4
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
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#5
FusionPBX

FusionPBX

FusionPBX is a domain-based, multi-tenant PBX and softswitch web UI for FreeSWITCH, supporting extensions, IVR, queues, voicemail, provisioning, and more.

FusionPBX screenshot

FusionPBX is a full-featured, domain-based multi-tenant PBX and voice switch built around FreeSWITCH. It provides a web-based interface to manage telephony features for businesses, service providers, and multi-customer environments.

Key Features

  • Domain-based multi-tenancy for hosting multiple independent PBX tenants
  • Extension and user management with typical PBX calling features
  • IVR, ring groups, call parking, and call routing via dialplan tools
  • Call queues / ACD and call center-oriented applications
  • Voicemail features including voicemail-to-email support
  • Device provisioning with troubleshooting logs for provisioning requests
  • Call detail records (CDR) and call recordings management
  • High availability and redundancy options for more resilient deployments

Use Cases

  • Hosting a multi-tenant PBX platform for customers or departments
  • Building a VoIP business phone system with IVR, voicemail, and queues
  • Running a FreeSWITCH-based softswitch with web-managed configuration

Limitations and Considerations

  • Requires deploying and operating FreeSWITCH alongside the FusionPBX web application
  • Some advanced features (for example, certain reporting modules and the REST API) may depend on membership or add-on applications

FusionPBX is well-suited for organizations that want a FreeSWITCH-based telephony stack with a comprehensive administrative UI and multi-tenant capabilities. It scales from single-instance PBX deployments to carrier-oriented environments when designed with redundancy in mind.

963stars
731forks
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#6
Flexisip

Flexisip

Open-source, modular SIP server (proxy, presence, conference, B2BUA) with push gateway, ICE/STUN/TURN, MySQL/SQLite support; optimized for mobile and embedded systems.

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.

174stars
78forks
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#7
Wazo Platform

Wazo Platform

Open-source, API-first platform for carrier-grade IP communications: VoIP, WebRTC, messaging, conferencing and programmable telephony microservices.

Wazo Platform screenshot

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)

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#8
FreePBX

FreePBX

Modular, PHP/JavaScript web GUI that configures and manages Asterisk PBX features, endpoints, and call routing for businesses and service providers.

FreePBX screenshot

FreePBX is an open-source, web-based graphical user interface designed to configure and manage Asterisk telephony servers. It provides a modular platform of built-in features and extensible modules to build IP PBX, UC, and call-centre systems.

Key Features

  • Web-based administrative GUI for configuring Asterisk dialplans, trunks, extensions, IVRs, queues, voicemail and CDR reporting
  • Modular architecture with many open-source modules and an add-on marketplace for commercial extensions (provisioning, call center features, CRM integration)
  • User Control Panel (UCP) for end-user voicemail, call handling, web softphone and customizable widgets
  • Zero-touch phone provisioning and endpoint management for supported IP phones
  • SIP trunking and provisioning integrations, session border controller (SBC) support, and tools for analog/PRI gateway integration
  • REST/API hooks and token-based access for automation and third-party integration
  • Built for common Linux stacks with emphasis on PHP/JavaScript modules and standard LAMP-style components

Use Cases

  • Small-to-medium businesses deploying a full-featured IP PBX with IVR, voicemail, and ring groups
  • Contact centers and help desks using queueing, CDR reporting, and commercial call-center modules
  • Integrators and service providers packaging custom modules, provisioning endpoints, and managing SIP trunking for customers

Limitations and Considerations

  • Core FreePBX is open source but many advanced or enterprise modules are commercial and sold through the add-on marketplace
  • Relies on Asterisk as the telephony engine; feature set and behavior depend on Asterisk versions and underlying Linux distribution

FreePBX is a mature, widely used platform for building customizable telephony systems. Its modular design and large ecosystem make it suitable for many business telecom deployments, while advanced features may require paid modules or vendor support.

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#9
OpenTalk

OpenTalk

OpenTalk is an open-source video conferencing platform offering GDPR-compliant SaaS and self-hosted deployments with features like recording, whiteboards, breakout rooms, polls and telephone dial-in.

OpenTalk screenshot

OpenTalk is an open-source video conferencing platform designed for data protection and digital sovereignty. It provides a full-featured meeting experience for organisations, with SaaS hosting in German data centers and a community/enterprise on‑premises distribution for private operation.

Key Features

  • WebRTC-based real-time audio/video (LiveKit/RTC components used in the stack) with support for screen sharing and staged media controls.
  • Moderator and workshop tools: breakout rooms, subroom audio, talking-stick, prepared polls, audit-proof voting and participant management.
  • Persistent features: meeting recording (open .webm format), storage management, streaming and download of recordings.
  • Collaboration tools: integrated chat, interactive whiteboard, meeting minutes and Etherpad/space-deck integrations for synchronous work.
  • Telephony & SIP support: telephone dial-in and SIP integration for participants without web clients.
  • Authentication and identity: Keycloak / OIDC integration for single sign-on and enterprise identity flows.
  • Componentised, container-first deployment: provided ot-setup templates use Docker and Docker Compose; uses PostgreSQL, RabbitMQ, Redis and S3-compatible object storage for state and media.

Use Cases

  • Public sector and education: GDPR-focused remote teaching, council meetings and secure workshops requiring audit-capable voting and logging.
  • Business collaboration and workshops: interactive workshops with breakout rooms, whiteboards and synchronized notes for cross-team facilitation.
  • Hosted service or private cloud: run as a managed SaaS or deploy via provided setup templates to operate in an organisation's trusted data center.

Limitations and Considerations

  • Default community deployment targets single-server/container setups; large-scale, highly available or Kubernetes-based clusters and advanced enterprise features require the Enterprise offering and additional orchestration.
  • Keycloak is the primary supported authentication provider in current documentation, which may require additional integration work for organisations using other IdP systems.

OpenTalk combines a modern feature set for meetings and workshops with an open-source, privacy-first architecture. It is intended for organisations that need GDPR-aligned hosting options and the ability to operate or extend the platform within their own infrastructure.

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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