Routr is a programmable SIP server that proxies, registers, and routes calls for carriers and communication platforms, replacing racks of inherited PBX plumbing with one cloud-ready process. Phones and trunks connect over UDP, TCP, TLS, WebSocket, or secure WebSocket, register against the built-in registrar, and find each other through the location service backed by memory, Redis, or Postgres. Routing is configurable per domain with intra-domain, ingress, egress, and peer-egress strategies, and multi-tenant domains carry their own access control lists. Load balancing with session affinity spreads calls across Asterisk and FreeSWITCH media servers, and RTPEngine middleware anchors media and adds SDES-SRTP toward TLS trunks. Web phones authenticate with JWT endpoint credentials, and every call message can be recorded and replayed for debugging. Operators manage the whole system through a gRPC API, the Node.js SDK, or the rctl command-line tool, and custom Processors and Middleware written in JavaScript reshape SIP messages as they pass. OpenTelemetry traces, metrics, and logs expose the server's behaviour in production, and the same deployment runs from a single Docker container to a Kubernetes cluster through the published Helm charts. There is no single point of failure in the design, so a registrar outage never takes the dial tone with it. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. MIT licensed.

One server replaces the phone system, the meeting link, and the fax line: AiPBX turns a Linux machine into a business phone system with extensions, video meetings, chat, and call queues. Every employee gets one extension that rings a desk phone, the browser softphone, and the mobile app, while ring groups, multi-level IVRs, and time conditions route callers without anyone editing dialplan files. Queues staff your support line with dynamic agent sign-in, hold music, and position announcements, and a live wallboard lets supervisors watch waiting callers and agent states, then listen, whisper, or barge into a call. Meetings run in the browser: schedule conferences with PINs, send guests a magic link into a WebRTC room with screen sharing, and let streams forward without transcoding. Chat lives in the same portal: photos, files, presence, read receipts, and fax is composed in a rich-text editor or uploaded as PDF and delivered over T.38. Desk phones provision themselves: a discovery daemon spots new Yealink, Fanvil, Grandstream, Snom, and Cisco handsets and writes their configuration files. An AI assistant plans dialplan changes in plain language with dry-run preview and one-click undo, cloud voices from Google, Polly, Azure, ElevenLabs, or OpenAI read announcements aloud, and GDPR, KVKK, HIPAA, and PCI-DSS presets govern recording consent and retention purges. A 100-plus-endpoint REST API with scoped keys and HMAC-signed webhooks makes everything scriptable. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. MIT licensed.

MikoPBX gives a small business a complete phone system on a server it controls, with extensions, call queues, IVR menus, and call recording instead of per-seat hosted PBX fees. Follow a call through it: incoming rules per provider match exact numbers or patterns, fall through by priority when nobody answers, and end at a default route. Each employee gets an extension with SIP credentials, forwarding for no-answer, busy, and unavailable states, and an optional recording switch. Queues offer seven distribution strategies, including Ring All, Least Recent, Fewest Calls, and Memory Hunt, and can announce each caller's position and estimated wait. IVR menus play a recorded greeting, route keypad choices to departments or extensions, and fall back to a default number after a set number of retries. Call history filters by caller, destination, or DID, flags missed calls in red, and plays or downloads recordings as WebM, MP3, WAV, or OGG. Voicemail, call transfer, call pickup, and a holiday and night-time switcher serve daily needs, while free and paid marketplace modules, among them a softphone, auto dialer, and LDAP/AD sync, handle the rest. A firewall, Fail2Ban, passkey sign-in, and CrowdSec integration protect the system, and a REST API with more than 250 endpoints lets other tools manage it. The interface speaks 26 languages. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. GPL-3.0 licensed.

FreeSBC sits at the edge of your phone system so your FreeSWITCH or Asterisk never has to face the public internet. It is the only element with a public address: phones register through it over SIP on UDP, TCP, or TLS, browsers call through WebSocket with WebRTC terminated at the edge (ICE-Lite, DTLS-SRTP, and RTCP-mux, relayed onward as plain RTP), and carriers reach your switch through it as an outbound proxy, with the switch's private addresses hidden from every SDP body, Contact, and Request-URI it constructs. The switch stays on a private LAN as the registrar and call brain, holding credentials and carrier accounts, while FreeSBC anchors all media on its own ports with hardened first-packet latching and a silence watchdog, and never transcodes. Against the background radiation of SIP scanning, it applies per-IP rate limits with separate limits for carrier sources, scanner fingerprinting with in-memory bans, and admission control for public INVITEs and REGISTER enumeration, and every denial is a silent drop. A multi-switch pool hashes users across switches, and one fixed private socket keeps the trusted side simple. The embedded WebUI, REST API, and Prometheus metrics sit behind a bcrypt-checked admin login, showing live calls and media ports. One static binary, one YAML file, no database, no Redis, and no separate media process to babysit. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. Apache-2.0 licensed.

With over 20,000 GitHub stars and adoption by companies building everything from telehealth platforms to AI voice agents, LiveKit is the most widely deployed open-source real-time communication server available. The Go-based Selective Forwarding Unit handles hundreds of concurrent participants per node with adaptive bitrate streaming, simulcast layers, SVC codec support for VP9 and AV1, and end-to-end encryption. Client SDKs span JavaScript, Swift, Kotlin, Flutter, React Native, Rust, Python, Unity, and ESP32 embedded devices, while server-side APIs cover Node.js, Go, Ruby, Java, Python, Rust, PHP, and .NET. The Agents framework enables building AI-powered voice and video applications — real-time speech-to-text, LLM-driven conversations, and computer vision pipelines — running as server-side participants in any room. Egress records sessions to S3-compatible storage or streams to RTMP endpoints, while Ingress pulls external feeds from OBS via RTMP, WHIP, or SRT into LiveKit rooms. The SIP bridge connects traditional telephony to WebRTC rooms for hybrid conferencing. JWT-based authentication, webhook notifications, room-level moderation APIs, and selective subscription give operators granular control. Deploy as a single binary for development, Docker Compose for production single-node, or Kubernetes with the official Helm chart for distributed multi-region clusters using Redis for state coordination. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. Apache 2.0 licensed.

One Janus installation can host a video meeting in one session, stream a camera feed in another, and bridge a browser call into a telephone network in a third, because Janus is a general-purpose WebRTC server whose work is done by interchangeable plugins. The VideoRoom plugin publishes each participant's streams to everyone else in the room with simulcast and viewport selection, so a class or team call adapts quality per viewer instead of forcing one bitrate on all. Streaming mounts live or prerecorded sources, including RTSP cameras and RTP feeds, and hands them to any number of WebRTC viewers, while AudioBridge mixes voices into conference rooms with recording and signed-token access. SIP and NoSIP plugins register with existing VoIP infrastructure and place or receive calls, translating between WebRTC media and telephone audio so a web page can ring a desk phone. Record&Play captures publisher streams to files you can replay through the same server, TextRoom adds data-channel chat rooms, and Lua or JavaScript plugins let you script custom media logic without recompiling the core. Applications talk to Janus over HTTP or WebSocket APIs, with RabbitMQ, MQTT, and NATS transports for bus-driven deployments, and event handlers report session activity to your monitoring stack. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. GPL-3.0 licensed.

Voice chat for a gaming community, a podcast crew, or a distributed team stops depending on someone else's platform once you run your own Mumble server. The Murmur server hosts channels where hundreds of users can talk at once with latency low enough for competitive play, and every conversation is encrypted in transit with certificate-based authentication for identities. Administrators shape access with a detailed permission system: channels can be public, password-gated, or arranged so two groups stay separate while their leaders can speak to both. Users get positional audio in supported games, an in-game overlay showing who is talking, a multi-channel recorder for podcast production, and a public server browser for finding communities. The server side is built for automation: the Ice middleware interface exposes user and channel management to scripts, authenticator modules check logins against existing account databases, and community web interfaces and channel viewers plug into the same API. Running the server is light on resources, and regulars can register their identity once and be recognized across visits without passwords. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. BSD-3-Clause licensed.

MiroTalk P2P gives your team browser meeting rooms with video, screen sharing, and chat, where the media travels directly between participants instead of through a rented cloud. Create a room, send the link, and guests join from any modern browser with nothing to install and no account to create. Calls carry video at high frame rates, and anyone can share a full screen, one window, or a browser tab to walk through documents and slides. The built-in recording captures screen, camera, and audio in the browser, ready to download and pass along after the meeting. Chat supports Markdown formatting and emoji, and the room also packs a collaborative whiteboard, a shared text editor, live polls, and breakout rooms. Files move between participants inside the call, and speech recognition plus voice commands keep hands free during long sessions. Hosts lock rooms with passwords, hold arrivals in a lobby, and require approval before strangers wander in. Sign-in through OIDC, per-room JWT credentials, and host protection controls keep recurring meetings behind your own identity provider. A full REST API creates rooms from your own tools, Slack and Mattermost commands start calls from chat, and an iframe embed drops meetings into existing pages. The interface speaks 133 languages, and an optional ChatGPT assistant joins rooms to answer questions in the conversation. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. AGPL-3.0 licensed.

Backed by nearly 30,000 GitHub stars and powering millions of meetings through meet.jit.si, Jitsi Meet is the open-source video conferencing platform that runs entirely in the browser with no plugins or downloads required. The WebRTC-based architecture delivers HD audio and video through the Jitsi Videobridge selective forwarding unit, while Prosody handles XMPP signaling and Jicofo manages conference focus and orechestration. End-to-end encryption uses AES-GCM 128-bit keys exchanged via the Olm protocol, protecting audio, video, and screen sharing through the Insertable Streams API in Chromium-based browsers. Moderators can split participants into breakout rooms with individual password protection, enable lobby waiting rooms for admission control, set meeting passwords, and track speaker statistics with per-participant talk time metrics. Virtual backgrounds leverage TensorFlow.js body segmentation with WebGPU and WebGL acceleration, automatically adapting quality through device tier detection. Server-side recording and YouTube live streaming run through Jibri, while Jigasi provides a SIP gateway for PSTN dial-in and speech-to-text transcription. The platform includes in-meeting chat with private conversations, polls, raise-hand functionality, emoji reactions, and tile and speaker view layouts. Mobile applications for iOS and Android complement the web client, and both web and native SDKs enable embedding video conferences into custom applications. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. Apache-2.0 licensed.

Openfire is a self-hosted XMPP server that gives your organization its own instant messaging network, with one-to-one chat, persistent group rooms, and federation to other XMPP servers, all administered from a web console. Team members connect with any XMPP client on desktop, mobile, or the web and get presence, rosters, offline message delivery, and file sharing inside conversations. Multi-user chat rooms stay available around the clock with configurable history, moderation roles, and access controls, so a support channel or project room keeps its context instead of dissolving into scattered direct messages. The plugin catalog installs from inside the admin console and adds capabilities like message archiving and compliance monitoring, a browser-based chat client, HTTP file upload, and push notifications for mobile clients. An LDAP directory or an existing database can supply users and groups, so accounts stay in sync with the directory your organization already maintains. Server-to-server federation connects your users with contacts on other XMPP domains over encrypted connections, and load balancing spreads busy deployments across multiple instances. TLS certificates, encrypted client ports, and IP-based access lists for the admin console keep the deployment under your control from the first setup wizard onward. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. Apache-2.0 licensed.

Running chat for a dozen people is easy; running it for a city-sized crowd, on servers you control, with group rooms, archived history, and phones that still buzz when the app is closed, is the job MongooseIM was built for. MongooseIM is an XMPP server at the core of a messaging platform, so any standard XMPP client on desktop, mobile, or web can connect, and browser clients reach it over WebSockets or BOSH. Group conversations run in multi-user chat rooms, message archive management keeps searchable history on the server, and message carbons deliver real-time copies to every signed-in device a person owns. Offline messages wait for reconnecting clients, rosters carry contact lists and presence, and privacy settings plus vCard profiles round out the everyday chat features. Operators manage users, rosters, messages, and settings through a GraphQL API over HTTP or the mongooseimctl command line, and more than fifty built-in metrics show what is happening inside each node. Accounts authenticate against internal storage, MySQL, PostgreSQL, Cassandra, LDAP, or external scripts, and multi-tenancy support creates thousands of chat domains dynamically on one deployment. Adding cluster nodes raises capacity as the audience grows, and the companion push component wakes mobile apps through push notifications. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. GPL-2.0 licensed.

A single ejabberd node carries chat, device telemetry, voice calls, and federated messaging at once, because the same Erlang server speaks XMPP, MQTT, SIP, and Matrix over one authentication, storage, and clustering layer. Users get standards-based instant messaging with group chats, message archives, publish-subscribe, message carbons, push notifications, and OMEMO encryption, and your server federates with the global XMPP network out of the box. Devices publish and subscribe through the built-in MQTT broker on TCP or WebSocket transports, sharing accounts and access rules with your chat users instead of needing a second broker. Voice and video calls route through the integrated SIP gateway and STUN/TURN support, so WebRTC clients get signalling and media relay from the same deployment. Matrix support bridges federated rooms into the same infrastructure, letting two decentralised ecosystems interoperate without a separate bridge stack to babysit. Clustering with consistent hashing scales a deployment across nodes, and a single node is rated at two million concurrent sessions for very large communities. Operators manage everything from the web admin and command line, with SQL or Mnesia storage, LDAP authentication, and virtual hosting for multiple domains on one server. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. GPL-2.0 licensed.

Why rent your messaging from a corporate walled garden when Tinode runs WhatsApp-style chats, groups, channels, and calls on a server you control? Teams get one-on-one and group chats that stay in sync across web, Android, and iOS, with typing indicators, delivery and read receipts, presence, replies, forwarding, editing, pinned messages, and notes to self. Communities broadcast through read-only channels with an unlimited number of subscribers, and granular permissions decide who can read, write, or invite on each topic. Support desks welcome anonymous visitors, mark staff and verified accounts with badges, and put chatbots to work with forms and templated replies. Messages take markdown-style formatting, inline images and videos, file attachments, and voice messages. One-to-one voice and video calls run peer to peer over WebRTC, using the TURN and STUN servers you configure. Administrators block unwanted contacts on the server, attach moderation plugins over gRPC, and script the rest with a command-line tool. Attachments can sit on local disk or Amazon S3, the database can be MySQL, PostgreSQL, or MongoDB, and sharded clustering with failover scales past one server. The clients offer light, dark, and system themes, custom chat wallpapers, and translations for more than a dozen languages. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. GPL-3.0 licensed.

Build a voice AI agent that answers calls, qualifies leads, books appointments, and transfers to a human when needed, all from a drag-and-drop workflow builder in your browser. Dograh ships as a Docker Compose stack (API, web UI, Postgres, Redis, MinIO) that you self-host on any Linux server with automatic HTTPS provisioning via Let's Encrypt. The visual workflow builder lets you design multi-turn conversation flows by connecting nodes for greetings, intent classification, tool calls, and handoffs; describe your use case in plain English and the platform generates the LLM prompt and node graph for you. Connect your own speech-to-text, LLM, and text-to-speech providers (OpenAI, Anthropic, Gemini, ElevenLabs, Deepgram, local Whisper, Kokoro, or any OpenAI-compatible endpoint) or use the built-in Speech-to-Speech mode with Gemini Flash Live and GPT-Realtime-2 for sub-200ms latency. Telephony plugs in through Twilio, Vonage, Vobiz, or Cloudonix for inbound and outbound calling, with live agent transfer when the conversation needs a human. Webhook tool calls connect to Salesforce, HubSpot, Google Calendar, Cal.com, or any REST API without writing orchestration code. The ClonedVoice feature mixes real human voice recordings for high-frequency phrases with neural TTS fallback for dynamic content, cutting costs while improving caller trust. A built-in MCP server lets AI coding agents like Claude Code or Cursor design, test, and edit workflows through natural language. Deploy on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. BSD 2-Clause licensed.

Your front desk phone rings at midnight and nobody sane is awake; QuickVoice answers it with an AI agent you configured, on servers you own, with call recordings you control. Build agents in the web console: write the prompt, pick speech-to-text, language model, and voice, set the first message, then link a real phone number from Twilio or Telnyx through LiveKit SIP trunks. Inbound callers get a receptionist that answers questions from your uploaded knowledge base, PDFs and web pages chunked and embedded for live retrieval, while outbound mode fires single quick calls or batch campaigns for reminders and follow-ups. Agents can invoke allowlisted MCP tools mid-call, so checking an order status or booking a slot happens inside the conversation instead of after it. Every call lands in the log with transcript, recording, and metadata, searchable from the analytics dashboard, and privacy controls redact personal data by default, can suppress recordings entirely, and sweep old transcripts on a retention schedule. Multi-language voices, per-agent runtime overrides for campaigns, organization accounts with API keys and audit logs, and Stripe billing hooks round out a platform built for teams who would rather inspect their voice stack than rent a black box. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. MIT licensed.