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.

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.

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.

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.

For online classes, telehealth visits, and support calls, OpenVidu Meet is a video meeting app that anyone with a link can join from a browser, with no account or download. A room needs only a name, and an optional wizard adds limits on participants (up to 30) and meeting length (up to a day), plus microphone and camera defaults. A shared Moderator or Speaker link admits anonymous guests, a personal link identifies each invited guest, and registered users sign in with their accounts. Developers can embed a room with an iframe or web component, and webhooks report meeting events to as many endpoints as needed. A smart mosaic favors recent speakers and counts the participants it hides, and a status rail shows the recording light, an encryption badge, and the time left. Moderators mute one participant's microphone, camera, or screen share, switch them off for everyone, remove people, and promote others. Virtual backgrounds blur the room or swap in an image, before joining or mid-call. Recordings start by hand or when the first participant, the second participant, or a moderator joins, in grid, speaker, or single-speaker layout. Permitted members can play, download, delete, or share saved recordings by link. End-to-end encryption protects audio, video, chat, and names behind a shared passphrase, though encrypted rooms cannot be recorded. 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.

Viewers arrive on browsers, phones, and desktop players, and SRS meets each of them with the right protocol from a single published stream. A broadcaster pushing RTMP can reach browser viewers through WebRTC with sub-second latency, phone apps through HLS, and desktop players through HTTP-FLV, all from the same live source without running parallel encoders. WHIP endpoints accept WebRTC ingest from browser studios and tools like OBS, while WHEP serves WebRTC playback, so an entire publish-to-viewer path can stay on modern web protocols. Operators shape each channel with virtual hosts, authenticate publishers and players through HTTP hooks that consult your own backend, and record live streams to disk as DVR archives for replay. Forwarding republishes a stream to YouTube, Facebook, or another SRS instance, and origin clustering lets edge servers pull from a central origin so one ingest feeds viewers in several regions. The built-in HTTP API reports streams, clients, and server health for monitoring, and GB28181 support ingests surveillance camera networks alongside creator broadcasts. Configuration lives in a single reloadable file, so changing a vhost or enabling HLS rarely means restarting the listeners your audience is attached to. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. MIT licensed.

MediaMTX sits between cameras and viewers as a self-hosted live media server, taking in video and audio streams over one protocol and handing them back out over another, so one IP camera feed can end up playing in a browser, a phone app, and a recording archive simultaneously. Publish with FFmpeg, GStreamer, OBS Studio, a browser, or a Raspberry Pi camera over RTSP, RTMP, SRT, WebRTC, Media-over-QUIC, HLS, MPEG-TS, or RTP, and read it back on any other protocol without a transcoding pipeline. Run dozens of camera feeds side by side, each on its own path, and keep every path available even while its publisher is offline. Record everything to disk in fMP4 or MPEG-TS as it streams, then play recordings back later through the same server instead of shipping files around. Authenticate publishers and viewers with built-in users, an external HTTP check, or JWT, and use the control API plus Prometheus metrics to wire the server into the rest of your monitoring stack. Forward streams to other servers, proxy cameras behind NAT, and fire hook commands whenever a client connects, publishes, or disconnects, so a doorbell camera can trigger a snapshot or an alert as it starts streaming. Reload the configuration while clients stay connected, and tune everything from the same mediamtx.yml file or environment variables, with no database or runtime dependencies to babysit. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. MIT licensed.

"Run the migration and call me if anything fails" is an instruction your agent can finally obey, because Agentring is the self-hosted bridge that lets AI agents ring your phone and talk it through with you. When an agent needs a decision, hits a wall, or finishes a long job, your phone rings even locked and in Doze; a realtime voice AI explains why it is calling, answers your questions, and hands the transcript back so the agent carries on with your decision. You can also call your agents back from the Android app and ask what CI said while driving; the voice AI can consult your other agents mid-call, and a slow answer arrives when ready, or as a message if you have hung up. A missed call turns into a message you can answer from the notification. Anything that speaks MCP can reach you, including Claude Code, Codex, opencode, OpenClaw, and Hermes, and scripts or n8n reach you through the REST API and CLI. Pick your voice: Gemini Live by default, OpenAI Realtime, Grok Voice, Phonic, or your own local speech server with no key at all. The relay and LiveKit run in one Docker Compose behind Caddy, the daemon sits beside your agents and connects outbound only, and FCM push wakes the locked phone. No phone numbers, no Twilio, no per-minute fees. 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.

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.

VoicEra lets you run your own voice AI phone platform, where callers talk to agents you configure and every recording and transcript stays on infrastructure you control. Build telephony agents that answer inbound calls or run outbound campaigns, choose the speech recognition, language model, and speech synthesis behind each one, and swap providers without rebuilding the platform. Connect phone numbers through telephony carriers, route live audio through the real time runtime, and let workers execute campaign calls in the background while you watch progress from the operator dashboard. Review call logs with recordings and transcripts after each conversation, including provider recordings when carriers supply them. Serve callers in Indian and low resource languages by combining Bhashini, AI4Bharat, and other Indic models, switch languages mid call, or run speech and language models locally through the optional model server gateway. Add new AI or telephony providers behind stable registry interfaces when a better model or carrier appears, authenticate operators, and keep media in your own object storage and database. Test agents in the browser before any phone number is involved, then scale the same configuration to real calls. 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.

Once a voice agent is imported or built, Lampline connects it to a phone line, sends an AI caller to grade real calls, and keeps it all on your own server. Start by importing a Dialogflow CX or CX Agent Studio export along with a report of anything to check, or build fresh in a drag-and-drop builder that mixes LLM steps with scripted pages for exact lines and form fields. Next, pick one of 20 model providers, including Claude, OpenAI, Gemini, Bedrock, Groq, and DeepSeek, or run Ollama with local speech and no API keys. Test by chat or microphone while each step lights up, and let agents call HTTP and MCP tools and save records in collections of their own. Then connect callers through Twilio, Telnyx, or any SIP trunk, PBX, or session border controller via jambonz, with transfers to a person that carry the call's details as SIP headers. Finally, evals phone the agent over a real line, where an AI Test Caller brings background noise, accents, and interruptions, grades each transcript check by check, and reports typical and p95 response times as callers hear them. Each agent stays a folder of YAML and Markdown that you can review in git and lint or test with the vk CLI. 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.