Sidekick is a cloud PBX, HD video meetings and team chat in one multi-tenant platform — built on SIP, WebRTC and open standards, with a softphone that runs right in your browser.
The product
Calls, meetings, chat and the contact center share one identity, one presence model and one API — so a ring, a meeting link and a message are all part of the same conversation.
01Calls
Place and answer calls straight from the browser over SIP on WebSockets — no plug-ins, no desk phone required. Hold, transfer and keypad work the way people expect, with live call-quality scores on every call.
02SBC & media
Every call enters through a Kamailio session border controller that authenticates, rate-limits and routes per tenant. rtpengine anchors the media and bridges browser DTLS-SRTP to plain RTP for the PBX, so NAT and encryption are handled before a call reaches your dial plan.
03Meetings
HD video rooms run on a LiveKit SFU with simulcast and an active-speaker spotlight. Turn any phone call into a video meeting in one click, admit guests from a lobby, and let people dial in from a regular phone.
04IVR builder
Wire menus, business hours, HTTP lookups and transfers together on a visual canvas. Every publish is a new version, so you can test, ship and roll back call routing without editing a dial plan.
05Queues & ACD
Route callers to the right agent with longest-idle, round-robin, ring-all or skills-based strategies. Position announcements, wrap-up time and a real-time wallboard keep agents, supervisors and callers in the loop.
06Team chat
Channels, DMs and threads with presence that reflects real call and meeting state, so you know who is free before you ring. Mentions, reactions, read receipts and files arrive instantly over Centrifugo.
07API & webhooks
The REST API is described by an OpenAPI 3.1 spec, errors are RFC 9457 problem details, and organization API keys authenticate with a single header. Subscribe to call, queue, meeting and chat events with signed Standard Webhooks.
Platform
Battle-tested open-source engines do the real-time heavy lifting; typed TypeScript services carry the product logic around them. No proprietary SIP stack, no black boxes.
Asterisk 22 LTS · ARI
Call routing, extensions, ring groups, feature codes, voicemail and call records — driven over ARI by a typed router.
Kamailio
The SIP edge: WebSocket and UDP/TCP/TLS, registrar and auth, NAT handling, scanner bans and per-tenant routing.
rtpengine
Anchors every edge stream, bridges DTLS-SRTP and RTP, and handles prompts, hold music and recordings.
LiveKit SFU
Video rooms with lobby, host controls, recording, phone dial-in and one-click call escalation.
Flow engine on ARI
Runs versioned call-flow graphs: menus, DTMF, schedules, conditions, HTTP lookups and transfers.
Valkey-backed dispatcher
Hold music, position announcements, routing strategies, agent states, wrap-up and live statistics.
Centrifugo
DMs, channels, threads, reactions and presence, plus the bridge that streams events to every browser.
Fastify API · Next.js app
Tenants, roles, provisioning, the public REST API and webhooks — and the app you're looking at.
Underneath:Postgres · Valkey · NATS JetStream · S3-compatible storageContainerised with Docker Compose for Ubuntu
Standards
Sidekick interoperates because it implements the specs, not a proprietary dialect. Here is what ships today.
The same SIP your desk phones speak — over UDP, TCP, TLS and WebSockets from the browser.
Session description and real-time transport.
Browser media is encrypted end to the edge.
Connectivity that survives real networks.
Standards-track browser real-time media.
Wideband voice, scalable video and out-of-band DTMF.
The product surface, held to the same bar.
On the roadmap: TURN (RFC 8656), session timers (RFC 4028), MWI and BLF for desk phones, STIR/SHAKEN, WHIP (RFC 9725), SIPREC (RFC 7866) and SFrame end-to-end encryption (RFC 9605).
Get started
Create a workspace, invite your team, claim extensions and place your first browser call in minutes. Free for 14 days — no credit card, no desk phones required.