Introduction

SIP.IO is an edge-native programmable-voice (CPaaS) and cloud-contact-center (CCaaS) platform. It gives developers the building blocks to create voice products (inbound and outbound calling, IVRs, call queues, business-hours routing, voicemail, conferences) and to run a full contact center on top of them.

If you’ve used Twilio Programmable Voice or Amazon Connect, the concepts will feel familiar: numbers, call flows, queues, agents, hours of operation. What’s different is the foundation.

The angle: edge-native, developer-first

Traditional voice platforms put their logic inside heavyweight, regional media servers. SIP.IO inverts that:

• The brain is an edge runtime. It owns authentication, call routing, the call-flow engine, ACD/presence, and concurrency control, and it runs at the edge, close to your callers, everywhere.
• The edge is a thin SIP node (the SIP signaling layer + the media relay + the media engine). It is deliberately “dumb and replaceable”: it asks the brain for every decision and executes pushed commands. Scale it horizontally; keep one global control plane.

The result is a platform that is global by default, low-latency, and fast to evolve: most product changes ship by deploying an edge runtime, with zero SBC restarts.

It’s also built for reach and scale: regional hubs anchor signaling and media close to your users, DID numbers are available internationally through carrier partnerships, and the edge is elastic by design (stateless nodes you scale horizontally to absorb high-concurrency, high-volume traffic) while one global brain stays authoritative.

Design philosophy

Keep the SIP edge dumb and replaceable; put the logic in the edge runtime. The dividing line for whether the platform inserts a media server at all is one question: does the platform need to answer the call itself? If not, the call is simply proxied and media flows through a kernel relay.

What you can build

Area	What it gives you
Programmable Voice	A visual, versioned call-flow engine: IVR menus, digit collection, dial, ring groups, queues, voicemail, conference, conditions, HTTP requests, and time conditions.
Numbers & Trunking	DIDs and extensions with typed route targets, per-number concurrency caps, outbound routes and trunks, and regex number/caller-ID transforms.
Contact Center (ACD)	Skills/tier-based queues, agents & presence, position announcements, music/message on hold, and overflow routing.
Business Hours	Reusable, timezone-aware schedules with holidays, attachable to numbers, extensions, queues, and ring groups.
Concurrency Control	Per-account/user/DID channel ceilings plus sliding-window rate rules.
Multi-tenant / white-label	A reseller hierarchy with per-account branding, language, and timezone defaults.

Who it’s for

SIP.IO is built for developers: teams shipping voice features into their products, BPOs and contact centers that want programmable routing, and resellers/white-label operators who need multi-tenancy out of the box.

How the docs are organized

• Concepts: how the platform works, the edge-native architecture, the control plane, the media edge, the data model, multi-tenancy, and observability.
• Programmable Voice: call flows, the flow node reference, numbers, outbound, media & TTS.
• Contact Center: queues/ACD and everything around them.
• Routing & Controls: business hours and concurrency.
• Guides: short, copy-pasteable, task-oriented how-tos.
• API Reference: the control-plane endpoints and the resource model.

Note

SIP.IO is under active development. Throughout the docs, anything not yet shipped is clearly flagged as Designed or Roadmap. The public /v1 REST API is on the roadmap; today the platform is driven by the control-plane endpoints and the resource model described in the API Reference.

Ready? Head to the Quickstart.