Metadata-Version: 2.4
Name: dialplane
Version: 0.0.1a0
Summary: A phone line for your Python process. Async SIP/RTP engine for AI voice agents.
Project-URL: Homepage, https://github.com/dialplane/dialplane
Project-URL: Repository, https://github.com/dialplane/dialplane
Project-URL: Issues, https://github.com/dialplane/dialplane/issues
Author: Moha Abdi
License-Expression: Apache-2.0
License-File: LICENSE
Keywords: ai,asyncio,pipecat,rtp,sip,telephony,voice-agents,voip
Classifier: Development Status :: 1 - Planning
Classifier: Framework :: AsyncIO
Classifier: Intended Audience :: Developers
Classifier: Intended Audience :: Telecommunications Industry
Classifier: Operating System :: OS Independent
Classifier: Programming Language :: Python :: 3
Classifier: Programming Language :: Python :: 3.11
Classifier: Programming Language :: Python :: 3.12
Classifier: Programming Language :: Python :: 3.13
Classifier: Topic :: Communications :: Internet Phone
Classifier: Topic :: Communications :: Telephony
Classifier: Topic :: Software Development :: Libraries :: Python Modules
Classifier: Typing :: Typed
Requires-Python: >=3.11
Description-Content-Type: text/markdown

# Dialplane

**A phone line for your Python process.**

Dialplane is an async, pure-Python SIP/RTP engine built for AI voice agents.
Connect any SIP trunk on earth directly to your Python code — answer and place
real phone calls, stream call audio as PCM frames, and plug into agent
frameworks like [Pipecat](https://github.com/pipecat-ai/pipecat) — with no
middleman cloud, no per-minute streaming fees, and no separate media server.

> **Status: pre-release.** This package is a placeholder while the engine
> lands. The first usable release is under active development.

## Why

Today, putting an AI agent on a real phone number means routing audio through
a metered websocket cloud or running a heavyweight media server cluster.
Dialplane's goal: `pip install dialplane`, point it at a SIP trunk, and your
agent is on the phone — self-hosted, wideband audio, audio never leaves your
box.

## Planned

- Async SIP (RFC 3261) + RTP engine, zero required dependencies
- Trunk registration (digest auth) and IP-auth modes
- Inbound answer / outbound dial with an async-first API
- G.711, G.722, and Opus; DTMF (RFC 2833); NAT traversal (STUN, symmetric RTP)
- First-class Pipecat transport; adapters for other frameworks
- Fully-local voice agent examples (faster-whisper + Ollama + Kokoro)

## License

Apache-2.0
