aiogotoconnect

Async Python client for GoTo Connect: send SMS via the Messaging v2 API and place outbound text-to-speech voice calls over SIP. Pure Python, asyncio only.

Used by ha-goto, the Home Assistant integration.

Formerly published as gotoconnect (v0.1.0); renamed so the unofficial client does not carry the bare product name. The old PyPI name stays claimed but dormant.

Install

pip install aiogotoconnect

Python 3.12+.

SMS

import asyncio
from aiogotoconnect import GoToOAuth, GoToMessaging

async def main():
    auth = GoToOAuth(client_id="...", client_secret="...", refresh_token="...")
    sms = GoToMessaging(auth.async_get_access_token)
    await sms.async_send_sms("+15551230000", ["+15551234567"], "Hello from aiogotoconnect")
    # GoTo may rotate the refresh token: persist auth.refresh_token after calls.

asyncio.run(main())

GoToMessaging takes any async callable returning a bearer token, so a host app (e.g. Home Assistant) can supply its own.

Voice calls

import asyncio
from aiogotoconnect import GoToVoice

async def main():
    voice = GoToVoice("sip-username", "sip-password")  # server="reg.jiveip.net"
    with open("message.wav", "rb") as f:  # 16-bit PCM WAV
        result = await voice.call_and_play("+15551234567", f.read(), ring_timeout=25, repeat=1)
    print(result)  # CallResult(answered=True, duration=4.2, reason='completed')

asyncio.run(main())

reason is one of completed, remote-hangup, no-answer, busy, rejected. bye_confirmed is False only when we hung up but the server never answered (or refused) our BYE, so the far side may still think the call is up; it is True when no BYE was needed or it was confirmed. Bad SIP credentials raise GoToAuthError.

UDP requests are retransmitted (0.5 s, 1 s, 2 s, ...) until the server responds, and the Record-Route set from the answer is honoured on the ACK and BYE. Cancelling the call_and_play task hangs up (or CANCELs a ringing call) before the cancellation propagates.

Logging

Everything logs to logging.getLogger("aiogotoconnect.sip.call") at DEBUG: SIP start-lines and status codes plus state changes, never headers or credentials. examples/make_call.py --verbose turns it on.

Exceptions

All library errors derive from GoToError:

• GoToAuthError: credentials rejected (OAuth refresh failed, SIP digest refused, SMS 401).
• GoToApiError: a REST call returned a non-success status (.status).
• GoToSipError: SIP registration/signalling failed, including network failures (timeouts, DNS, TLS errors are wrapped, never raised raw).
  • GoToCallError: the call itself failed (e.g. 4xx/5xx to the INVITE, no usable SDP).

Bad arguments raise plain ValueError before any network traffic: an invalid destination number, a WAV that is unreadable, empty or has more than two channels, or an unknown codec.

Create the SIP device in GoTo

1. In GoTo Connect Admin, go to Devices -> Add Device -> SIP Device.
2. Copy the SIP Username, Password, and the device's SIP Server and Outbound Proxy. The registrar is typically reg.jiveip.net (the library's default); the outbound proxy is account-specific (e.g. <account>.jive.rtcfront.net) — pass it as proxy=. Calls proven live against GoTo used both.

Transport

UDP (port 5060) is the default. TLS (port 5061) is supported: GoToVoice(..., transport="tls"). TLS verifies the server certificate with the system CAs; pass ssl_context= to override. An outbound proxy= host, if set, is the connection target while server stays in the SIP URIs.

Live smoke test

GOTO_SIP_USERNAME=... GOTO_SIP_PASSWORD=... python examples/make_call.py --to +15551234567 --wav message.wav --transport tls

Development

python3 -m venv .venv
.venv/bin/pip install -e '.[dev]'
.venv/bin/python -m pytest