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baresipy

A python wrapper around baresip, the portable SIP user-agent.

Use it to place and receive VoIP calls from python: dial out, answer inbound calls, stream text-to-speech or arbitrary audio into a call, send/receive DTMF, transcribe what a caller says, and build interactive voice bots — with or without a SIP registrar, with or without a sound card.

Install

sudo apt-get install baresip ffmpeg   # Debian/Ubuntu, see docs/setup.md for other distros
pip install baresipy

baresip is the SIP engine baresipy drives via pexpect; ffmpeg is used by pydub for audio conversion. See docs/setup.md for per-distro install commands, SIP account requirements, and troubleshooting.

Optional extras:

extra installs needed for
baresipy[ovos] ovos-plugin-manager, phoonnx, ovos-simple-listener speak()/say() default TTS engine, baresipy.ovos.BareSIPMicrophone, voice bots
baresipy[server] fastapi, uvicorn, python-multipart baresipy-gateway HTTP/WebSocket API, see docs/http-gateway.md
baresipy[test] pytest, pytest-cov, fastapi, httpx, python-multipart, uvicorn running the test suite

Quickstart

Scripted call

Dial a number, speak, hang up:

from baresipy import BareSIP
from time import sleep

b = BareSIP("your_user", "your_password", "your_sip_gateway.example")
b.call("someone@your_sip_gateway.example")

while b.running:
    sleep(0.5)
    if b.call_established:
        b.speak("hello, this is a test")
        b.hang()
        b.quit()
        break

Full runnable version: examples/scripted_call.py.

Answering bot

Subclass BareSIP and override the event handlers:

from baresipy import BareSIP
from time import sleep


class JokeBOT(BareSIP):
    def handle_incoming_call(self, number):
        self.accept_call()

    def handle_call_established(self):
        self.speak("Welcome to the jokes bot")
        self.speak("Goodbye")
        self.hang()


b = JokeBOT("your_user", "your_password", "your_sip_gateway.example")
while b.running:
    sleep(1)

Full runnable version: examples/events.py.

OVOS voice bot

Answer a call and transcribe the caller with an OVOS STT/VAD pipeline, using the incoming call audio as a microphone source:

from baresipy import BareSIP
from baresipy.ovos import BareSIPMicrophone
from ovos_plugin_manager.stt import OVOSSTTFactory
from ovos_plugin_manager.vad import OVOSVADFactory
from ovos_simple_listener import SimpleListener, ListenerCallbacks


class VoiceBot(BareSIP):
    def handle_incoming_call(self, number):
        self.accept_call()

    def handle_call_established(self):
        mic = BareSIPMicrophone(sip=self)
        SimpleListener(mic=mic, wakeword=None,
                        vad=OVOSVADFactory.create(),
                        stt=OVOSSTTFactory.create(),
                        callbacks=ListenerCallbacks()).start()


bot = VoiceBot("your_user", "your_password", "your_sip_gateway.example", record_rx=True)

Full runnable version with two-way TTS replies: examples/voice_bot.py. See docs/ovos-integration.md for the complete walkthrough.

Examples

Example Demonstrates
examples/scripted_call.py dial out, speak, send DTMF, hang up
examples/events.py subclassing BareSIP to auto-answer and speak
examples/voice_bot.py two-way OVOS STT/TTS voice bot
examples/voicemail.py voicemail / answering machine, with optional transcription
examples/call_transfer.py receptionist-style call transfer
examples/outbound_campaign.py sequential outbound dialing campaign
examples/conversational_agent.py barge-in voice agent with a pluggable response function
examples/multi_line.py concurrent multi-line SIP in one process
examples/secure_trunk.py registered account over TLS transport + SRTP media
examples/gateway_client.py driving baresipy-gateway over HTTP/WebSocket
examples/contact_list.py local JSON contact store
examples/ivr_menu.py IVR menu (DTMF-driven call routing) - not yet added, see docs/call-control.md for the DTMF primitives it will use

See docs/call-control.md for transfer, DTMF, barge-in, and call-metadata details used across these examples.

Features

Feature How
Registered SIP account calls BareSIP(user, pwd, gateway)
Registrar-less direct SIP calls BareSIP() + call("sip:user@ip:5060"), see docs/direct-calls.md
Headless operation (no sound card) BareSIP(headless=True)
Text-to-speech into a call speak() via any OPM TTS plugin
Arbitrary audio into a call send_audio(path)
DTMF (send/receive) send_dtmf(digits, mode="keys"), handle_dtmf_received()
Recording inbound call audio record_rx=True, get_rx_wav() / get_rx_stream()
OVOS microphone plugin baresipy.ovos.BareSIPMicrophone
Local contact list baresipy.contacts.ContactList
Event-driven call handling override handle_* methods

Documentation

Credits

This work has been sponsored by Matt Keys, eZuce Inc

License

MIT

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