rfcvoip 2.8.5

PyVoIP is a pure python VoIP/SIP/RTP library.

pyVoIP

PyVoIP is a pure python VoIP/SIP/RTP library. Currently, it supports PCMA, PCMU, PCMA-WB, PCMU-WB, telephone-event, optional Opus when libopus is available, and optional SILK when a pysilk backend is available.

This library does not depend on a sound library, i.e. you can use any sound library that can handle linear sound data i.e. pyaudio or even wave. The public PyVoIP audio format is unsigned 8-bit mono with a configurable sample rate. By default, PyVoIP uses the negotiated codec's preferred public sample rate; pass audio_sample_rate=8000 to keep the legacy 8 kHz behavior. Keep in mind PCMU/PCMA are still 8000Hz RTP codecs on the wire.

Opus support is optional. PyVoIP first reuses the libopus handle already loaded, then falls back to common system libopus names. If libopus is not available, Opus is reported as unavailable and is not included in SIP offers. The public PyVoIP audio read/write format remains unsigned 8-bit mono. When audio_sample_rate is omitted, Opus uses a 48000Hz public sample rate and converts internally as needed.

SILK support is optional. Install pyVoIP[silk] or install silk-python separately to provide the pysilk backend. When available, PyVoIP advertises SILK as dynamic RTP payloads for 24000Hz, 16000Hz, 12000Hz, and 8000Hz. The public PyVoIP audio read/write format remains unsigned 8-bit mono. When audio_sample_rate is omitted, SILK uses the selected SILK RTP clock as the public sample rate and converts internally as needed.

PCMA-WB and PCMU-WB are implemented as RFC 5391 / G.711.1 R1 core-layer payloads. PyVoIP advertises mode-set=1, uses a 16000Hz RTP clock, and keeps the public read/write format as unsigned 8-bit mono with a configurable sample rate. When audio_sample_rate is omitted, these codecs use a 16000Hz public sample rate. Incoming G.711.1 packets in wider modes are decoded from their G.711-compatible L0 core layer.

Getting Started

Simply run pip install pyVoIP, or if installing from source:

git clone https://github.com/tayler6000/pyVoIP.git
cd pyVoIP
pip install .

Don't forget to check out the documentation!

Basic Example

This basic code will simple make a phone that will automatically answer then hang up.

from pyVoIP.VoIP import VoIPPhone, InvalidStateError


def answer(call):  # This will be your callback function for when you receive a phone call.
    try:
        call.answer()
        call.hangup()
    except InvalidStateError:
        pass


if __name__ == "__main__":
    phone = VoIPPhone(
        <SIP Server IP>,
        <SIP Server Port>,
        <SIP Server Username>,
        <SIP Server Password>,
        callCallback=answer,
        myIP=<Your computer's local IP>,
        sipPort=<Port to use for SIP (int, default 5060)>,
        rtpPortLow=<low end of the RTP Port Range>,
        rtpPortHigh=<high end of the RTP Port Range>,
        audio_sample_rate=None,  # None = codec-preferred; 8000 = legacy behavior
    )
    phone.start()
    input("Press enter to disable the phone")
    phone.stop()

Using an outbound SIP proxy

If your provider requires a separate outbound proxy, keep server pointed at the SIP domain / registrar and pass the proxy host separately.

phone = VoIPPhone(
    "sip.example.com",
    5060,
    "alice",
    "secret",
    myIP="192.0.2.10",
    proxy="pbx.example.net",
    proxyPort=5060,
    auth_username="alice-auth-id",  # Optional: used for Proxy-Authorization
    callCallback=answer,
)

proxy may be a hostname, host:port, or a SIP URI such as sip:pbx.example.net:5060.

TCP/TLS SIP transport with RFC 3263-compliant resolution

# TCP, explicit URI transport wins.
phone = VoIPPhone(
    "sip:registrar.example.com;transport=tcp",
    None,
    "alice",
    "secret",
    myIP="192.0.2.10",
)

# TLS via sips: and RFC 3263 DNS if no explicit port is present.
phone = VoIPPhone(
    "sips:registrar.example.com",
    None,
    "alice",
    "secret",
    myIP="192.0.2.10",
    tls_server_name="registrar.example.com",
)

# Legacy host + explicit port stays simple unless transport is supplied.
phone = VoIPPhone(
    "registrar.example.com",
    5060,
    "alice",
    "secret",
    myIP="192.0.2.10",
    transport="tcp",
)

Inspecting and prioritising supported codecs

Parsed SIP/SDP messages and active calls can report the codecs offered by the remote endpoint and the codecs supported by PyVoIP.

import pyVoIP


def answer(call):
    report = call.codec_support_report()

    print("Remote codecs:", report["remote"])
    print("PyVoIP codecs:", report["pyvoip"])
    print("Compatible codecs:", report["compatible"])
    print("Unsupported remote codecs:", report["unsupported"])


# Also available for any parsed SIPMessage containing SDP:
remote_codecs = sip_message.supported_codecs()
report = sip_message.codec_support_report()

# And at module level:
pyvoip_codecs = pyVoIP.supported_codecs()

# Include optional codecs that are currently unavailable, useful for frontend UI:
codec_status = pyVoIP.codec_availability()
all_known_codecs = pyVoIP.supported_codecs(include_unavailable=True)

# Codec priority scores control local SDP offer order and negotiated codec
# selection. Higher scores are preferred. Defaults prefer Opus, then SILK,
# then G.711.1 wideband core codecs, then narrowband G.711.
print(pyVoIP.codec_priorities())
pyVoIP.set_codec_priority(pyVoIP.PayloadType.PCMA, 950)
pyVoIP.set_codec_priority(pyVoIP.PayloadType.PCMU, 900)
pyVoIP.reset_codec_priorities()

If Python loads libopus after PyVoIP has already been imported, refresh the codec registry before creating or placing calls:

import pyVoIP
pyVoIP.refresh_supported_codecs()
print(pyVoIP.codec_availability())

Pre-call codec inspection

Local codec information is available without an active SIP session. Remote codec information can be probed before starting a call with SIP OPTIONS when the peer/provider returns SDP in its OPTIONS response.

from pyVoIP.VoIP import VoIPPhone


phone = VoIPPhone(
    "sip.example.com",
    5060,
    "alice",
    "secret",
    myIP="192.0.2.10",
)

# Purely local; this works before phone.start().
local_report = phone.available_codecs()
print("Local codec offer:", local_report["local_offer"])

# Remote probing requires SIP signalling, so start/register first.
phone.start()
remote_report = phone.available_codecs("1001")
print("Remote codecs:", remote_report["remote"])
print("Call-compatible audio:", remote_report["call_compatible"])
print("Can start call from codec data:", remote_report["can_start_call"])

If the remote side does not include SDP in the OPTIONS response, remote_report["remote"] will be empty and can_start_call will be None.

Development checks

The repository includes CI checks for package compilation and pytest-based tests. To run the same checks locally:

python -m pip install -r requirements-dev.txt
python -m pip install .
python -m compileall -q pyVoIP
python -m pytest -q

Contributors

• TJ Porter (original PyVoIP implementation)

• synodriver (pysilk, a python binding for silk-v3-decoder)
• Nabu Casa
• Home Assistant