sipx

sipx is a Python SIP protocol library: an httpx-style async client, sans-I/O SIP/SDP/RTP primitives, and media helpers. Use it to send REGISTER, OPTIONS, MESSAGE, INVITE, SUBSCRIBE, and arbitrary SIP methods against any cooperative peer, with Digest authentication, dialog tracking, and UDP/TCP/TLS transports.

Requires Python >=3.14.

Install

From the repository root (editable workspace install):

uv sync

Or install the published package when available:

pip install sipx

Quick start

import asyncio
from sipx import AsyncClient, AuthDigest, Settings

settings = Settings(
    from_uri="sip:1001@example.com",
    timeout=10.0,
)
auth = AuthDigest(username="1001", password="secret")

async def main() -> None:
    async with AsyncClient(settings=settings, auth=auth) as client:
        response = await client.register("sip:pbx.example.com:5060")
        print(response.status_code, response.reason)

asyncio.run(main())

AsyncClient

AsyncClient is the main runtime. One client, one Settings, optional AuthDigest, and optional event_hooks.

UAC methods

Method	RFC	Purpose
register(uri, **headers)	3261	REGISTER / unregister (Expires: 0)
options(uri)	3261 §11	Capability probe
message(uri, body, **headers)	3428	SIP MESSAGE
invite(uri, **kwargs)	3261 §13	Start a session
subscribe(uri, **headers)	3265	Event subscription
request(method, uri, **kwargs)	—	Any SIP method (INFO, NOTIFY, …)
ack(call_id)	3261 §13.2.2.4	Confirm a 2xx INVITE
bye(call_id)	3261 §15	Terminate a dialog
cancel(call_id)	3261 §9	Abort a pending INVITE

UAS handlers

Register inbound handlers with decorators; deliver requests with handle_request():

from sipx import AsyncClient, Response

client = AsyncClient()

@client.on_options
async def handle_options(request) -> Response:
    return Response(
        status_code=200,
        reason="OK",
        headers={"Allow": "INVITE, ACK, BYE, OPTIONS, MESSAGE"},
        request=request,
    )

# In your receive loop:
# response = await client.handle_request(inbound_request)

Responses and history

Every UAC call returns one final Response (the first >= 200). Provisional 1xx replies and 401/407 Digest challenges are collected on response.history in arrival order, each with its own .request:

response = await client.invite("sip:6000@pbx.example.com")
for item in response.history:
    print(item.status_code, item.reason)
print(response.status_code)  # final

Event hooks

Hooks follow the httpx pattern — a dict of event name to list of callables (sync or async, side-effect only):

def log_request(request) -> None:
    print(f"> {request.method} {request.uri}")

def log_response(response) -> None:
    print(f"< {response.status_code}")

client = AsyncClient(
    event_hooks={"request": [log_request], "response": [log_response]},
)

CANCEL a ringing INVITE

invite() blocks until the final response. To abort a still-ringing call, run the INVITE in one task and cancel() from another. Pass an explicit Call-ID so the cancel task knows what to target:

import asyncio

async with AsyncClient(settings=settings, auth=auth) as client:
    call_id = "demo-call-1"
    invite = asyncio.create_task(
        client.invite("sip:6000@pbx.example.com", **{"Call-ID": call_id})
    )
    await asyncio.sleep(1)
    await client.cancel(call_id)   # 200 OK to CANCEL
    response = await invite        # 487 Request Terminated

Settings

from sipx import Settings

settings = Settings(
    from_uri="sip:1001@example.com",
    contact_uri="sip:1001@192.168.1.10:5060",
    local_host="0.0.0.0",
    local_port=0,          # 0 = ephemeral
    timeout=10.0,
    rport=True,            # RFC 3581; add ;rport to UDP Via
    retransmit=True,       # RFC 3261 §17 UDP retransmission
    user_agent="my-app/1.0",
)

AsyncClient.learned_address exposes the public (host, port) learned from received/rport parameters echoed on response Via headers.

RFC behavior

AsyncClient correlates replies by Call-ID, CSeq number/method, top Via branch, and destination port. For hostname targets the source IP is accepted (the datagram arrives from the resolved address); for IP-literal targets the source host must match.

Implemented client behavior:

• UDP retransmission (§17). T1/T2 backoff until reply or timeout; INVITE stops after the first provisional; TCP/TLS never retransmit.
• Non-2xx INVITE ACK (§17.1.1.3). Auto-ACK on 3xx–6xx finals.
• CANCEL (§9). cancel(call_id) for a pending INVITE.
• rport (RFC 3581). Outgoing UDP Via carries ;rport; toggle with Settings.rport.
• PRACK / 100rel (RFC 3262). Reliable provisionals (RSeq + 100rel) are auto-PRACKed with RAck.
• Digest (RFC 7616/8760). AuthDigest supports MD5, MD5-sess, SHA-256, SHA-256-sess; one challenge retry per request.
• Dialog tags (§12.2.2). UAC dialogs reject conflicting From/To tags.
• Content-Length. Added on every outbound Request.to_bytes() (needed for TCP/TLS framing).

Inbound requests are not auto-routed: the receive loop dispatches responses to in-flight calls; call handle_request() yourself to reach on_* handlers.

Standalone extension handlers under sipx/extensions/ (PRACK, DNS, events, presence, outbound) are test-only and not wired into AsyncClient.

Core modules

Module	Contents
sipx.client	AsyncClient, transports, response correlation
sipx.models	Request, Response dataclasses
sipx.config	Settings
sipx.protocol	Transactions, dialog, AuthDigest, event hooks
sipx.transport	UDP, TCP, TLS, registry
sipx.sip	Sans-I/O parser, serializer, URI, Digest helpers
sipx.sdp	Session description, offer/answer (PCMU, PCMA, telephone-event)
sipx.rtp	Packet parse/serialize, G.711, RFC4733 DTMF, jitter buffer, RtpAudioSession
sipx.media	AudioFrame, synthetic silence/noise sources, optional PyAudio

Root exports include AsyncClient, AuthDigest, Settings, Request, Response, EventHooks, and the SIP/SDP/RTP types above.

Examples

Runnable scripts under sipx.examples use only the core package. Run from the repository root with uv run. They default to the public Mizu demo account but read generic SIPX_* env vars so the same code targets any SIP provider:

export SIPX_LOCAL_HOST=<your-local-ip>
export SIPX_TARGET=sip:<target>@demo.mizu-voip.com:37075

# Optional overrides for non-Mizu providers:
export SIPX_AOR=sip:1001@example.com
export SIPX_REGISTRAR=sip:pbx.example.com:5060
export SIPX_USERNAME=1001
export SIPX_PASSWORD=...
export SIPX_REMOTE_HOST=pbx.example.com
export SIPX_REMOTE_PORT=5060

uv run python -m sipx.examples.register
uv run python -m sipx.examples.unregister
uv run python -m sipx.examples.options
uv run python -m sipx.examples.message
uv run python -m sipx.examples.subscribe
uv run python -m sipx.examples.invite
uv run python -m sipx.examples.call
uv run python -m sipx.examples.cancel
uv run python -m sipx.examples.info_dtmf
uv run python -m sipx.examples.hooks_history
uv run python -m sipx.examples.server

Set SIPX_DEBUG=1 to print SIP wire to stderr, for example SIPX_DEBUG=1 uv run python -m sipx.examples.options.

Example	Needs live peer	Notes
register, unregister	yes	SIPX_EXPIRES
options, message	yes	SIPX_MESSAGE, SIPX_CONTENT_TYPE
subscribe	yes	SIPX_EVENT, SIPX_ACCEPT
invite, call	yes	SIPX_CODECS, SIPX_RTP_PORT
cancel	yes	SIPX_CANCEL_AFTER (seconds before CANCEL)
info_dtmf, hooks_history	yes	in-dialog INFO / event hooks demo
server	no	offline UAS via handle_request

Development

ruff format --check .
ruff check .
uv run ty check
uv run pytest

Core tests live under tests/. See SPEC.md, DESIGN.md, and CHANGELOG.md for implementation detail and version history.

Security

• Redact Authorization and Proxy-Authorization before logging SIP wire.
• Redact SDP a=crypto lines in artifacts and debug output.
• Do not commit real account passwords or provider API keys.
• Use TLS (transport="tls") for SIP over untrusted networks.
• Obtain consent and respect CPS limits before placing outbound calls.