Asyncio SIP micro-library for Python
Project description
aiosipua
Asyncio SIP micro-library for Python. Companion to aiortp.
Built for voice AI backends that need SIP signaling without the bloat of a full SIP stack. Zero runtime dependencies, strict type hints, Python 3.11+.
Features
- SIP message parsing and serialization — RFC 3261 compliant, compact header expansion, multi-value header splitting, structured accessors
- SDP parsing, building, and negotiation — RFC 4566 / RFC 3264, codec selection, DTMF, direction handling, bandwidth support
- Video SDP negotiation —
negotiate_video_sdp,negotiate_av_sdpfor combined audio+video,build_video_sdpfor outgoing video offers - Transports — UDP (
DatagramProtocol) and TCP (Content-Length framing) - UAS — incoming call handling with INVITE/BYE/CANCEL/OPTIONS dispatch,
auto 100 Trying,
IncomingCallhigh-level API - UAC — outbound INVITE with
send_invite, backend-initiated BYE, re-INVITE (hold/unhold), CANCEL, INFO (DTMF) - SIP digest authentication — automatic 401/407 retry with
SipDigestAuthfor outbound calls - Dialog management — RFC 3261 dialog state machine, Record-Route support, in-dialog request/response creation
- Transaction matching — client and server transaction layer
- aiortp bridge —
CallSessionfor audio RTP andVideoCallSessionfor video RTP, bridging SDP negotiation to media with callbacks - X-header support — pass application metadata (room ID, session ID, tenant) through SIP headers
Installation
pip install aiosipua
# With optional RTP support
pip install aiosipua[rtp]
Examples
Parse a SIP message
from aiosipua import SipMessage, parse_sdp
raw = (
"INVITE sip:bob@example.com SIP/2.0\r\n"
"Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK776asdhds\r\n"
"From: Alice <sip:alice@example.com>;tag=1928301774\r\n"
"To: Bob <sip:bob@example.com>\r\n"
"Call-ID: a84b4c76e66710@example.com\r\n"
"CSeq: 314159 INVITE\r\n"
"Contact: <sip:alice@10.0.0.1:5060>\r\n"
"Content-Type: application/sdp\r\n"
"Content-Length: 142\r\n"
"\r\n"
"v=0\r\n"
"o=- 2890844526 2890844526 IN IP4 10.0.0.1\r\n"
"s=-\r\n"
"c=IN IP4 10.0.0.1\r\n"
"t=0 0\r\n"
"m=audio 20000 RTP/AVP 0 8\r\n"
"a=rtpmap:0 PCMU/8000\r\n"
"a=rtpmap:8 PCMA/8000\r\n"
"a=sendrecv\r\n"
)
msg = SipMessage.parse(raw)
# Structured header access
print(msg.from_addr.display_name) # "Alice"
print(msg.from_addr.uri.user) # "alice"
print(msg.to_addr.uri.host) # "example.com"
print(msg.via[0].branch) # "z9hG4bK776asdhds"
print(msg.cseq.method) # "INVITE"
print(msg.call_id) # "a84b4c76e66710@example.com"
# Parse the SDP body
sdp = parse_sdp(msg.body)
audio = sdp.audio
print(audio.port) # 20000
print(audio.codecs[0].encoding_name) # "PCMU"
print(sdp.rtp_address) # ("10.0.0.1", 20000)
SDP negotiation
from aiosipua import parse_sdp, negotiate_sdp, serialize_sdp
# Parse an incoming SDP offer
offer = parse_sdp(sdp_body)
# Negotiate: pick the best codec, build an answer
answer, chosen_pt = negotiate_sdp(
offer=offer,
local_ip="10.0.0.5",
rtp_port=30000,
supported_codecs=[0, 8], # PCMU, PCMA
)
print(f"Chosen codec: payload type {chosen_pt}")
print(serialize_sdp(answer))
Video SDP negotiation
from aiosipua import parse_sdp, negotiate_av_sdp, serialize_sdp
# Negotiate both audio and video from a single offer
offer = parse_sdp(sdp_body)
answer, audio_pt, video_pt = negotiate_av_sdp(
offer=offer,
local_ip="10.0.0.5",
audio_rtp_port=30000,
video_rtp_port=30002,
supported_video_codecs=["H264", "VP8"],
)
print(f"Audio PT: {audio_pt}, Video PT: {video_pt}")
print(serialize_sdp(answer))
Receive calls with the UAS
import asyncio
from aiosipua import IncomingCall, SipUAS
from aiosipua.rtp_bridge import CallSession
from aiosipua.transport import UdpSipTransport
async def handle_invite(call: IncomingCall):
print(f"Incoming call: {call.caller} -> {call.callee}")
print(f"X-headers: {call.x_headers}")
if call.sdp_offer is None:
call.reject(488, "Not Acceptable Here")
return
# Negotiate SDP and create RTP session
session = CallSession(
local_ip="10.0.0.5",
rtp_port=30000,
offer=call.sdp_offer,
)
# Accept the call with the SDP answer
call.ringing()
call.accept(session.sdp_answer)
await session.start()
# Wire up audio and DTMF callbacks
session.on_audio = lambda pcm, ts: print(f"Audio: {len(pcm)} bytes")
session.on_dtmf = lambda digit, dur: print(f"DTMF: {digit}")
def handle_bye(call: IncomingCall, request):
print(f"Call ended: {call.call_id}")
async def main():
transport = UdpSipTransport(local_addr=("0.0.0.0", 5060))
uas = SipUAS(transport)
uas.on_invite = lambda call: asyncio.get_running_loop().create_task(handle_invite(call))
uas.on_bye = handle_bye
await uas.start()
print("Listening on port 5060...")
await asyncio.Event().wait()
asyncio.run(main())
Video call session
from aiosipua import parse_sdp
from aiosipua.video_bridge import VideoCallSession
offer = parse_sdp(sdp_body)
session = VideoCallSession(
local_ip="10.0.0.5",
rtp_port=30002,
offer=offer,
supported_video_codecs=["H264"],
)
await session.start()
# Receive video frames
session.on_frame = lambda nal, ts, kf: process_video(nal, ts, kf)
session.on_keyframe_needed = lambda: encoder.force_keyframe()
# Send video frames
session.send_frame(nal_units, timestamp, keyframe=True)
# Request a keyframe from remote
session.request_keyframe()
await session.close()
Backend-initiated actions with the UAC
from aiosipua import SipUAC
from aiosipua.transport import UdpSipTransport
transport = UdpSipTransport(local_addr=("0.0.0.0", 5060))
uac = SipUAC(transport)
# Hang up a call
uac.send_bye(dialog, remote_addr=("10.0.0.1", 5060))
# Put a call on hold with re-INVITE
from aiosipua import build_sdp
hold_sdp = build_sdp(
local_ip="10.0.0.5",
rtp_port=30000,
payload_type=0,
direction="sendonly",
)
uac.send_reinvite(dialog, sdp=hold_sdp, remote_addr=("10.0.0.1", 5060))
# Send DTMF via SIP INFO
uac.send_info(
dialog,
body="Signal=5\r\nDuration=250\r\n",
content_type="application/dtmf-relay",
remote_addr=("10.0.0.1", 5060),
)
Outbound calls with digest authentication
from aiosipua import SipUAC, SipDigestAuth, build_sdp
from aiosipua.transport import UdpSipTransport
transport = UdpSipTransport(local_addr=("0.0.0.0", 5060))
uac = SipUAC(transport)
sdp = build_sdp(local_ip="10.0.0.5", rtp_port=30000, payload_type=0, codec_name="PCMU")
auth = SipDigestAuth(username="alice", password="secret")
call = uac.send_invite(
from_uri="sip:alice@example.com",
to_uri="sip:bob@example.com",
remote_addr=("proxy.example.com", 5060),
sdp=sdp,
auth=auth, # auto-retries on 401/407
)
Build a SIP message from scratch
from aiosipua import SipRequest, SipResponse, generate_branch, generate_call_id, generate_tag
# Build a SIP request
request = SipRequest(method="OPTIONS", uri="sip:bob@example.com")
request.headers.set_single("Via", f"SIP/2.0/UDP 10.0.0.1:5060;branch={generate_branch()}")
request.headers.set_single("From", f"<sip:alice@example.com>;tag={generate_tag()}")
request.headers.set_single("To", "<sip:bob@example.com>")
request.headers.set_single("Call-ID", generate_call_id())
request.headers.set_single("CSeq", "1 OPTIONS")
# Serialize to bytes for the wire
raw_bytes = bytes(request)
Modify and re-serialize
from aiosipua import SipMessage
msg = SipMessage.parse(raw_sip_text)
# Add a Via header
msg.headers.append("Via", "SIP/2.0/UDP proxy.example.com:5060;branch=z9hG4bKnew")
# Change the Contact
msg.headers.set_single("Contact", "<sip:newhost@10.0.0.99:5060>")
# Add custom X-headers
msg.headers.set_single("X-Room-ID", "room-42")
msg.headers.set_single("X-Session-ID", "sess-abc123")
# Re-serialize (Content-Length auto-updated)
print(msg.serialize())
TCP transport
import asyncio
from aiosipua.transport import TcpSipTransport
async def main():
transport = TcpSipTransport(local_addr=("0.0.0.0", 5060))
# As a server
transport.on_message = lambda msg, addr: print(f"Received from {addr}")
await transport.start()
# Or connect as a client
await transport.connect(("proxy.example.com", 5060))
transport.send(request, ("proxy.example.com", 5060))
asyncio.run(main())
Architecture
┌─────────────┐ ┌──────────────┐ ┌────────────┐
│ SipUAS │────▶│ Dialog │────▶│ SipUAC │
│ (incoming) │ │ (state mgr) │ │ (outgoing)│
└──────┬──────┘ └──────────────┘ └─────┬──────┘
│ │
▼ ▼
┌──────────────┐ ┌──────────────┐ ┌──────────────┐
│ Transaction │ │ SDP/Codec │ │ CallSession │
│ Layer │ │ Negotiation │ │ (audio RTP) │
└──────┬───────┘ └──────┬───────┘ └──────┬───────┘
│ │ │
│ ┌──────┴───────┐ ┌──────┴────────┐
│ │ Video SDP │ │ VideoCall │
│ │ Negotiation │ │ Session │
│ └──────────────┘ │ (video RTP) │
│ └──────┬────────┘
▼ ▼
┌──────────────┐ ┌──────────────┐
│ Transport │ │ aiortp │
│ (UDP / TCP) │ │ (optional) │
└──────────────┘ └──────────────┘
More examples
See the examples/ directory:
echo_server.py— Receives audio via RTP and echoes it backdtmf_ivr.py— Collects DTMF digits and hangs up on#roomkit_prototype.py— Voice AI backend integration with X-header metadata
License
MIT. See LICENSE for details.
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