baresdk 1.0.0

Python binding for baresdk — SIP/WebRTC/NAT SDK

baresdk — Python SIP/VoIP SDK

Python binding for baresdk — a thread-safe SIP/RTP library for desktop softphones, VoIP clients, and call-center apps. Built on baresip and libre, delivered as a self-contained shared library.

pip install baresdk

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Features

Category	What's included
Transport	UDP · TCP · TLS · WS · WSS
Crypto	SRTP-SDES · DTLS-SRTP
NAT	ICE · STUN · TURN
Calls	Invite · Answer · Hangup · Hold/Resume · DTMF
Transfer	Blind (REFER) · Attended (REFER w/ Replaces)
Messaging	SIP MESSAGE send/receive
Presence	PUBLISH · SUBSCRIBE/NOTIFY · BLF · MWI
Reliability	PRACK / 100rel (RFC 3262)
Audio	Mic/speaker gain · mute · device hot-switch · AEC · NS · AGC · PCM tap · WAV recording
Observability	SIP trace · SDP diff · pcap · RTCP/MOS stats · jitter buffer · bandwidth
Multi-account	Yes — any number of accounts per stack
Thread safety	Full — call any API from any thread

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Platform requirements

Linux

One library is not pre-installed on all distros:

# Ubuntu / Debian
sudo apt install libwebrtc-audio-processing-1

# Fedora / RHEL / CentOS
sudo dnf install webrtc-audio-processing

# Arch
sudo pacman -S webrtc-audio-processing

Everything else (libssl, libz, libpulse, …) is already on every desktop Linux.

If the library is missing, import baresdk raises ImportError with the exact install command for your distro — you don't need to remember the package name.

Windows

No extra installs needed. OpenSSL, zlib, and opus are all statically embedded in baresdk.dll. The only requirement is the Visual C++ Redistributable (2015–2022).

Quick start

Register and receive calls

from baresdk import SDK, create_account, register, answer, hangup

with SDK(log_level=1) as sdk:
    account = create_account(sdk, "[e-mail]", "secret")
    register(account)

    for ev in account.events(timeout=60):
        if ev.type == "reg_state" and ev.state == "registered":
            print("Registered!")

        elif ev.type == "incoming_call":
            print(f"Incoming call from {ev.from_uri}")
            answer(ev.call)

        elif ev.type == "call_state" and ev.state in ("ended", "failed", "cancelled"):
            print("Call ended.")
            break

Make an outgoing call

from baresdk import SDK, create_account, register, dial

with SDK(log_level=1) as sdk:
    account = create_account(sdk, "[e-mail]", "secret",
                             transport="tls")
    register(account)

    for ev in account.events(timeout=30):
        if ev.type == "reg_state" and ev.state == "registered":
            call = dial(account, "[e-mail]")

        elif ev.type == "media_stats":
            print(f"MOS: {ev.mos_lq:.2f}  RTT: {ev.rtt_ms:.0f} ms")

        elif ev.type == "call_state" and ev.state == "ended":
            break

WSS + DTLS-SRTP + ICE (WebRTC-compatible PBX)

from baresdk import SDK, create_account, register

with SDK(log_level=1) as sdk:
    account = create_account(sdk,
        "[e-mail]", "secret",
        transport   = "wss",
        server_url  = "wss://pbx.example.com/ws",
        media_enc   = "dtls_srtp",
        ice_enabled = True,
        stun_server = "stun:stun.l.google.com:19302",
        verify_tls  = True,
    )
    register(account)

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Audio controls

# Echo cancellation, noise suppression, AGC
sdk.set_aec(True)
sdk.set_ns(True)
sdk.set_agc(True)

# Mic and speaker gain (dB, range −20 to +20; 0.0 = bypass)
sdk.set_mic_gain(6.0)      # boost a quiet mic
sdk.set_speaker_gain(-3.0) # attenuate speaker

# Jitter buffer
sdk.set_jitter_buffer(20, 200)

Echo cancellation modes

Mode	Duplex	CPU	Best for
AEC_SUPPRESSOR (default)	Half-duplex	Negligible	Mobile / simple calls
AEC_WEBRTC	Full-duplex	Moderate	Desktop softphone / call center

from baresdk import AEC_WEBRTC

sdk = SDK(aec_mode=AEC_WEBRTC)   # full-duplex — requires libwebrtc-audio-processing-1

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create_account options

Parameter	Type	Values
transport	str	"udp" "tcp" "tls" "ws" "wss"
media_enc	str	"none" "sdes" "dtls_srtp"
ice_enabled	bool
stun_server	str	e.g. "stun:stun.l.google.com:19302"
turn_server	str	e.g. "turn:turn.example.com:3478"
server_url	str	WebSocket URL — overrides transport/host/port
verify_tls	bool
audio_codecs	list[str]	["opus", "ulaw", "alaw"]
extra_headers	dict[str,str]	Added to every outbound REGISTER/INVITE

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Events reference

ev.type	Fired when
"reg_state"	Registration state changed
"incoming_call"	Incoming INVITE received
"call_state"	Call state changed
"dtmf"	DTMF digit received
"media_stats"	Periodic RTCP/MOS stats
"sip_trace"	Raw SIP message (trace_sip=True)
"transfer_request"	Incoming REFER
"mwi"	Voicemail notification
"message"	SIP MESSAGE received
"presence_state"	Buddy presence changed
"quality_alert"	MOS/loss/jitter/RTT threshold crossed

reg_state values: "unregistered" "registering" "registered" "failed" "unregistering"

call_state values: "calling" "ringing" "established" "held" "ended" "cancelled" "failed"