baresdk 2.1.0

Python binding for baresdk — SIP/WebRTC/NAT SDK

baresdk

Python bindings for a thread-safe SIP/RTP library. Built on baresip + libre.

pip install baresdk

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Features

Category
Transport	UDP · TCP · TLS · WS · WSS
Crypto	SRTP-SDES · DTLS-SRTP
NAT	ICE · STUN · TURN
Calls	Invite · Answer · Hangup · Hold/Resume · DTMF · Blind/Attended transfer
Messaging	SIP MESSAGE · Presence · BLF · MWI
Audio	AEC · NS · AGC · gain · mute · device hot-switch · PCM tap · WAV recording
Stats	RTCP/MOS · jitter · bandwidth · SIP trace · pcap
Multi-account	✓ — unlimited accounts per stack
Thread-safe	✓ — call any API from any thread

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Requirements

Linux — two runtime libraries required:

sudo apt install libssl3 libpulse0                    # Ubuntu/Debian
sudo dnf install openssl-libs pulseaudio-libs         # Fedora/RHEL
sudo pacman -S openssl libpulse                       # Arch

import baresdk prints the exact install command if either is missing.
libwebrtc-audio-processing-1 is optional — enables AEC_WEBRTC full-duplex mode; the SDK works without it.

Windows — only the Visual C++ Redistributable 2015–2022 required. OpenSSL, zlib, and opus are statically embedded.

Quick start

import baresdk as sdk

# Receive calls
sdk.configure(log_level=1)
acc = sdk.create_account("[e-mail]", "secret")
acc.register()

@sdk.on("incoming_call")
def _(ev):
    ev.call.answer()

@sdk.on("ended")
def _(ev):
    sdk.stop()

sdk.run()

# Make a call
sdk.configure(log_level=1)
acc = sdk.create_account("[e-mail]", "secret")
acc.register()

@sdk.on("registered")
def _(ev):
    sdk.call("[e-mail]")

@sdk.on("ended")
def _(ev):
    sdk.stop()

sdk.run()

WSS + DTLS-SRTP + ICE:

sdk.configure(verify_server=True)
acc = sdk.create_account("[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")

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Audio

sdk.set_aec(True)
sdk.set_ns(True)
sdk.set_agc(True)
sdk.set_mic_gain(6.0)       # dB, range −20 to +20
sdk.set_speaker_gain(-3.0)
sdk.set_jitter_buffer(20, 200)

AEC mode	Quality	Requires
Suppressor (default, mode=1)	Half-duplex	nothing
WebRTC (mode=2)	Full-duplex	libwebrtc-audio-processing-1

# WebRTC full-duplex AEC — configure before first account (desktop only, opt-in build)
sdk.configure(aec_mode=2)   # 2 = WEBRTC; requires cmake -DBARESDK_WITH_WEBRTC_AEC=ON

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sdk.create_account parameters

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

sdk.configure() accepts global options (applies before first account):

Key	Example
log_level	0–3
stats_interval_ms	5000
verify_server	False to skip TLS verification
transport	"wss"

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Events — @sdk.on(name) names

Name	When	Sub-names (also available)
reg_state	Registration changed	registering · registered · unregistered · reg_failed
incoming_call	INVITE received	—
call_state	Call changed	calling · ringing · established · held · ended · cancelled · call_failed
dtmf	Digit received	—
media_stats	Periodic RTCP/MOS report	—
transfer_request	REFER received	—
message	SIP MESSAGE received	—
presence_state	Buddy presence changed	—
mwi	Voicemail notification	—
quality_alert	MOS/loss/jitter/RTT threshold crossed	—
sip_trace	Raw SIP message (enable with trace_sip=True)	—
*	Every event (wildcard)	—

@sdk.on("registered")
def _(ev):
    print("registered!")

@sdk.on("incoming_call")
def _(ev):
    print(f"call from {ev.from_uri}")
    ev.call.answer()

@sdk.on("media_stats")
def _(ev):
    print(f"MOS={ev.mos_lq:.2f}  RTT={ev.rtt_ms:.0f}ms  loss={ev.loss_pct:.1f}%")

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