baresdk 1.3.0

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

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

# Receive calls
with SDK() as sdk:
    acc = create_account(sdk, "[e-mail]", "secret")
    register(acc)
    for ev in acc.events(timeout=60):
        if ev.type == "incoming_call":
            answer(ev.call)
        elif ev.type == "call_state" and ev.state in ("ended", "failed"):
            break

# Make a call
with SDK() as sdk:
    acc = create_account(sdk, "[e-mail]", "secret")
    register(acc)
    for ev in acc.events(timeout=30):
        if ev.type == "reg_state" and ev.state == "registered":
            call = dial(acc, "[e-mail]")
        elif ev.type == "call_state" and ev.state == "ended":
            break

WSS + DTLS-SRTP + ICE:

acc = 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")

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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
AEC_SUPPRESSOR (default)	Half-duplex	nothing
AEC_WEBRTC	Full-duplex	libwebrtc-audio-processing-1

from baresdk import AEC_WEBRTC
sdk = SDK(aec_mode=AEC_WEBRTC)

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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
verify_tls	bool
audio_codecs	list[str]	["opus", "ulaw", "alaw"]
extra_headers	dict	Added to every REGISTER/INVITE

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Events

ev.type	When
reg_state	Registration changed — unregistered · registering · registered · failed
incoming_call	INVITE received
call_state	Call changed — calling · ringing · established · held · ended · 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)

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