xbridge

A self-hosted voice gateway that connects SIP phone calls to WebSocket audio and a REST API. One binary, a YAML config, and your app gets real-time call audio over WebSocket and full call control over REST.

xbridge is optional — use it when your application is in Python, Node, or another language without a native xphone library. If you're building in Go or Rust, use xphone directly instead. Powered by xphone-rust.

Status — Beta

xbridge is in active development and used in production alongside xphone-rust. Single-node only — no built-in clustering or HA. Run multiple instances behind a load balancer for redundancy.

Scope and limitations

xbridge is a voice data plane — real-time SIP signaling, RTP media, and WebSocket audio streaming. It is intentionally stateless: no database, no persistent storage, no recording engine.

What xbridge is not:

Not a full Twilio replacement. xbridge provides Twilio-compatible WebSocket audio framing, but it does not include number provisioning, billing, CDR storage, call recording, or dashboards. Those are your application's responsibility.
Audio only — no video support. xbridge handles voice calls exclusively.
Blind transfer only — attended transfer is not available via the REST API.
Narrowband audio — WebSocket audio is 8 kHz (mu-law or PCM16). Wideband codecs (G.722, Opus) are negotiated on the SIP side but downsampled for the WebSocket stream.
Single-node — no built-in clustering or HA.
SRTP uses SDES key exchange only — DTLS-SRTP is not supported.
Webhook dead letter queue is in-memory only — undelivered webhook events are lost on restart.

For production deployments that need recordings, CDR storage, billing, or dashboards, build a separate control plane that consumes xbridge's webhooks and WebSocket audio.

Tested against

Category	Tested with
SIP trunks	Telnyx, Twilio SIP, VoIP.ms
PBXes	Asterisk (via xpbx), FreePBX, 3CX
NAT traversal	STUN and TURN supported (configured via xphone-rust engine)
WebSocket clients	Twilio Media Streams-compatible apps, native binary mode clients

This is not a comprehensive compatibility matrix. If you hit issues with a provider or PBX not listed here, please open an issue.

Use cases

AI voice agents — caller audio arrives over WebSocket, your app runs STT + LLM + TTS, sends audio back
IVR systems — DTMF detection, audio playback, and call routing driven by your backend via REST + webhooks
Call centers — route incoming calls via webhooks, hold/transfer/mute via REST, tap audio for transcription
Call recording — connect to the WebSocket audio stream, write frames to disk or S3
Outbound dialers — originate calls via REST API, play audio, detect DTMF responses

Quick Start

With Docker

docker run -v ./config.yaml:/etc/xbridge/config.yaml \
  -p 8080:8080 -p 10000-10100:10000-10100/udp \
  ghcr.io/x-phone/xbridge:latest

Pre-built Binaries

Download from GitHub Releases for Linux and macOS (amd64/arm64):

curl -L https://github.com/x-phone/xbridge/releases/latest/download/xbridge-linux-amd64 -o xbridge
chmod +x xbridge
./xbridge --config config.yaml

From Source

cargo build --release
./target/release/xbridge --config config.yaml

Set RUST_LOG=info (or debug, trace) for logging output.

Demo

A full-stack Voice AI demo is included: softphone → PBX → xbridge → AI agent with live transcription.

Softphone (ext 1001)
    → Asterisk/xpbx (dial 2000)
        → xbridge (SIP trunk host, WebSocket audio)
            → voice-app (Deepgram STT/TTS, React UI)

git clone https://github.com/x-phone/xpbx.git   # sibling directory
cd xbridge/demo/voice-ai
echo "EXTERNAL_IP=$(ipconfig getifaddr en0)" > .env  # macOS
docker compose up --build

Open http://localhost:3000, enter a Deepgram API key, register a softphone as extension 1001 (password: password123) on your machine's IP port 5060, and dial 2000 to talk to the AI agent.

A simpler SIP Trunk Demo connects xbridge directly to Twilio or Telnyx for PSTN calls, no PBX needed.

Connection Modes

xbridge supports three ways to connect to the phone network. All three deliver the same interface to your app — same webhooks, same WebSocket audio, same REST API. Your app speaks HTTP and WebSocket; xbridge handles all the SIP/RTP complexity.

SIP extension — register on any PBX (Asterisk, FreePBX, 3CX) as an extension, just like a softphone would
SIP trunk client — register with a cloud provider (Telnyx, Twilio, VoIP.ms) to get a real phone number
SIP trunk host — run a SIP server and accept calls directly from PBX systems or trunk providers

All three modes can run simultaneously in the same configuration.

Configuration

xbridge loads config from a YAML or TOML file. See config.example.yaml for all options.

xbridge --config config.yaml

SIP Extension (register on a PBX)

Register as an extension on your office PBX — the simplest mode. Calls to your extension trigger webhooks and audio streams to your app:

sip:
  username: "2000"
  password: "secret"
  host: "192.168.1.10"       # your PBX address
  rtp_port_min: 10000
  rtp_port_max: 10100

SIP Trunk Client (cloud provider)

Register with a SIP trunk provider to get a real phone number:

sip:
  username: "your-username"
  password: "your-password"
  host: "sip.telnyx.com"
  transport: "tls"
  srtp: true

For multiple providers, use the trunks array:

trunks:
  - name: "telnyx"
    username: "1001"
    password: "secret1"
    host: "sip.telnyx.com"
    transport: "tls"
    srtp: true
  - name: "twilio"
    username: "2001"
    password: "secret2"
    host: "sip.twilio.com"

SIP Trunk Host (accept calls from PBX)

Run a SIP server that PBX systems connect to directly:

server:
  listen: "0.0.0.0:5080"
  peers:
    - name: "office-pbx"
      host: "192.168.1.10"
      codecs: ["ulaw", "alaw"]
    - name: "remote-office"
      auth:
        username: "remote-trunk"
        password: "secret"

Peers authenticate via IP allowlist (host), SIP digest credentials (auth), or both.

Webhook and Stream

webhook:
  url: "http://localhost:3000/events"
  timeout: "5s"
  retry: 1

stream:
  encoding: "audio/x-mulaw"   # audio/x-mulaw | audio/x-l16
  sample_rate: 8000

Environment Variable Overrides

Every config field can be overridden via XBRIDGE_* environment variables:

Variable	Description
`XBRIDGE_LISTEN_HTTP`	HTTP listen address (default: `0.0.0.0:8080`)
`XBRIDGE_SIP_USERNAME`	SIP username
`XBRIDGE_SIP_PASSWORD`	SIP password
`XBRIDGE_SIP_HOST`	SIP registrar host
`XBRIDGE_SIP_TRANSPORT`	`udp`, `tcp`, or `tls`
`XBRIDGE_SIP_SRTP`	`true` or `false`
`XBRIDGE_SIP_STUN_SERVER`	STUN server address
`XBRIDGE_SIP_RTP_PORT_MIN`	RTP port range start
`XBRIDGE_SIP_RTP_PORT_MAX`	RTP port range end
`XBRIDGE_WEBHOOK_URL`	Webhook endpoint URL
`XBRIDGE_WEBHOOK_TIMEOUT`	Webhook timeout (e.g. `5s`)
`XBRIDGE_WEBHOOK_RETRY`	Webhook retry count
`XBRIDGE_STREAM_ENCODING`	`audio/x-mulaw` or `audio/x-l16`
`XBRIDGE_STREAM_SAMPLE_RATE`	Sample rate in Hz
`XBRIDGE_AUTH_API_KEY`	API key for Bearer token auth
`XBRIDGE_RATE_LIMIT_RPS`	Requests per second limit
`XBRIDGE_TLS_CERT`	Path to TLS certificate PEM
`XBRIDGE_TLS_KEY`	Path to TLS private key PEM

REST API

All endpoints require Authorization: Bearer <api_key> when auth.api_key is configured.

Method	Endpoint	Description
`POST`	`/v1/calls`	Create outbound call (via trunk, peer, or extension)
`GET`	`/v1/calls`	List active calls
`GET`	`/v1/calls/{id}`	Get call details
`DELETE`	`/v1/calls/{id}`	Hang up call
`POST`	`/v1/calls/{id}/hold`	Hold call
`POST`	`/v1/calls/{id}/resume`	Resume held call
`POST`	`/v1/calls/{id}/transfer`	Blind transfer
`POST`	`/v1/calls/{id}/dtmf`	Send DTMF digits
`POST`	`/v1/calls/{id}/mute`	Mute outbound audio
`POST`	`/v1/calls/{id}/unmute`	Unmute
`POST`	`/v1/calls/{id}/play`	Play audio (URL or inline PCM)
`POST`	`/v1/calls/{id}/play/stop`	Stop playback
`GET`	`/v1/webhooks/failures`	List failed webhook deliveries
`DELETE`	`/v1/webhooks/failures`	Drain failed webhook queue
`GET`	`/health`	Health check
`GET`	`/metrics`	Prometheus metrics

See the API Reference for request/response schemas and examples.

WebSocket Audio

Connect to ws://host:8080/ws/{call_id} to stream audio for a call.

Twilio-compatible mode (default) — JSON text frames with base64-encoded audio. Compatible with apps built for Twilio Media Streams.

Native binary mode (?mode=native) — binary frames with raw PCM16 audio. Lower overhead, no JSON/base64 encoding per frame.

Both modes deliver the same lifecycle events (connected, start, media, dtmf, mark, stop). See the API Reference for protocol details.

Webhooks

xbridge fires webhook events to your configured URL for call lifecycle management.

Incoming calls are POSTed to {webhook_url}/incoming. Your app responds with accept or reject.

Lifecycle events are POSTed to {webhook_url}:

Event	Description
`call.ringing`	Outbound call is ringing
`call.answered`	Call was answered
`call.ended`	Call ended (includes reason and duration)
`call.dtmf`	DTMF digit received
`call.hold`	Call placed on hold
`call.resumed`	Call resumed from hold
`call.play_finished`	Audio playback completed or interrupted

Failed deliveries are retried with exponential backoff and stored in an in-memory dead letter queue. See the API Reference for payload schemas.

Architecture

xbridge is a data plane — it handles real-time SIP signaling, RTP media, and WebSocket audio streaming. It is intentionally stateless: no database, no persistent storage, no recording engine.

                                  webhooks     ┌─────────────────┐
                              ──────────────> │                 │
┌──────────────┐              ┌───────────┐   │   Your App      │──> DB, S3, dashboards
│  SIP Trunk   │──SIP/RTP──> │           │ <──│                 │
│  (Telnyx,    │   client    │           │    └────────┬────────┘
│   Twilio)    │              │  xbridge  │             │
└──────────────┘              │           │        WebSocket
                              │           │       (audio stream)
┌──────────────┐              │           │             │
│  PBX         │──SIP/RTP──> │           │             v
│  (Asterisk,  │   :5080     │           │    ┌──────────────┐
│   FreePBX)   │              └───────────┘    │  Recording / │
└──────────────┘                               │  Transcription│
                                               └──────────────┘

Monitoring

Health Check

GET /health → {"status": "ok", "sip_trunks": 1, "sip_server": false, "active_calls": 0}

Prometheus Metrics

GET /metrics

Metric	Type	Description
`xbridge_calls_total`	counter	Total calls by direction
`xbridge_active_calls`	gauge	Currently active calls
`xbridge_call_duration_seconds`	histogram	Call duration
`xbridge_http_requests_total`	counter	HTTP API requests
`xbridge_http_request_duration_seconds`	histogram	HTTP request latency
`xbridge_ws_connections`	gauge	Active WebSocket connections
`xbridge_ws_frames_total`	counter	WebSocket frames sent/received
`xbridge_webhooks_total`	counter	Webhook deliveries by result
`xbridge_webhook_duration_seconds`	histogram	Webhook delivery latency
`xbridge_trunk_calls_total`	counter	Calls from trunk host peers
`xbridge_rate_limit_rejections_total`	counter	Rate-limited requests

Features

Connection modes — stable

SIP extension (register on PBX)
SIP trunk client (Telnyx, Twilio, VoIP.ms, etc.)
SIP trunk host (accept calls from PBX systems)
Multi-trunk (multiple SIP providers simultaneously)

Call control — stable

Inbound and outbound calls
Hold / resume
Blind transfer
DTMF send and receive
Mute / unmute
Audio playback (URL or inline PCM)

Audio streaming — stable

Twilio-compatible WebSocket mode (JSON/base64)
Native binary WebSocket mode (PCM16)
Configurable encoding (mu-law, linear16)

Security — stable

Bearer token API authentication
IP allowlist and SIP digest peer authentication
Rate limiting
TLS (rustls)
SRTP (via SIP trunk)

Observability — stable

Prometheus metrics
Health check endpoint
Webhook dead letter queue

Ops — stable

Single binary, zero runtime dependencies
YAML / TOML / env var configuration
Docker image (amd64 + arm64)
Graceful shutdown (SIGTERM/SIGINT)

Integration Guide

See the Integration Guide for a step-by-step walkthrough of building an AI voice agent with xbridge, including Python code examples, Twilio migration instructions, and a production checklist.

Roadmap

DTLS-SRTP key exchange (blocked on xphone-rust support)
Attended transfer via REST API
Wideband WebSocket audio (G.722/Opus passthrough without downsampling)

Clustering and HA are intentionally out of scope — xbridge is designed as a stateless single-node process. Scale horizontally by running multiple instances.

Development

cargo test          # Run all tests
cargo clippy        # Lint
cargo fmt           # Format

See CONTRIBUTING.md for project layout and architecture details.

License

MIT

Name		Name	Last commit message	Last commit date
Latest commit History 66 Commits
.github		.github
demo		demo
docs		docs
src		src
tests		tests
.gitignore		.gitignore
CONTRIBUTING.md		CONTRIBUTING.md
Cargo.lock		Cargo.lock
Cargo.toml		Cargo.toml
Dockerfile		Dockerfile
Dockerfile.release		Dockerfile.release
LICENSE		LICENSE
README.md		README.md
config.example.yaml		config.example.yaml
docker-compose.yaml		docker-compose.yaml
gh-setup.yaml		gh-setup.yaml

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xbridge

Table of Contents

Status — Beta

Scope and limitations

Tested against

Use cases

Quick Start

With Docker

Pre-built Binaries

From Source

Demo

Connection Modes

Configuration

SIP Extension (register on a PBX)

SIP Trunk Client (cloud provider)

SIP Trunk Host (accept calls from PBX)

Webhook and Stream

Environment Variable Overrides

REST API

WebSocket Audio

Webhooks

Architecture

Monitoring

Health Check

Prometheus Metrics

Features

Connection modes — stable

Call control — stable

Audio streaming — stable

Security — stable

Observability — stable

Ops — stable

Integration Guide

Roadmap

Development

License

About

Topics

Resources

License

Contributing

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