Active Call

active-call is a standalone Rust crate designed for building AI Voice Agents. It provides a high-performance infrastructure for bridging AI models with real-word telephony and web communications.

Key Capabilities

1. Multi-Protocol Audio Gateway

active-call supports a wide range of communication protocols, ensuring compatibility with both legacy and modern systems:

• SIP (Telephony): Supports standard SIP signaling. Can act as a SIP Client/Extension to register with PBX systems like FreeSWITCH or Asterisk, or handle direct SIP incoming/outgoing calls.
• WebRTC: Direct browser-to-agent communication with low-latency SRTP.
• Voice over WebSocket: A highly flexible API for custom integrations. Push raw PCM/encoded audio over WebSocket and receive real-time events.

2. Dual-Engine Dialogue Support

Choose the pipeline that fits your latency and cost requirements:

• Traditional Serial Pipeline: Integrated VAD → ASR → LLM → TTS flow. Supports various providers (OpenAI, Aliyun, Azure, Tencent) with optimized buffering and endpoints.
• Realtime Streaming Pipeline: Native support for OpenAI/Azure Realtime API. True full-duplex conversational AI with ultra-low latency, server-side VAD, and emotional nuance.

3. Playbook: The Best Practice for Dialogue

The Playbook system is our recommended way to build complex, stateful voice agents:

• Markdown-Driven: Define personas, instructions, and flows in readable Markdown files.
• Stateful Scenes: Manage conversation stages with easy transitions (Scene switching).
• Tool Integration: Built-in support for DTMF, SIP Refer (Transfer), and custom Function Calling.
• Advanced Interaction: Smart interruptions, filler word filtering, background ambiance, and automated post-call summaries via Webhooks.

4. High-Performance Media Core

• Low-Latency VAD: Includes TinySilero (optimized Rust implementation), significantly faster than standard ONNX models.
• Flexible Processing Chain: Easily add noise reduction, echo cancellation, or custom audio processors.
• Codec Support: PCM16, G.711 (PCMU/PCMA), G.722, and Opus.

Protocol Flexibility

Voice over WebSocket

For developers who need full control, active-call provides a raw audio-over-websocket interface. This is ideal for custom web apps or integration with existing AI pipelines where you want to handle the audio stream manually.

SIP PBX Integration

active-call can be integrated into existing corporate telephony:

• As an Extension: Register active-call to your FreeSWITCH or Asterisk PBX like any other VoIP phone. AI agents can then receive calls from internal extensions or external trunks.
• As a Trunk: Handle incoming SIP traffic directly from carriers.

SIP Invitation Handlers

active-call provides flexible handling of incoming SIP invitations through configurable handlers:

Quick CLI Configuration

For rapid setup without editing config files, use the --handler CLI parameter:

# Webhook handler
./active-call --handler https://example.com/webhook

# Playbook handler (default playbook)
./active-call --handler default.md

The handler type is automatically detected:

• URLs starting with http:// or https:// become Webhook handlers
• Files ending with .md become Playbook handlers (set as default)

Webhook Handler

Forward incoming SIP invitations to an HTTP endpoint for custom call routing logic:

[handler]
type = "webhook"
url = "http://localhost:8090/webhook"
method = "POST"

Playbook Handler

Automatically route calls to specific playbooks based on caller/callee patterns using regex matching:

[handler]
type = "playbook"
default = "default.md"  # Optional: fallback when no rules match

[[handler.rules]]
caller = "^\\+1\\d{10}$"        # Match US numbers
callee = "^sip:support@.*"      # Match support line
playbook = "support.md"         # Use support playbook

[[handler.rules]]
caller = "^\\+86\\d+"           # Match Chinese numbers
playbook = "chinese.md"         # Language-specific playbook

[[handler.rules]]
callee = "^sip:sales@.*"        # Match sales line
playbook = "sales.md"           # Use sales playbook

How it works:

• Rules are evaluated in order from top to bottom
• Each rule can specify caller and/or callee regex patterns
• Both patterns must match for the rule to apply (omitted patterns match any value)
• The first matching rule determines which playbook to use
• If no rules match and no default is set, the call is rejected
• Playbook files should be placed in the config/playbook/ directory

Use cases:

• Route calls from different regions to language-specific agents
• Direct calls to different departments based on the dialed number
• Provide specialized handling for VIP customers based on caller ID
• Implement time-based routing with multiple handler configurations

SIP Call with Playbook Example

Here's a complete example of setting up active-call to handle incoming SIP calls with playbooks:

1. Create a Playbook File

Create config/playbook/greeting.md:

---
asr:
  provider: "openai"
tts:
  provider: "openai"
  voice: "alloy"
llm:
  provider: "openai"
  model: "gpt-4o-mini"
---

# Scene: greeting

You are a friendly AI assistant. Greet the caller warmly and ask how you can help them today.

2. Configure the Handler

Option A: Using CLI (Quick Start):

# Start with playbook handler
./active-call --handler config/playbook/greeting.md --sip 0.0.0.0:5060

Option B: Using Configuration File (config.toml):

[handler]
type = "playbook"
default = "greeting.md"

# Optional: Add routing rules
[[handler.rules]]
callee = "^sip:sales@.*"
playbook = "sales.md"

[[handler.rules]]
callee = "^sip:support@.*"
playbook = "support.md"

3. Make a SIP Call

Using any SIP client (like linphone, zoiper, or sipbot):

# Using pjsua (PJSIP command line tool)
pjsua --use-ice --null-audio sip:agent@your-server-ip:5060

# Or using sipbot (for testing)
sipbot -target sip:agent@your-server-ip:5060 -duration 30

4. What Happens

1. active-call receives the SIP INVITE
2. Playbook handler matches the call (by rules or default)
3. Validates that the playbook file exists
4. Accepts the SIP call and establishes RTP media
5. Loads and runs the playbook
6. AI agent converses with the caller
7. Call ends when either party hangs up

5. Monitoring

Check logs to see the call flow:

RUST_LOG=info ./active-call --handler greeting.md

# You'll see:
# INFO active_call::useragent::playbook_handler: matched playbook for invite
# INFO active_call::handler::handler: Playbook runner started for greeting.md
# INFO active_call::handler::handler: new call started

Error Handling

If the playbook file doesn't exist:

• The handler validates the file before accepting the SIP call
• Sends 503 Service Unavailable response to the caller
• Logs: Playbook file not found, rejecting SIP call

This prevents accepting calls that can't be properly handled.

Playbook Demo

VAD Performance

Benchmarked on 60 seconds of 16kHz audio (Release mode):

VAD Engine	Implementation	Time (60s)	RTF (Ratio)	Note
TinySilero	Rust (Optimized)	~60.0 ms	0.0010	>2.5x faster than ONNX
ONNX Silero	ONNX Runtime	~158.3 ms	0.0026	Standard baseline
WebRTC VAD	C/C++ (Bind)	~3.1 ms	0.00005	Legacy, less accurate

API Documentation

For detailed information on REST endpoints and WebSocket protocols, please refer to the API Documentation.

SDKs

• Go SDK: rustpbxgo - Official Go client for building voice applications with active-call.

Dialogue Features

active-call provides a rich set of features for building natural and responsive voice agents:

• Scene Management (Playbook):

  • Markdown-based configuration: Define agent behavior using simple Markdown files.
  • Multiple Scenes: Split long conversations into manageable stages (Scenes) using # Scene: name headers.
  • Dynamic Variables: Supports {{ variable }} syntax using minijinja for personalized prompts.
  • LLM Integration: Streaming responses with configurable prompts and tool-like tags.
  • Background Ambiance: Supports looping background audio (e.g., office noise) with auto-ducking during conversation.
  • Follow-up: Automatically follow up with the user after a period of silence (e.g., "Are you still there?").
  • DTMF Support: Standard IVR functionality. Define global or scene-specific keypress actions (jump to scene, transfer call, or hang up).
  • Pre-recorded Audio: Play fixed audio files (.wav/.pcm) by adding <play file="path/to/audio.wav" /> at the beginning of a scene.
  • Post-hook & Summary: Automatically generate conversation summaries and send them to a Webhook URL after the call ends. Supports multiple summary templates:
    • short: One or two sentence summary.
    • detailed: Comprehensive summary with key points and decisions.
    • intent: Extracted user intent.
    • json: Structured JSON summary.
    • custom: Fully custom summary prompt.

• Advanced Voice Interaction:

  • Smart Interruption: Multiple strategies (vad, asr, or both), filler word filtering, and protection periods.
  • Graceful Interruption: Supports audio fade-out instead of hard cuts when the user starts speaking.
  • EOU (End of Utterance) Optimization: Starts LLM inference as soon as silence is detected, before ASR final results are ready.

Playbook Configuration

Playbooks are defined in Markdown files with YAML frontmatter. The frontmatter configures the voice capabilities, while the body defines the agent's persona and instructions.

Example

---
asr:
  provider: "openai"
tts:
  provider: "openai"
llm:
  provider: "openai"
  model: "gpt-4-turbo"
dtmf:
  "0": { action: "hangup" }
followup:
  timeout: 10000
  max: 3
posthook:
  url: "https://api.example.com/webhook"
  summary: "detailed"
  includeHistory: true
---

# Scene: greeting
<dtmf digit="1" action="goto" scene="tech_support" />

You are a friendly AI for {{ company_name }}. 
Greet the caller and ask if they need technical support (Press 1) or billing help.

# Scene: tech_support
You are now in tech support. How can I help with your system?
To speak to a human, I can transfer you: <refer to="sip:human@domain.com" />

Configuration Reference

Section	Field	Description
asr	provider	Provider name (e.g., aliyun, openai, tencent).
tts	provider	Provider name.
llm	provider	LLM Provider.
	model	Model name (e.g., gpt-4, qwen-plus).
dtmf	digit	Mapping of keys (0-9, *, #) to actions.
	action	goto (scene), transfer (SIP), or hangup.
interruption	strategy	both, vad, asr, or none.
	fillerWordFilter	Enable filler word filtering (true/false).
vad	provider	VAD provider (e.g., silero).
realtime	provider	openai or azure. Enable low-latency streaming pipeline.
	model	Specific realtime model (e.g., gpt-4o-realtime).
ambiance	path	Path to background audio file.
	duckLevel	Volume level when agent is speaking (0.0-1.0).
followup	timeout	Silence timeout in ms before triggering follow-up.
	max	Maximum number of follow-up attempts.
recorder	recorderFile	Path template for recording (e.g., call_{id}.wav).
denoise	-	Enable/Disable noise suppression (true/false).
greeting	-	Initial greeting message.

Docker Usage

Pull the Image

docker pull ghcr.io/restsend/active-call:latest

Configuration

Copy the example config and customize it:

cp active-call.example.toml config.toml

Run with Docker

docker run -d \
  --name active-call \
  -p 8080:8080 \
  -p 13050:13050/udp \
  -v $(pwd)/config.toml:/app/config.toml:ro \
  -v $(pwd)/config:/app/config \
  ghcr.io/restsend/active-call:latest

CLI Options

You can override configuration with CLI parameters:

docker run -d \
  --name active-call \
  -p 8080:8080 \
  -p 13050:13050/udp \
  -v $(pwd)/config:/app/config \
  ghcr.io/restsend/active-call:latest \
  --http 0.0.0.0:8080 \
  --sip 0.0.0.0:13050 \
  --handler https://api.example.com/webhook

Supported CLI options:

• --conf <path>: Path to config file
• --http <addr:port>: HTTP server address
• --sip <addr:port>: SIP server address
• --handler <url|file.md>: Quick handler setup (webhook URL or playbook file)

Environment Variables

If you have API keys, save them in an .env file:

TENCENT_APPID=your_app_id
TENCENT_SECRET_ID=your_secret_id
TENCENT_SECRET_KEY=your_secret_key
DASHSCOPE_API_KEY=your_dashscope_api_key

And mount it in the container like :

  -v $(pwd)/.env:/app/.env \

Port Range

Use small range 20000-20100 for local development, bigger range like 20000-30000, or host network for production.

License

This project is licensed under the MIT License.