feat(realtime): Add Voice Pipeline Orchestration for gpt-realtime

Implements comprehensive voice pipeline orchestration for OpenAI's Realtime API:

- Voice Pipeline class for managing TTS/STT orchestration with gpt-realtime
- Support for Marin and Cedar realtime voices
- Whisper STT integration for speech-to-text
- WebRTC support for ultra-low latency (<100ms)
- Voice Activity Detection (VAD) capabilities
- Audio processing with configurable settings
- Metrics monitoring for pipeline performance
- Plugin system for easy RealtimeSession integration

The Voice Pipeline provides a framework for building voice-enabled applications
using OpenAI's Realtime API, handling the complexity of audio streaming,
transcription, and synthesis while maintaining low latency.

Features:
- Seamless integration with RealtimeSession
- Configurable audio processing (sample rate, encoding, buffer sizes)
- Real-time metrics (STT/TTS latency, processing time)
- WebRTC support for browser-based voice applications
- Event-driven architecture for audio and speech events

Author: Joe-Simo

docs/src/content/docs/guides/voice-pipeline.mdx

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+# Voice Pipeline Orchestration Example
+
+This example demonstrates the Voice Pipeline Orchestration feature for OpenAI's gpt-realtime model, providing seamless TTS/STT capabilities.
+
+## Features Demonstrated
+
+- **gpt-realtime Integration**: Native support for OpenAI's realtime model
+- **Realtime Voices**: Marin and Cedar voice options
+- **Whisper STT**: High-quality speech recognition
+- **WebRTC Support**: Ultra-low latency (<100ms) voice streaming
+- **Voice Activity Detection**: Automatic speech detection
+- **Audio Enhancement**: Echo/noise suppression and gain control
+- **Metrics Monitoring**: Track pipeline performance
+
+## Prerequisites
+
+1. OpenAI API key with access to:
+   - gpt-realtime model
+   - Whisper (speech-to-text)
+   - Realtime voices (Marin, Cedar)
+
+## Setup
+
+```bash
+# Install dependencies
+pnpm install
+
+# Set environment variables
+export OPENAI_API_KEY="your-api-key"
+```
+
+## Running the Example
+
+```bash
+# Run the example
+pnpm start
+
+# Run in development mode with auto-reload
+pnpm dev
+```
+
+## What It Does
+
+1. **Initializes Voice Pipeline**: Sets up gpt-realtime with Whisper STT
+2. **Demonstrates Voice Switching**: Shows switching between Marin and Cedar voices
+3. **Simulates Conversation**: Processes sample voice interactions
+4. **Shows Tool Usage**: Weather, calculator, and timer tools
+5. **Monitors Metrics**: Displays latency and performance metrics
+6. **WebRTC Mode**: Optional ultra-low latency configuration
+
+## Key Components
+
+### gpt-realtime Model
+
+The cutting-edge realtime model providing natural voice interactions with minimal latency.
+
+### Realtime Voices
+
+- **Marin**: Optimized for clarity and professional tone
+- **Cedar**: Warm and friendly for conversational interactions
+
+### Whisper STT
+
+OpenAI's state-of-the-art speech recognition for accurate transcription.
+
+### WebRTC Integration
+
+Enables ultra-low latency (<100ms) for real-time conversations.
+
+## Architecture
+
+```
+User Audio → Whisper STT → gpt-realtime → Realtime Voice → Audio Output
+     ↑                            ↓
+     └─── Voice Activity ←────────┘
+          Detection (VAD)
+```
+
+## Configuration Options
+
+### Audio Settings
+
+- Sample Rate: 24kHz (optimized for realtime)
+- Encoding: PCM16 or Opus (for WebRTC)
+- Channels: Mono
+
+### Voice Activity Detection
+
+- Threshold: 0.5 (adjustable sensitivity)
+- Max Silence: 2000ms
+- Debounce: 300ms
+
+### WebRTC Settings
+
+- ICE Servers: STUN for NAT traversal
+- Audio Constraints: Echo/noise suppression
+- Target Latency: <100ms
+
+## Customization
+
+Edit `voice-pipeline-example.ts` to:
+
+- Adjust voice settings (Marin/Cedar)
+- Modify VAD parameters
+- Add custom tools
+- Change audio configuration
+- Enable/disable WebRTC mode
+
+## Production Considerations
+
+1. **API Keys**: Store securely, never commit to version control
+2. **Error Handling**: Implement robust error recovery
+3. **Latency**: Use WebRTC for lowest latency requirements
+4. **Audio Quality**: Balance quality vs bandwidth based on use case
+5. **Rate Limiting**: Monitor API usage and implement appropriate limits
+
+## Troubleshooting
+
+### High Latency
+
+- Enable WebRTC mode for ultra-low latency
+- Check network connection quality
+- Optimize audio buffer sizes
+
+### Audio Quality Issues
+
+- Adjust VAD threshold for your environment
+- Enable noise suppression
+- Check microphone quality
+
+### Connection Issues
+
+- Verify API key has necessary permissions
+- Check firewall settings for WebRTC
+- Ensure stable internet connection
+
+## Related Resources
+
+- [Voice Agents Guide](../../docs/src/content/docs/guides/voice-agents)
+- [Realtime API Documentation](https://platform.openai.com/docs/guides/realtime)
+- [OpenAI Agents SDK Documentation](../../docs)

examples/voice-pipeline/README.md

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+{
+  "name": "voice-pipeline-example",
+  "version": "0.1.0",
+  "description": "Voice Pipeline Orchestration example for OpenAI Agents SDK",
+  "main": "voice-pipeline-example.ts",
+  "scripts": {
+    "start": "tsx voice-pipeline-example.ts",
+    "dev": "tsx watch voice-pipeline-example.ts"
+  },
+  "dependencies": {
+    "@openai/agents": "workspace:*",
+    "openai": "^4.79.1"
+  },
+  "devDependencies": {
+    "@types/node": "^22.10.5",
+    "tsx": "^4.19.2",
+    "typescript": "^5.7.2"
+  }
+}