1. Introduction

Telepath is the Real-Time Voice AI Infrastructure that bridges the gap between traditional telephony (PSTN/SIP) and modern AI Agents. We provide a Zero-Latency websocket bridge, allowing you to connect any SIP carrier (Twilio, Telnyx, etc.) directly to OpenAI, ElevenLabs, or your custom AI backend with millisecond precision.

Signal Flow

1. PSTN/Carrier (Twilio/Telnyx) sends audio via SIP to Telepath.
2. Telepath acts as a Relay, converting SIP/RTP into a WebSocket stream.
3. AI Agent (OpenAI/Custom) processes the stream and responds.
4. Telepath streams the response back to the caller in real-time.

2. Quick Start

Prerequisites

• A SIP Trunking provider (Twilio, Telnyx, Vonage, Bandwidth, SignalWire, Plivo, etc.)
• An API Key from an AI provider (OpenAI, ElevenLabs, etc.)

Step 1: Create a Connection

1. Log in to the Telepath Dashboard.
2. Click "Create Connection".
3. Enter a SIP Username (e.g., my-sales-bot).
4. Select your AI Provider (e.g., OpenAI) and enter your API Key.
5. Important: Copy the generated SIP Password immediately. You will not be able to see it again.

Step 2: Point Your Carrier to Telepath

Configure your carrier's **Origination SIP URI** to point to your new connection:

sip:{sip_username}@sip.telepath.live

Step 3: Make a Test Call

Call your carrier's phone number. The call will route through Telepath to your AI agent. You should hear the AI respond instantly.

3. Creating a Connection

A "Connection" (or Trunk) links a specific SIP Username to an AI Backend configuration.

Required Fields

• SIP Username: unique identifier (3-30 characters) for the SIP address.
• AI Provider: The service handling the logic (OpenAI, ElevenLabs, Custom).
• Provider Credentials: Your private API Key (stored with AES-256 encryption).

Optional Fields

• Inbound Numbers: List of E.164 phone numbers (e.g., +14155550100) allowed to use this trunk.
• Status Webhook URL: An HTTP endpoint where Telepath sends call lifecycle events (POST).
• Webhook Secret: A secret key to sign webhook requests for security.

Integration URI

Once created, your connection is accessible at:

sip:{sip_username}@sip.telepath.live

4. AI Voice Agent Provider

5. Carrier Configuration

Telepath is carrier-agnostic. You can use any provider that supports SIP Sending (Origination).

Supported Carriers

• Twilio: Search for "Elastic SIP Trunking" -> "Origination".
• Telnyx: "SIP Connections" -> "Outbound Options".
• Vonage: "SIP Trunking".
• Bandwidth: "Unified Communications".
• SignalWire: "Domain App".
• Plivo: "SIP Trunking".

Note: Ensure your carrier supports G.711 (PCMU/PCMA) or G.722 codecs.

6. Dashboard

Call Logs

The Logs view provides a real-time audit trail of every call.

• ID: Unique Call UUID.
• Billable Duration: The talk time (Answer to Hangup), rounded up for billing.
• Start Time: When the call was initialized (SIP INVITE).
• Answer Time: When the AI actually picked up.
• Blame: System analysis of who caused a disconnect (e.g., AI Agent, Caller, Carrier).
• Disconnect Explanation: Human-readable reason for the call end.

Analytics

View aggregate metrics like Average Latency, Concurrent Calls, and Call Volume over the last 24-72 hours.

7. Webhooks

Configuration

Set the Status Webhook URL in your Connection settings to receive POST events when calls end.

Behavior

• Trigger: Fires when a call ends (Completed, Failed, Busy).
• Retry: Fire-and-forget (no retries).
• Timeout: 5s.

Payload Format

{
  "sip_call_id": "a1b2c3d4-...",
  "timestamp": "2025-01-15T10:30:00Z",
  "direction": "inbound",
  "from_number": "+15550100",
  "to_number": "+15550200",
  "carrier": "twilio",
  "status": "completed",
  "disconnect_reason": "hangup",
  "disconnect_explanation": "User hung up",
  "blame": "caller",
  "duration": 45.2,
  "billable_duration": 46.0,
  "ai_latency": 450,
  "jitter_max": 12.5
}

8. Codec & VAD

Supported Codecs

Telepath supports high-definition audio transcoding.

• G.711 PCMU (u-law): Standard US telephony.
• G.711 PCMA (a-law): Standard EU telephony.
• G.722: HD Voice (Wideband audio) for superior AI transcription accuracy.

Voice Activity Detection (VAD)

Telepath implements an adaptive audio system that works in tandem with your Voice AI Agent's native Voice Activity Detection.

• Natural Interruptions: Seamless barge-in handling allowing users to speak over the AI naturally.
• Silence Detection: Adaptive sensitivity to ensure accurate end-of-turn detection in various environments.

9. Security

Encryption

• Credentials: All API Keys (OpenAI, etc.) are encrypted at rest using AES-256.
• Media: We support standard RTP.
• Data Retention: Audio is ephemeral and never stored. Logs are retained for analytics only.

SIP Password Rotation

If you believe a SIP password has been compromised, use the Rotate Password feature in the Dashboard or API to immediately generate a new one and revoke the old one.

10. Troubleshooting

Disconnect Reasons

• BYE: Normal hangup.
• 486 Busy: Agent or number busy.
• 503 Service Unavailable: Upstream provider outage.
• 401 Unauthorized: Invalid SIP Credentials.

11. Advanced Integration

DTMF (Keypad Input)

Telepath supports RFC 4733 (formerly RFC 2833) for DTMF signaling.

• Behavior: When a caller presses a key, Telepath converts the RTP event into a JSON message sent to your AI Agent's WebSocket.
• Payload: {"type": "dtmf", "digit": "1"}

Custom SIP Headers (Context Passing)

You can pass context from your carrier (e.g., User ID, Campaign ID) directly to your AI Agent.

1. Carrier Side: Add a SIP Header starting with X- (e.g., X-Customer-ID: 12345).
2. Telepath: Extracts all X- headers from the INVITE.
3. AI Side: Receives these headers in the initial connection payload.

Standard SIP Headers

Telepath automatically forwards critical SIP metadata to the AI Agent context:

• X-SIP-Call-ID: The carrier's unique Call ID.
• From: The caller's number (E.164).
• To: The called number (E.164).