🌐 Connect Your Phone Network to a 24/7 Conversational AI Agent

Connect your compatible phone network — such as Twilio, Telnyx, AVOXI, Movistar, Claro, AT&T, and more — to a 24/7 Conversational AI Agent.

DO IT YOURSELF: USD $0
WE CONNECT IT FOR YOU: USD $336

🧩 Step-by-Step Guide

Requirements for Connecting a 24/7 Conversational AI Agent with SIP Trunking / Telephony Integration

✅ Main Requirements

SIP-Compatible PBX or VoIP Infrastructure

You must have a system or provider capable of sending incoming calls as SIP INVITE requests to the Conversational AI Agent.

Examples: AVOXI, a VoIP provider, or your own PBX.

Phone Numbers to Connect

Your existing phone numbers (or DIDs) that receive calls must be configured to route traffic to your SIP trunk.

Administrative Access to SIP Trunk / Configuration

You need the ability to define:

• Destination URI

• Credentials

• Routes

• Transport protocols, etc.

Firewall / Network / Ports

Allow SIP traffic to and from the Conversational AI Agent servers.

Open:

• TCP/TLS ports 5060/5061 for signaling

• UDP RTP port range for audio streaming
  Ensure proper NAT traversal and routing of RTP packets.

TLS Transport and Media Encryption (Optional but Recommended)

• Use SIP over TLS for encrypted signaling.

• Enable SRTP for encrypted audio streams.

• The AI Agent supports media encryption modes: Required / Allowed / Disabled.

• For production, valid TLS certificates are recommended.

Audio Codec Compatibility

The AI Conversational Agent expects your VoIP trunk to support G.711 (μ-law or A-law) or G.722.
If your provider uses another codec (e.g. G.729), transcoding is required.

Authentication / Access Control

Two methods are supported:

1. Digest Authentication (username/password)

2. ACL (allowlist by IP)

If using ACL, whitelist the IP ranges used by the AI Conversational Agent for signaling.
Enterprise accounts offer static IP blocks for firewall whitelisting.

Import / Register the SIP Trunk in the Conversational AI Platform

In the Conversational AI Agent dashboard:

Phone Numbers → Import a phone number from SIP trunk.

Configure:

• Label (descriptive name)

• Number in E.164 format (e.g., +598...)

• Transport (TCP / TLS)

• Termination address (hostname or IP, no sip: prefix)

• Media encryption (enabled or disabled)

• Custom headers (if required by your provider)

• Credentials (if using Digest Authentication)

Assign the imported SIP number to your conversational agent.

Optional SIP Headers / Metadata

To improve traceability, the AI Agent recommends adding custom headers to SIP traffic:

• X-CALL-ID

• X-CALLER-ID

These help associate the phone call with the conversational context.

Limitations and Considerations

• The number of simultaneous calls depends on your AI Agent plan.

• Call recording, metrics, or logs may require extra setup.

• SIP signaling uses TCP/TLS, not UDP.

• Media (voice) is transmitted via RTP over UDP.

• Proper NAT and port translation are crucial.

• If using “media encryption: Required,” both ends must support SRTP.

☎️ Telephony Provider SIP Forwarding Requirements

Most telecom companies offer SIP forwarding / SIP URI forwarding to redirect incoming calls to an external SIP endpoint.

✅ What to Request or Confirm with Your Telephony Provider

SIP URI / Forwarding Configuration

In your telecom provider portal, assign a destination SIP URI for your number (or routing rule):
user@my_sip_server.com:5060 or my_sip_server.com:port

The provider must let you select the protocol (UDP, TCP, or TLS).

Protocols (UDP / TCP / TLS)

• Providers usually support forwarding over UDP, TCP, and TLS.

• If using TLS, ensure valid TLS certificates can be loaded or used (e.g. with AVOXI).

RTP (Audio) Port Ranges

Typical default RTP port range: 16384–25999 UDP
Allow this range in your firewall.

Signaling Ports

• SIP over UDP/TCP: port 5060

• SIP over TLS: port 5061 (or provider-specific)

Supported Codecs

Example:
AVOXI supports G.711 μ-law (PCMU) and G.711 A-law (PCMA).
If your AI Agent uses G.722, you’ll need transcoding or provider support.

DTMF (Tones)

The provider must support DTMF via RFC 4733 / RFC 2833.

Telecom Provider IP Addresses

Your provider will list designated IPs to allow through your firewall for inbound traffic.
Example: 104.196.177.56, 35.231.194.250, etc.
These must be whitelisted for proper call forwarding.

SIP URI Testing / Call Tests

Your provider should offer “SIP URI Testing” to validate routes.
Make test calls from your phone number to confirm the SIP endpoint receives them correctly.

TLS / Certificates

If you use TLS/SRTP:

• Manage valid certificates for your SIP server.

• Download the provider’s TLS certificate if needed for validation.

🏗️ Concrete Steps to Connect (AVOXI / Movistar / Claro / AT&T, etc.)

1. Configure your number with SIP forwarding to:
   sip:your_endpoint@host:port
   (This endpoint can be your SIP server or the AI Conversational Agent if it supports SIP directly.)

2. Choose TLS for encryption (or TCP/UDP as fallback).

3. Open ports 5060/5061 (SIP) and RTP 16384–25999 (UDP) in your firewall.

4. On the AI Agent side, import the SIP trunk with credentials provided by your telecom.

5. Set the inbound URI and transport (TCP/TLS).

6. Select audio codecs (G.711 / G.722).

7. Configure authentication (Digest / ACL).

8. Assign the SIP number to your AI agent.

9. Test real inbound calls and verify SIP signaling (INVITE, ACK, BYE) and RTP streams.

📌 Critical Requirements Checklist

• Supported audio codec (G.711 / G.722) or transcoding

• Open SIP ports (5060/5061) and RTP UDP range

• TLS / SRTP support (optional, recommended)

• Administrative control over SIP URI configuration (e.g., AVOXI)

• Stable and reachable SIP endpoint (hostname/IP)

• Authentication setup (Digest or ACL)

• Proper firewall / NAT handling

• Whitelist telecom and AI Agent IP addresses

• Import trunk into the AI platform and assign to the agent

Would you like me to turn this into a professional bilingual (EN/ES) technical setup guide in PDF format — ready to send to telecoms or IT teams (with your logo “LIVE PLANETA IA”)?
It would look like a branded “Integration Manual: AI Agent ↔ SIP Trunk / Telephony.”