SIP Trunking

What is SIP Trunking?

SIP trunking connects your VoIP system or AI voice agent directly to CloudVNO's carrier network. You get carrier-grade PSTN access with ultra-low latency — ideal for AI voice agents, contact centers, and custom phone systems.

SIP Trunk Configuration

Your SIP trunk credentials are in the dashboard under Voice → SIP Trunks.

Setting	Value
SIP Domain	sip.cloudvno.com
Port	5060 (UDP/TCP) or 5061 (TLS)
Authentication	Digest (username/password) or IP allowlist
Transport	UDP, TCP, or TLS
SRTP	Optional (recommended for encryption)

Connect an AI Voice Agent (Vapi/Retell)

Vapi Configuration

1. In Vapi dashboard, go to Phone Numbers → Import
2. Select SIP Trunk
3. Enter your CloudVNO SIP credentials
4. Assign a CloudVNO DID number to the trunk

Retell AI Configuration

# Provision a number and assign to Retell via CloudVNO API
number = client.numbers.provision(
    country="US",
    type="local",
    area_code="415"
)

# Configure SIP forwarding to Retell
client.incoming_phone_numbers(number.phone_number).update(
    voice_url="sip:your-retell-agent@sip.retellai.com"
)

WebSocket SIP

For AI voice agents that need real-time audio streaming, CloudVNO supports WebSocket SIP:

wss://sip.cloudvno.com:8443

This provides sub-50ms media latency required for natural-sounding AI conversations.

STIR/SHAKEN Attestation

All outbound calls via SIP trunk automatically receive STIR/SHAKEN attestation based on your account's KYC status:

• Full (A): For numbers you own and control
• Partial (B): For numbers provisioned on behalf of customers
• Gateway (C): For calls transiting through your network

Contact support to upgrade your attestation level.