SIP Trunk Import

If you already have phone numbers with a carrier, you can connect them to ScaleTalk via SIP trunking without porting.

What is SIP Trunking?

SIP (Session Initiation Protocol) trunking allows you to route calls from your existing carrier to ScaleTalk's voice AI infrastructure without changing your phone numbers.

Benefits

• Keep Your Numbers - No need to port numbers
• Carrier Flexibility - Use any SIP-compatible carrier
• Existing Contracts - Maintain your current agreements
• Failover Options - Configure backup routing

Supported Carriers

ScaleTalk works with any carrier that supports standard SIP:

• Twilio - Native integration
• Telnyx - Fully tested
• Vonage - Supported
• Bandwidth - Supported
• Custom Carriers - Contact support

Setup Process

Step 1: Get SIP Credentials

Contact your carrier to obtain:

• SIP Server Address (e.g., sip.carrier.com)
• Username (authentication)
• Password (authentication)
• Port (usually 5060 for UDP/TCP, 5061 for TLS)

Step 2: Configure in ScaleTalk

1. Navigate to Phone Numbers
2. Click Import SIP Number
3. Enter your credentials:

4. Click Test Connection
5. If successful, click Save

Step 3: Configure Carrier Routing

On your carrier's side, configure:

• Forwarding Destination: ScaleTalk's SIP address
• Authentication: Credentials provided by ScaleTalk
• Failover: Optional backup routing

Security Best Practices

Use TLS Transport

Always use TLS (Transport Layer Security) for SIP:

• Encrypts signaling data
• Prevents eavesdropping
• Required for compliance

Authentication Methods

Media Encryption (SRTP)

Enable SRTP for encrypted audio:

1. Enable Secure Media in your SIP configuration
2. Ensure your carrier supports SRTP
3. Test call quality after enabling

Codec Configuration

ScaleTalk supports standard audio codecs:

Recommended Codec Order

1. Opus (if supported)
2. G.711 PCMU (µ-law)
3. G.711 PCMA (A-law)
4. G.729

Troubleshooting

Connection Test Failed

1. Verify SIP server address is correct
2. Check username and password
3. Ensure port is not blocked by firewall
4. Try different transport (UDP vs TLS)

Calls Not Connecting

1. Verify carrier is forwarding to correct destination
2. Check authentication credentials
3. Review carrier logs for errors
4. Contact ScaleTalk support with carrier logs

Audio Quality Issues

1. Check codec negotiation
2. Verify bandwidth is sufficient
3. Enable jitter buffer on carrier
4. Test with different codec

One-Way Audio

1. Check NAT configuration on carrier
2. Verify media IP is reachable
3. Enable symmetric RTP if available
4. Review firewall rules for media ports

Advanced Configuration

Multiple Numbers

You can import multiple numbers from the same SIP trunk:

1. Create one SIP configuration
2. Add each phone number
3. Assign different agents to each

Failover Configuration

Configure backup routing:

1. Primary: ScaleTalk
2. Secondary: Voicemail or another destination

Outbound Caller ID

When making outbound calls:

• Use imported numbers as caller ID
• Requires carrier configuration
• May have regional restrictions