Custom VoIP Integration

TLS (SRTP) Encryption

Secures your VoIP traffic by encrypting both the signaling (via TLS) and media (via SRTP). Activate this toggle only if your VoIP provider requires it.

When to Enable It:

• Your SIP provider specifically mentions TLS or SRTP support.
• You want to ensure secure, encrypted voice communication (e.g., for compliance or privacy reasons).

When to Leave It Off:

• Your provider doesn’t support TLS/SRTP — enabling it may block calls or cause failed connections.
• You’re using basic test environments or local SIP servers without encryption.

Authentication Methods

Choose one of the following methods to authorize your outbound VoIP traffic:IP Address Authentication:Allow traffic from specific IP addresses — ideal if your VoIP infrastructure has a static IP.IP Address: Enter the public IP address(es) of your SIP servers so we can accept and process requests from them.

Credentials Authentication (Optional):Use a username and password to authenticate each outbound SIP request.Username: Alphanumeric string (max 32 characters)
Password: At least 12 characters, with 1 uppercase, 1 lowercase, 1 digit, and 1 special character.

This method is useful if your infrastructure is cloud-based and IPs can change, or you want multi-tenant control.

Once you save the inbound configuration, you will receive a SIP Domain to connect. You will need to add this SIP Domain to your VoIP configuration so it can transfer the call to our platform. If you choose Credentials Authentication, make sure to also include the username and password you select in your VoIP settings.

TLS (SRTP) Encryption

Secures your VoIP traffic by encrypting both the signaling (via TLS) and media (via SRTP). Activate this toggle only if your VoIP provider requires it.

When to Enable It:

• Your SIP provider specifically mentions TLS or SRTP support.
• You want to ensure secure, encrypted voice communication (e.g., for compliance or privacy reasons).

When to Leave It Off:

• Your provider doesn’t support TLS/SRTP — enabling it may block calls or cause failed connections.
• You’re using basic test environments or local SIP servers without encryption.

SIP Trunk URL

This is the address your outbound calls will be sent to — typically provided by your VoIP provider. Enter the SIP trunk URL where your calls should be routed.

User Part

This is the SIP user identity used in the “From” header of your outbound SIP requests. It’s how your VoIP system identifies the origin of the call.What to enter: You can use one of the following formats depending on your provider:

• A SIP username (e.g. caller01)
• A DID or phone number in E.164 format (e.g. +1234567890)
• An extension (e.g. 201)
• A SIP URI (e.g. caller01@example.com) — less common

Data Center

Select the geographic location (network edge) closest to your VoIP infrastructure to optimize call routing and reduce latency.

Transfer Call Using SIP Refer

This toggle enables or disables call transfers using the SIP REFER method.When enabled, it allows:

• Transferring a live call from one destination to another without media proxying
• Offloading media streams directly between endpoints
• Reducing latency and server load

Authentication Methods (optional based on your VoIP)

Credentials Authentication:Use this method if your VoIP provider requires username/password credentials to authenticate outbound calls.Username: This is the identifier associated with your SIP account.Password: A secure password used to authenticate the SIP user. Make sure it meets your provider’s security requirements (e.g. minimum 12 characters, including upper/lowercase letters, a number, and a special character).

Header-Based Authentication:Use this method if your VoIP provider requires custom headers for SIP request authentication.Header Key: This is the name of the custom SIP header field expected by your provider (e.g., X-Custom-Auth, X-Access-Token, etc.).Header Value: The corresponding value used to validate the request (e.g., a token, API key, or static secret). Example: AuthZ-87!t!link

Header-based auth is typically used for advanced integrations or when SIP gateways are behind proxies/firewalls that don’t support IP or credential-based auth. Ensure this matches exactly what your provider expects, as even small typos (like case or symbols) can lead to rejections.

You can add several key-value pairs if needed — for example, to meet multi-token security policies.

• IP Address Allow List:
  IP Address: See Twilio’s SIP Trunking IP Addresses.