SIP Trunk Service
Manage SIP trunks for enterprise voice communications. Create trunks with automatic SIP domain provisioning, configure authentication methods, and set up outbound routing for scalable telephony infrastructure.
What is a SIP Trunk?
A SIP trunk is a virtual connection that enables voice communication over IP networks. Each trunk serves as a dedicated voice gateway with its own authentication, rate limits, and routing configuration. Vobiz automatically provisions each trunk with a unique SIP domain and integrates it with Kamailio for robust SIP routing and authentication.
- Auto-generated SIP domains: Each trunk gets a unique domain like
trunkId.sip.vobiz.ai - Flexible authentication: Support for both username/password and IP-based authentication
- Rate limiting: Configure concurrent call limits and calls-per-second (CPS) throttling
- Outbound routing: Define origination URIs with priority-based failover and load balancing
Key Features
Dual Authentication
Choose between username/password credentials or IP whitelisting. Combine both methods for maximum security.
Intelligent Routing
Configure multiple origination URIs with priority-based failover and weight-based load balancing for resilient outbound calling.
Rate Limiting
Protect against traffic spikes with configurable concurrent call limits and calls-per-second (CPS) throttling.
Kamailio Integration
Seamless integration with Kamailio SIP proxy for robust routing, authentication, and load balancing infrastructure.
Trunk Management
The core trunk operations allow you to create, configure, and manage your SIP trunks. Each trunk acts as an independent voice gateway with its own configuration and rate limits.
The Trunk Object
Complete structure and attributes of a trunk
Create a Trunk
Provision a new SIP trunk with auto-generated domain
Retrieve a Trunk
Get details of a specific trunk by ID
Retrieve All Trunks
List all trunks with pagination support
Update a Trunk
Modify trunk configuration and rate limits
Delete a Trunk
Permanently remove a trunk and its resources
Credentials
Credentials provide username/password authentication for your SIP trunk. You can create multiple credentials per trunk for different devices or use cases. Passwords are securely hashed and never returned in API responses after creation.
View Credentials DocumentationIP Access Control Lists
IP Access Control Lists (IP ACLs) enable IP-based authentication by whitelisting specific IPv4 addresses. Devices calling from whitelisted IPs can use the trunk without password authentication. This is ideal for PBX systems, SIP gateways, and carrier interconnections with static IP addresses.
View IP ACL DocumentationOrigination URIs
Origination URIs define where outbound calls from your trunk should be routed. Configure multiple URIs with priority-based failover (lower priority tried first) and weight-based load balancing (higher weight receives more traffic). This enables resilient, distributed call routing with automatic failover.
View Origination URI DocumentationWebhooks
Configure a webhook URL on your trunk to receive real-time HTTP callbacks for call events. VoBiz sends notifications when a call is initiated (admitted or rejected) and when a call ends (hangup with full duration, cost, and quality metrics). Webhooks are sent asynchronously and never delay the SIP call flow.
View Webhook DocumentationAuthentication Methods
Username/Password
SIP digest authentication using credentials. Ideal for devices with dynamic IP addresses or mobile softphones. Kamailio validates credentials against the subscriber database.
IP Whitelisting
IP-based authentication without passwords. Best for PBX systems, SIP gateways, and carrier connections with static IP addresses. Provides higher security for production deployments.
SIP Call Log Error Statuses
When reviewing your SIP call logs, you may encounter various error and failure statuses. Understanding these statuses is essential for troubleshooting SIP connections, routing issues, and endpoint configurations. The table below describes common SIP error logs and what they indicate:
| Status Code | Description / Cause |
|---|---|
NORMAL_CLEARING | The call was cleared normally by one of the parties hanging up. This is a standard end-of-call status. |
USER_BUSY | The called party is busy (e.g., already on another call or declined the call). |
NO_ANSWER | The called party did not answer the call within the specified timeout period. |
ORIGINATOR_CANCEL | The caller (originator) hung up or cancelled the call before it could be answered. |
CALL_REJECTED | The call was explicitly rejected, potentially due to blocking rules, spam filtering, or the destination refusing the connection. |
REJECTED | A general rejection status, often occurring when the SIP endpoint refuses the incoming INVITE. |
INVALID_NUMBER | The dialed number format is incorrect, invalid, or does not exist. |
UNALLOCATED_NUMBER | The dialed number is formally valid but is not currently allocated or assigned to any active subscriber. |
SERVICE_UNAVAILABLE | The destination service or endpoint is temporarily completely unavailable (e.g., SIP 503 error). |
SERVER_ERROR | An internal server error occurred while attempting to process or route the SIP call. |
MEDIA_TIMEOUT | The call was disconnected because no RTP media packets were received for an extended period, suggesting a firewall or network dropout issue. |
PROTOCOL_ERROR | A fundamental SIP protocol violation or malformed SIP message occurred during call setup. |
NETWORK_OUT_OF_ORDER | A severe network failure prevented the call from being routed to its destination. |
DESTINATION_OUT_OF_ORDER | The specific destination endpoint or PBX cannot physically accept the call, often indicating an offline server. |
NORMAL_TEMPORARY_FAILURE | A generic temporary failure in routing the call. Re-attempting the call might succeed. |
SWITCH_CONGESTION | The telephone network or SIP switch is experiencing high traffic and cannot handle the required capacity at this moment. |
UNKNOWN | An error occurred that could not be mapped to any standard SIP error or ISUP release cause code. |
Getting Started
Quick Setup Guide
- 1
Create a Trunk
Use the Create Trunk endpoint to provision a new SIP trunk. You'll receive a unique SIP domain and trunk ID.
- 2
Configure Authentication
Add credentials for password authentication or configure IP ACLs for IP-based authentication.
- 3
Set Up Routing
Configure origination URIs to define where outbound calls should be routed. Set priorities for failover and weights for load balancing.
- 4
Test Your Trunk
Configure your SIP client with the trunk's domain and credentials, then make a test call to verify everything is working correctly.