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SIP trunking

Connect your existing phone system with ElevenLabs conversational AI agents using SIP trunking

Overview

SIP (Session Initiation Protocol) trunking allows you to connect your existing telephony infrastructure directly to ElevenLabs conversational AI agents. This integration enables enterprise customers to use their existing phone systems while leveraging ElevenLabs’ advanced voice AI capabilities.

With SIP trunking, you can:

  • Connect your Private Branch Exchange (PBX) or SIP-enabled phone system to ElevenLabs’ voice AI platform
  • Route calls to AI agents without changing your existing phone infrastructure
  • Handle both inbound and outbound calls

How SIP trunking works

SIP trunking establishes a direct connection between your telephony infrastructure and the ElevenLabs platform:

  1. Inbound calls: Calls from your SIP trunk are routed to the ElevenLabs platform using our origination URI.
  2. Outbound calls: Calls initiated by ElevenLabs are routed to your SIP trunk using your termination URI, enabling your agents to make outgoing calls.
  3. Authentication: Connection security for the signaling is maintained through either digest authentication (username/password) or Access Control List (ACL) authentication based on the signaling source IP.
  4. Signaling and Media: The initial call setup (signaling) uses TCP. Once the call is established, the actual audio data (RTP stream) is transmitted over UDP for low latency.

Requirements

Before setting up SIP trunking, ensure you have:

  1. A SIP-compatible PBX or telephony system
  2. Phone numbers that you want to connect to ElevenLabs
  3. Administrator access to your SIP trunk configuration
  4. Appropriate firewall settings to allow SIP traffic
  5. Audio codec compatibility: Your system must support 48kHz audio or be capable of resampling audio on your end, as ElevenLabs’ SIP deployment outputs and receives audio at this sample rate. This is independent of any audio format configured on the agent for direct websocket connections.

Setting up SIP trunking

2

Import SIP Trunk

Click on “Import a phone number from SIP trunk” button to open the configuration dialog.

Select SIP trunk option
SIP trunk configuration dialog

When you import a SIP trunk, the system automatically configures the ElevenLabs origination URI for inbound calls: sip:sip.rtc.elevenlabs.io:5060;transport=tcp

This pre-populated URI cannot be modified and serves as the destination endpoint where your system should route all inbound calls (calls from your system to ElevenLabs).

The transport=tcp parameter specifies that the initial SIP signaling connection uses TCP. However, the actual audio data (RTP stream) is transmitted over UDP for low-latency performance, which is standard for real-time communication. This distinction is important: only the call setup uses TCP; the voice data itself uses UDP.

3

Enter configuration details

Complete the form with the following information:

  • Label: A descriptive name for the phone number
  • Phone Number: The E.164 formatted phone number to connect (e.g., +15551234567)
  • Termination URI: Your SIP trunk’s termination URI (where ElevenLabs will send outbound calls)
SIP trunk inbound configuration
4

Configure authentication (optional)

If your SIP provider requires digest authentication:

  • Enter the username for SIP digest authentication
  • Enter the password for SIP digest authentication
SIP trunk outbound configuration

If left empty, Access Control List (ACL) authentication based on the source IP address will be attempted. In this case, you’ll need to ensure your system sends SIP requests from an IP address that you allowlist in your provider’s settings, and that your firewall allows return traffic from ElevenLabs.

Signaling vs. Media IPs:

  • The SIP signaling connection originates from sip.rtc.elevenlabs.io (currently resolving to 34.49.132.122). This IP is static and can be allowlisted if using ACL authentication.
  • The RTP media stream (audio data) uses dynamic IP addresses from an autoscaling pool. These IPs change and cannot be reliably allowlisted.

For robust security without relying on IP allowlisting for media, Digest Authentication is strongly recommended.

5

Complete Setup

Click “Import” to finalize the configuration.

Assigning Agents to Phone Numbers

After importing your SIP trunk phone number, you can assign it to a conversational AI agent:

  1. Go to the Phone Numbers section in the Conversational AI dashboard
  2. Select your imported SIP trunk phone number
  3. Click “Assign Agent”
  4. Select the agent you want to handle calls to this number

Troubleshooting

If you’re experiencing connection problems:

  1. Verify your SIP trunk configuration on both the ElevenLabs side and your provider side.
  2. Check that your firewall allows SIP signaling traffic (TCP/UDP port 5060) to sip.rtc.elevenlabs.io (34.49.132.122) and allows RTP traffic
  3. Confirm that your termination URI is correctly formatted.
  4. Test with and without digest authentication credentials.

If calls are failing due to authentication issues:

  1. Double-check your username and password if using digest authentication.
  2. If using ACL authentication, ensure your provider is configured to trust signaling traffic from sip.rtc.elevenlabs.io (34.49.132.122). Remember that ACL does not apply to the media stream IPs.
  3. Check your SIP trunk provider’s logs for specific authenticatio nerror messages.

If the call connects but there’s no audio or audio only flows one way:

  1. Verify that your firewall allows UDP traffic for the RTP media stream (typically ports 10000-60000). Since these IPs change, ensure the rule is not restricted to specific static IPs.
  2. Check for Network Address Translation (NAT) issues that might be blocking the RTP stream.

If you experience poor audio quality:

  1. Ensure your network has sufficient bandwidth (at least 100 Kbps per call) and low latency/jitter for UDP traffic.
  2. Check for network congestion or packet loss, particularly on the UDP path.
  3. Verify codec settings match on both ends.

Limitations and Considerations

  • Support for multiple concurrent calls depends on your subscription tier
  • Call recording and analytics features are available but may require additional configuration
  • Outbound calling capabilities may be limited by your SIP trunk provider
  • Audio format: ElevenLabs’ SIP deployment outputs and receives audio at 48kHz sample rate. This is independent of any audio format configured on the agent for direct websocket connections. Your SIP trunk system must either support this format natively or perform resampling to match your system’s requirements

FAQ

Yes, SIP trunking allows you to connect your existing phone numbers directly to ElevenLabs’ conversational AI platform without porting them.

ElevenLabs is compatible with most standard SIP trunk providers including Twilio, Vonage, RingCentral, Sinch, Infobip, Telnyx, Exotel, Plivo, Bandwidth, and others that support SIP protocol standards.

The number of concurrent calls depends on your subscription plan. Enterprise plans typically allow for higher volumes of concurrent calls.

Yes, ElevenLabs supports encrypted SIP communications (SIPS) for enhanced security. Contact support for specific configuration requirements.

Yes, you can use your existing PBX system’s routing rules to direct calls to different phone numbers, each connected to different ElevenLabs agents.

Next steps