How to have Twilio SIP trunk point to locally hosted SIP server via Docker https://docs.livekit.io/sip/

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How can I figure out what my SIP host would be for a self-hosted SIP server? What does it mean to self-host an SIP server and how does it compare with using LiveKit Cloud?

To determine your SIP host for a self-hosted SIP server, it would typically be the public IP address or domain name of the server where your SIP service is hosted. This address must be accessible from the Internet, and the necessary ports (e.g., 5060 for SIP signaling and 10000-20000 for media) should be open. Self-hosting a SIP server means you are responsible for deploying and managing the SIP service infrastructure. This includes setting up the server, ensuring its availability, and maintaining it over time. In contrast, using LiveKit Cloud means that LiveKit manages the infrastructure for you, providing benefits like analytics, telemetry, and a 99.99% uptime guarantee. While both self-hosted and LiveKit Cloud options support SIP (telephony integration), the key differences are: • Open Source (Self-hosted): You manage the infrastructure yourself, and it uses a Single-home SFU architecture. • LiveKit Cloud: LiveKit manages the infrastructure, uses a Mesh SFU architecture, and offers additional features like analytics and telemetry. For more details, you can refer to the documentation: LiveKit SIP Documentation. Sources: LiveKit Cloud | LiveKit Docs | SIP overview | LiveKit Docs

If my SIP service is hosted locally does this mean the SIP host would be my public IP, what would the docker-compose.yaml file look like?

If your SIP service is hosted locally, then yes, the SIP host would typically be your public IP address. Ensure that the SIP signaling port (5060) and media port range (10000-20000) are accessible from the Internet for proper functionality. To set up the SIP service using Docker Compose, you can download a pre-configured

docker-compose.yaml

file from the LiveKit repository using the following commands:

wget <https://raw.githubusercontent.com/livekit/sip/main/docker-compose.yaml>
docker compose up

While the exact content of the

docker-compose.yaml

file is not provided in the sources, you may need to modify it to include your public IP address or domain name if required. Ensure that your network and firewall settings allow external access to the necessary ports. For further details, you can refer to the documentation: LiveKit SIP Documentation. Sources: Self-hosted SIP server | LiveKit Docs

How can I ensure that my network and firewall settings allow external access to the SIP signaling port (5060) and media port range (10000-20000).

To ensure that your network and firewall settings allow external access to the SIP signaling port (5060) and media port range (10000-20000), follow these steps: 1. Verify that your firewall rules allow inbound and outbound traffic on the following ports: ◦ SIP signaling: Port 5060 (for SIP/UDP and SIP/TCP) and optionally 5061 (for SIP/TLS). ◦ Media transmission: Port range 10000-20000 (for SIP RTP/UDP). 1. Ensure that these ports are not blocked by your router or ISP. 2. Update the

config.yaml

file for your SIP server to match these port configurations:

sip_port: 5060
rtp_port: 10000-20000

3. Test the connectivity to these ports from an external network to confirm they are accessible. For more details, you can refer to the documentation: LiveKit SIP Documentation. Sources: Self-hosted SIP server | LiveKit Docs | Ports and Firewall | LiveKit Docs

How can I check this on Ubuntu 1. Verify that your firewall rules allow inbound and outbound traffic on the following ports: ◦ SIP signaling: Port 5060 (for SIP/UDP and SIP/TCP) and optionally 5061 (for SIP/TLS). ◦ Media transmission: Port range 10000-20000 (for SIP RTP/UDP). 2. Ensure that these ports are not blocked by your router or ISP.

To verify that your firewall rules allow inbound and outbound traffic on ports 5060 and 10000-20000 on Ubuntu, you can use the following steps: 1. Check the status of your firewall: `sudo ufw status`This will display the current rules and whether the firewall is active. 2. If the firewall is active, add rules to allow the required ports: `sudo ufw allow 5060/tcp sudo ufw allow 5060/udp sudo ufw allow 10000:20000/udp`These commands will open the SIP signaling port (5060) for both TCP and UDP, and the media port range (10000-20000) for UDP. 3. Reload the firewall to apply the changes:

sudo ufw reload

4. Verify the updated rules: `sudo ufw status`Ensure that the required ports are listed as "ALLOW" in the output. Additionally, ensure that your router or ISP is not blocking these ports. You may need to configure port forwarding on your router if the SIP server is behind NAT.

How can I check that my router or ISP is not blocking these ports. You may need to configure port forwarding on your router if the SIP server is behind NAT.

To check if your router or ISP is blocking the required ports (5060 and 10000-20000), and to configure port forwarding if necessary, follow these steps: 1. Check for blocked ports: ◦ Use an online port checker tool to test if the ports are open. These tools can scan your public IP address for open ports. ◦ Alternatively, you can use the

telnet

command from an external network to test connectivity to your public IP and the specific ports. 1. Configure port forwarding on your router: ◦ Log in to your router's admin interface (usually accessible via a web browser). ◦ Locate the port forwarding section (this may be under "Advanced Settings" or "NAT Settings"). ◦ Add rules to forward traffic on ports 5060 (TCP/UDP) and 10000-20000 (UDP) to the internal IP address of your SIP server. ◦ Save the changes and restart your router if necessary. 1. Contact your ISP: ◦ If the ports remain inaccessible, contact your ISP to confirm that they are not blocking these ports. Some ISPs block certain ports by default for security reasons. After completing these steps, test the connectivity again to ensure the ports are accessible from the Internet.

Do I need to manually set the external_ip to my public ip to connect https://hub.docker.com/r/livekit/sip

For the LiveKit SIP server, you generally do not need to manually set the external IP address. Instead, you can set

use_external_ip

to

true

in the configuration. This allows LiveKit to use STUN to automatically discover your server's public IP address and advertise it to clients. Here is an example configuration snippet:

use_external_ip: true

This setting is recommended for most cloud environments or any setup where the host has a public IP address but is not directly exposed to the process. If you are running the SIP server in an environment where STUN cannot determine the public IP, you may need to manually specify the external IP address. For more details, you can refer to the documentation: LiveKit SIP Docker Hub. Sources: Self-hosted SIP server | LiveKit Docs | Deploying LiveKit | LiveKit Docs

I'm running into this error when trying to connect to my locally hosted SIP server from my Twilio SIP trunk

sip-1      | 2025-03-28T15:16:38.478Z   DEBUG   sip     sip/inbound.go:413      Rejecting inbound flood

The error message

Rejecting inbound flood

indicates that the SIP server is rejecting incoming requests due to a potential flood of traffic. This could happen if the server detects too many requests in a short period of time, which might be interpreted as an attack or misconfiguration. To troubleshoot this issue: 1. Check the configuration of your SIP server to see if there are any rate-limiting or flood protection settings that might be causing this behavior. 2. Verify that your Twilio SIP trunk is correctly configured and not sending an excessive number of requests to the server. 3. Review the server logs for additional details about the rejected requests to identify the source and reason for the flood detection. 4. Ensure that your network and firewall settings are not causing unintended traffic loops or retries. If the issue persists, you may need to adjust the flood protection settings on your SIP server or investigate further into the traffic patterns from your Twilio SIP trunk.

here is the specific information with senstive information removed, the main difficulty comes from source sip-1 | 2025-03-28T151638.478Z DEBUG sip sip/inbound.go:413 Rejecting inbound flood {"nodeID": "node_id_placeholder", "callID": "call_id_placeholder", "fromIP": "public_ip", "toIP": "local_ip:port", "fromHost": "host_placeholder", "fromUser": "phone_number_placeholder", "toHost": "local_ip", "toUser": "phone_number_placeholder", "sipTag": "sip_tag_placeholder", "sipCallID": "sip_call_id_placeholder", "twilioAccSID": "twilio_acc_sid_placeholder", "twilioCallSID": "twilio_call_sid_placeholder"}

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How do I add auth to Twilio SIP trunk for locally host SIP server?