Mastering SIP Signaling and Session Maintenance

Why this matters

Without a precise understanding of SIP signaling, you will constantly face ghost calls, one-way audio, and unexpected call drops that frustrate your customers. Misinterpreting how devices stay connected leads to ineffective troubleshooting, often causing you to blame the carrier when the issue actually resides in the local network configuration.

The core idea

Session Initiation Protocol, or SIP, acts as the signaling language that sets up, modifies, and terminates multimedia sessions. While most people recognize the INVITE method as the starting point for a call, the actual health of that connection relies on background maintenance. The OPTIONS method is a vital signaling tool used to query the capabilities of an endpoint or simply verify that it is still responsive. Think of OPTIONS as a heartbeat signal sent between your VoIP phone or PBX and the service provider. By periodically sending these packets, the internal NAT (Network Address Translation) table of the office router stays populated.

This creates a pinhole, a temporary opening in the firewall that allows return traffic—like incoming calls or audio packets—to reach the internal phone without being blocked by security policies. Without this constant "poking" of the firewall, the path back to your phone closes, and the phone becomes unreachable until it registers again.

How it works in practice

In our environment, working with vendors like Cisco, Poly, and Yealink, the SIP stack is configured to perform periodic keep-alives. When a device registers with a SIP trunk or a hosted UCaaS provider, it typically sends a REGISTER request to tell the server its current IP address. However, the REGISTER request is not designed to be sent every few seconds, as this would overwhelm the server. Instead, we configure SIP timers for OPTIONS packets. In a typical scenario using a platform like BroadSoft or Metaswitch, your equipment is set to send an OPTIONS ping every 30 to 60 seconds.

If the endpoint does not receive a 200 OK response from the server, it recognizes that the connection has been lost or the network path is blocked. Our best practice at this company is to disable SIP ALG (Application Layer Gateway) on all edge routers. SIP ALG attempts to inspect and modify SIP packets, but it frequently corrupts the OPTIONS and INVITE signaling, leading to degraded service. Instead of relying on ALG, we rely on properly configured keep-alive intervals on the PBX.

Worked example

A customer complains that their office phones stop receiving calls exactly two hours after a reboot. You first suspect a registration timeout issue and attempt to send a test INVITE, but the INVITE fails because the firewall has already closed the port after a period of inactivity. Initially, you might try to solve this by shortening the registration interval, which forces the phone to re-register every five minutes. This creates massive overhead and increases the risk of being blacklisted by the provider for flooding the server. The correct approach is to check the device settings for the Keep-Alive or OPTIONS ping interval.

By setting the OPTIONS timer to 30 seconds and disabling the router's SIP ALG, you ensure that the firewall pinhole remains open indefinitely without overwhelming the registrar. Now, the phone responds to the server's periodic probes, and the firewall remains "aware" of the active session, allowing incoming calls to arrive instantly without interruption.

Where people go wrong

First, many technicians confuse call initiation with session maintenance. They assume that because an INVITE is used to start a call, it is the only way to signal status, but INVITE is far too resource-heavy to act as a heartbeat. Second, relying on SIP ALG is a common mistake; it is an outdated feature that rarely plays nice with modern hosted VoIP services and often mangles the very SIP headers needed to maintain the session. Third, failing to verify the NAT traversal settings results in asymmetric traffic, where outgoing calls work perfectly, but incoming calls fail after a short duration because the NAT pinhole has timed out.

Always verify the Keep-Alive settings in the phone's web interface rather than relying on global firewall settings.

Key takeaways

Use the OPTIONS method to maintain active NAT pinholes, not the INVITE method.

Always disable SIP ALG on customer edge routers to prevent signaling corruption.

Set your keep-alive intervals between 30 and 60 seconds to balance firewall requirements and server load.

If calls drop at fixed time intervals, suspect an expired NAT pinhole rather than a registration failure.

Check for 200 OK responses in your packet captures to confirm that your SIP heartbeat is successfully traversing the network.