Troubleshooting a 32-Second Disconnection Issue (Broken ACK)

What causes a 32-second call drop?

A disconnection exactly at the 32-second mark is almost always caused by a broken ACK in the SIP signaling.

Here’s what happens:

  • Once a call is answered, the SIP server sends a 200 OK message, waiting for an ACK (acknowledgement) from the receiving device.

  • This ACK tells the server that the endpoint is ready to start the call.

  • However, in this case, the ACK never reaches the server, or is not in the expected format.

  • As a result, the server continues to send 200 OK messages, hoping to get a valid ACK.

  • After 32 seconds, the SIP server times out and sends a BYE, ending the call.

Try testing with a different SIP device or brand.
If the issue disappears, it’s likely that your current device sends broken or incompatible ACK messages.

 

⚠️ Other Possible Causes for a 32-Second Drop

While a broken ACK is the most common reason, other network or device misconfigurations may trigger this issue:

 

NAT (Network Address Translation) Problems

Routers or firewalls with NAT may block SIP or RTP packets if mappings are incorrect or keep-alives are missing.

Solution:

  • Disable SIP ALG (Application Layer Gateway) in your router settings.

  • Use proper NAT traversal settings (STUN, TURN, etc.).

  • Set up port forwarding for SIP and RTP if behind NAT.

 

Firewall Timeouts

Some firewalls drop SIP sessions after a default timeout (often 32 seconds) if packets aren’t acknowledged.

Solution:

  • Increase session timer limits on your firewall or SBC.

  • Ensure SIP and RTP traffic is allowed and kept stateful.

 

⏱️ SIP Timer Expiry (e.g., Timer B or Timer C)

If expected responses (like 200 OK or BYE) are not received in time, SIP timers may expire and end the session.

Solution:

  • Enable SIP keep-alives (SIP OPTIONS, re-INVITEs).

  • Ensure both endpoints support and use keep-alive signaling.

 

RTP or Media Path Blockages

If no media (RTP) packets arrive within a set period, the call is assumed dead and disconnected.

Solution:

  • Use Wireshark or similar tools to analyze RTP flow.

  • Make sure media is not blocked by NAT or firewall.

 

Session Timer Mismatch (RFC 4028)

A disagreement on session refresh timers between devices can lead to a call being dropped.

Solution:

  • Ensure matching Session-Expires values on both endpoints.

  • Adjust the timers or disable session expiration where possible.

 

Codec Negotiation Failures

A failed or unsupported codec can lead to mid-call termination.

Solution:

  • Check codec compatibility on both devices.

  • Avoid obscure codecs or unnecessary transcoding.

 

Still Stuck?

  • Run a SIP trace (Wireshark is your friend) and inspect the call flow, especially the ACK.

  • Check for firmware updates on your device or softphone — some issues are fixed in later versions.

 

If you’ve tried the steps above and still run into problems, please reach out to our support team.
We’re happy to assist you in getting your connection stable again.

 

 

Last updated: 01/04/2025

 

 


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Given the fast evolution of the telecom sector, Voyced does not offer any guarantee about the accuracy and completeness of the information offered. 
We explicitly do not give anyone permission to use this information in any other way than to inform themselves about what (legal) requirements there may be and they need to comply with when registering Services and/or DIDs with Voyced.

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