Call connects but no audio (agent responds in logs) and drops after ~30 seconds

Sergio Alejandro Sanchez Ortiz
Wed, 25 Mar 2026 20:33:18 +0000
Telephony

We are integrating an Asterisk-based PBX with LiveKit via SIP trunk. The call connects successfully and signaling looks fine, but there is no audio during the call.

From Agent Insights, we can see that the agent is receiving input and generating responses, so the agent itself seems to be working correctly. However, nothing is heard on the call.

After around 30 seconds of silence, the call gets disconnected automatically.

Here are the call details:

Call ID: SCL_4iPbjY9yFfSH

Direction: Inbound

To: 2001

We suspect this might be related to RTP (possibly one-way audio or NAT/firewall issues), but wanted to confirm if there’s anything on the LiveKit side that could cause this behavior.

I’m attaching the PCAP for reference.

Thanks for your help.

SCL_4iPbjY9yFfSH_c3451aec-0873-47a6-bd9b-1cdca0262711.pcap (7.1 KB)

More calls for refernce:
SCL_ontzFkkCY9uS

SCL_ontzFkkCY9uS_c6115b38-6c27-4dde-8ede-e1f6a5cf3e09.pcap (175.5 KB)

5 posts - 3 participants