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

Telephony
,
1

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)

2

IP incorrect?

……………….

3

But I see the agent answering, or what do you mean by IP incorrect?

4

Everytime I’ve heard someone report this issue in the past it has turned out to be an issue with their firewall. Please make sure you firewall allows the audio packets through.

1 Like
5

The PBX owner mentioned there is no firewall, only NAT.
What configuration would you recommend we verify? Or what do you think could be causing this?