Voice issue on WEBRTC(WSS)

Hello Team,
I have successfully integrated WebRTC, and all functionalities are working seamlessly when making calls within the same PBX network, specifically from extension to extension. However, I encounter an issue when attempting to make calls to external destinations, such as GSM numbers. In these cases, there is no voice transmission for both the sender and the receiver.

I have thoroughly reviewed my configurations and settings, but the challenge persists. Given your experience in this domain, I would greatly appreciate any insights or guidance you can provide to help resolve this issue.

This could be a codec issue. Can you provide a freeswitch call log example when the call fails?