RTP to ICE-lite with RTPEngine 12.5.1.2+0~mr12.5.1.2 – No Audio

[xiaofang han (hance)]

Hello everyone,

I’m running a media relay setup with the following flow:

FreeSWITCH(RTP, codec=OPUS) → RTPEngine → WhatsAPP(ICE-lite)

I’m encountering a complete no-audio issue after routing calls through RTPEngine, and I’m looking for help troubleshooting this.

Here is the core RTPEngine routing logic in my kamailio.cfg:

if {is_request() && !has_totag()){
rtpengine_manage("replace-origin replace-session-connection transport-protocol=UDP/TLS/RTP/SAVPF ICE=force trickle-ICE no-rtcp-attribute SDES=off generate-mid");
} else {
rtpengine_manage("replace-origin replace-session-connection");
}

Calls establish successfully (SIP signaling works fine), but there is no audio path in either direction.

Full RTPEngine debug logs are available here for reference:

https://pastebin.com/C6Wk8D62

[Richard Fuchs]

On 17/04/2026 06.02, xiaofang han (hance) wrote:
> Calls establish successfully (SIP signaling works fine), but there is
> no audio path in either direction.

From the log it looks like basic negotiation to the WebRTC endpoint
works fine. You probably want to add `rtcp-mux-require` to your flags
though.

There is audio received from the WebRTC endpoint and forwarded to the
RTP one, but nothing is ever received from the RTP endpoint. So either
you have a networking problem that prevents media flow to/from the RTP
endpoint, or there's something about the SDP that it doesn't like and
refuses to send/play RTP. So you should investigate on that side.

Cheers

[xiaofang han (hance)]

Hi Richard,

You were exactly right. The problem is now fixed, and everything is working as expected.

Many thanks for your great help!

Best regards,