diff --git a/docs/changelog.md b/docs/changelog.md
index ff2f5477..2c20b69b 100644
--- a/docs/changelog.md
+++ b/docs/changelog.md
@@ -19,6 +19,7 @@
 - Broadcast status change is sent to all admins now.
 - NordVPN replaced with Sponsorblock extension in default configuration #144.
 - Removed `vncviewer` image, as its functionality is replaced and extended by remmina.
+- Opus uses `useinbandfec=1` from now on, hopefully fixes minor audio loss issues.
 
 ## [n.eko v2.5](https://github.com/m1k1o/neko/releases/tag/v2.5)
 
diff --git a/server/internal/gst/gst.go b/server/internal/gst/gst.go
index a63c3f62..e1d270c8 100644
--- a/server/internal/gst/gst.go
+++ b/server/internal/gst/gst.go
@@ -160,7 +160,7 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
 			return nil, err
 		}
 
-		pipelineStr = fmt.Sprintf(audioSrc+"opusenc bitrate=%d"+pipelineStr, pipelineDevice, bitrate*1000)
+		pipelineStr = fmt.Sprintf(audioSrc+"opusenc inband-fec=true bitrate=%d"+pipelineStr, pipelineDevice, bitrate*1000)
 	case "G722":
 		// https://gstreamer.freedesktop.org/documentation/libav/avenc_g722.html?gi-language=c
 		// gstreamer1.0-libav
diff --git a/server/internal/webrtc/webrtc.go b/server/internal/webrtc/webrtc.go
index 44748b74..126d40b9 100644
--- a/server/internal/webrtc/webrtc.go
+++ b/server/internal/webrtc/webrtc.go
@@ -322,7 +322,7 @@ func (manager *WebRTCManager) createTrack(codecName string) (*webrtc.TrackLocalS
 		codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeH264, ClockRate: 90000, Channels: 0, SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f", RTCPFeedback: fb}, PayloadType: 102}
 		id = "video"
 	case "Opus":
-		codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeOpus, ClockRate: 48000, Channels: 2, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 111}
+		codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeOpus, ClockRate: 48000, Channels: 2, SDPFmtpLine: "useinbandfec=1", RTCPFeedback: fb}, PayloadType: 111}
 		id = "audio"
 	case "G722":
 		codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeG722, ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 9}