diff --git a/README.md b/README.md
index 00d43d09..dbdc1df7 100644
--- a/README.md
+++ b/README.md
@@ -107,6 +107,7 @@ _rtsp-simple-server_ has been rebranded as _MediaMTX_. The reason is pretty obvi
   * [Decrease latency](#decrease-latency-1)
 * [WebRTC protocol](#webrtc-protocol)
   * [General usage](#general-usage-3)
+  * [WHIP and WHEP](#whip-and-whep)
   * [Usage inside a container or behind a NAT](#usage-inside-a-container-or-behind-a-nat)
   * [Embedding](#embedding-1)
 * [Standards](#standards)
@@ -1198,12 +1199,34 @@ To decrease the latency, you can:
 
 ### General usage
 
-Every stream published to the server can be read with WebRTC by visiting:
+You can publish a stream from the browser to the server with WebRTC by visiting:
+
+```
+http://localhost:8889/mystream/publish
+```
+
+You can read a stream from the browser with WebRTC by visiting:
 
 ```
 http://localhost:8889/mystream
 ```
 
+### WHIP and WHEP
+
+WHIP and WHEP are two WebRTC extensions that allows to publish and read streams with WebRTC without passing through a web page. This allows to use WebRTC as a general purpose streaming protocol.
+
+If you are using a software that supports WHIP, you can publish a stream to the server by using this WHIP URL:
+
+```
+http://localhost:8889/mystream/whip
+```
+
+If you are using a software that supports WHEP, you can read a stream from the server by using this WHEP URL:
+
+```
+http://localhost:8889/mystream/whep
+```
+
 ### Usage inside a container or behind a NAT
 
 If the server is hosted inside a container or is behind a NAT, additional configuration is required in order to allow the two WebRTC parts (the browser and the server) to establish a connection (WebRTC/ICE connection).
diff --git a/internal/core/webrtc_incoming_track.go b/internal/core/webrtc_incoming_track.go
index 1489c294..4081f58c 100644
--- a/internal/core/webrtc_incoming_track.go
+++ b/internal/core/webrtc_incoming_track.go
@@ -2,6 +2,7 @@ package core
 
 import (
 	"fmt"
+	"strings"
 	"time"
 
 	"github.com/bluenviron/gortsplib/v3/pkg/formats"
@@ -35,49 +36,53 @@ func newWebRTCIncomingTrack(
 		writeRTCP: writeRTCP,
 	}
 
-	switch track.Codec().MimeType {
-	case webrtc.MimeTypeAV1:
+	switch strings.ToLower(track.Codec().MimeType) {
+	case strings.ToLower(webrtc.MimeTypeAV1):
 		t.mediaType = media.TypeVideo
 		t.format = &formats.AV1{
 			PayloadTyp: uint8(track.PayloadType()),
 		}
 
-	case webrtc.MimeTypeVP9:
+	case strings.ToLower(webrtc.MimeTypeVP9):
 		t.mediaType = media.TypeVideo
 		t.format = &formats.VP9{
 			PayloadTyp: uint8(track.PayloadType()),
 		}
 
-	case webrtc.MimeTypeVP8:
+	case strings.ToLower(webrtc.MimeTypeVP8):
 		t.mediaType = media.TypeVideo
 		t.format = &formats.VP8{
 			PayloadTyp: uint8(track.PayloadType()),
 		}
 
-	case webrtc.MimeTypeH264:
+	case strings.ToLower(webrtc.MimeTypeH264):
 		t.mediaType = media.TypeVideo
 		t.format = &formats.H264{
 			PayloadTyp:        uint8(track.PayloadType()),
 			PacketizationMode: 1,
 		}
 
-	case webrtc.MimeTypeOpus:
+	case strings.ToLower(webrtc.MimeTypeOpus):
 		t.mediaType = media.TypeAudio
 		t.format = &formats.Opus{
 			PayloadTyp: uint8(track.PayloadType()),
 		}
 
-	case webrtc.MimeTypeG722:
+	case strings.ToLower(webrtc.MimeTypeG722):
 		t.mediaType = media.TypeAudio
 		t.format = &formats.G722{}
 
-	case webrtc.MimeTypePCMU:
+	case strings.ToLower(webrtc.MimeTypePCMU):
 		t.mediaType = media.TypeAudio
-		t.format = &formats.G711{MULaw: true}
+		t.format = &formats.G711{
+			MULaw: true,
+		}
 
-	case webrtc.MimeTypePCMA:
+	case strings.ToLower(webrtc.MimeTypePCMA):
 		t.mediaType = media.TypeAudio
-		t.format = &formats.G711{MULaw: false}
+		t.format = &formats.G711{
+			MULaw: false,
+		}
 
 	default:
 		return nil, fmt.Errorf("unsupported codec: %v", track.Codec())