diff --git a/README.md b/README.md
index 25ec0c46..3e09e429 100644
--- a/README.md
+++ b/README.md
@@ -22,8 +22,8 @@ Live streams can be published to the server with:
 |--------|--------|------------|------------|
 |[SRT clients](#srt-clients)||H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3|
 |[SRT cameras and servers](#srt-cameras-and-servers)||H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3|
-|[WebRTC clients](#webrtc-clients)|Browser-based, WHIP|AV1, VP9, VP8, H264|Opus, G722, G711 (PCMA, PCMU)|
-|[WebRTC servers](#webrtc-servers)|WHEP|AV1, VP9, VP8, H264|Opus, G722, G711 (PCMA, PCMU)|
+|[WebRTC clients](#webrtc-clients)|Browser-based, WHIP|AV1, VP9, VP8, H265, H264|Opus, G722, G711 (PCMA, PCMU)|
+|[WebRTC servers](#webrtc-servers)|WHEP|AV1, VP9, VP8, H265, H264|Opus, G722, G711 (PCMA, PCMU)|
 |[RTSP clients](#rtsp-clients)|UDP, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3, G726, G722, G711 (PCMA, PCMU), LPCM and any RTP-compatible codec|
 |[RTSP cameras and servers](#rtsp-cameras-and-servers)|UDP, UDP-Multicast, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3, G726, G722, G711 (PCMA, PCMU), LPCM and any RTP-compatible codec|
 |[RTMP clients](#rtmp-clients)|RTMP, RTMPS, Enhanced RTMP|AV1, VP9, H265, H264|MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), G711 (PCMA, PCMU), LPCM|
diff --git a/internal/protocols/webrtc/incoming_track.go b/internal/protocols/webrtc/incoming_track.go
index 96944031..71016e96 100644
--- a/internal/protocols/webrtc/incoming_track.go
+++ b/internal/protocols/webrtc/incoming_track.go
@@ -5,6 +5,7 @@ import (
 	"strings"
 	"time"
 
+	"github.com/bluenviron/gortsplib/v4/pkg/description"
 	"github.com/bluenviron/gortsplib/v4/pkg/format"
 	"github.com/bluenviron/gortsplib/v4/pkg/liberrors"
 	"github.com/bluenviron/gortsplib/v4/pkg/rtpreorderer"
@@ -81,9 +82,8 @@ var incomingVideoCodecs = []webrtc.RTPCodecParameters{
 	},
 	{
 		RTPCodecCapability: webrtc.RTPCodecCapability{
-			MimeType:    webrtc.MimeTypeH264,
-			ClockRate:   90000,
-			SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f",
+			MimeType:  webrtc.MimeTypeH265,
+			ClockRate: 90000,
 		},
 		PayloadType: 103,
 	},
@@ -91,10 +91,18 @@ var incomingVideoCodecs = []webrtc.RTPCodecParameters{
 		RTPCodecCapability: webrtc.RTPCodecCapability{
 			MimeType:    webrtc.MimeTypeH264,
 			ClockRate:   90000,
-			SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f",
+			SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f",
 		},
 		PayloadType: 104,
 	},
+	{
+		RTPCodecCapability: webrtc.RTPCodecCapability{
+			MimeType:    webrtc.MimeTypeH264,
+			ClockRate:   90000,
+			SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f",
+		},
+		PayloadType: 105,
+	},
 }
 
 var incomingAudioCodecs = []webrtc.RTPCodecParameters{
@@ -229,6 +237,7 @@ type IncomingTrack struct {
 	track *webrtc.TrackRemote
 	log   logger.Writer
 
+	typ       description.MediaType
 	format    format.Format
 	reorderer *rtpreorderer.Reorderer
 	pkts      []*rtp.Packet
@@ -246,41 +255,47 @@ func newIncomingTrack(
 		reorderer: rtpreorderer.New(),
 	}
 
-	isVideo := false
-
 	switch strings.ToLower(track.Codec().MimeType) {
 	case strings.ToLower(webrtc.MimeTypeAV1):
-		isVideo = true
+		t.typ = description.MediaTypeVideo
 		t.format = &format.AV1{
 			PayloadTyp: uint8(track.PayloadType()),
 		}
 
 	case strings.ToLower(webrtc.MimeTypeVP9):
-		isVideo = true
+		t.typ = description.MediaTypeVideo
 		t.format = &format.VP9{
 			PayloadTyp: uint8(track.PayloadType()),
 		}
 
 	case strings.ToLower(webrtc.MimeTypeVP8):
-		isVideo = true
+		t.typ = description.MediaTypeVideo
 		t.format = &format.VP8{
 			PayloadTyp: uint8(track.PayloadType()),
 		}
 
+	case strings.ToLower(webrtc.MimeTypeH265):
+		t.typ = description.MediaTypeVideo
+		t.format = &format.H265{
+			PayloadTyp: uint8(track.PayloadType()),
+		}
+
 	case strings.ToLower(webrtc.MimeTypeH264):
-		isVideo = true
+		t.typ = description.MediaTypeVideo
 		t.format = &format.H264{
 			PayloadTyp:        uint8(track.PayloadType()),
 			PacketizationMode: 1,
 		}
 
 	case strings.ToLower(mimeTypeMultiopus):
+		t.typ = description.MediaTypeAudio
 		t.format = &format.Opus{
 			PayloadTyp:   uint8(track.PayloadType()),
 			ChannelCount: int(track.Codec().Channels),
 		}
 
 	case strings.ToLower(webrtc.MimeTypeOpus):
+		t.typ = description.MediaTypeAudio
 		t.format = &format.Opus{
 			PayloadTyp: uint8(track.PayloadType()),
 			ChannelCount: func() int {
@@ -292,9 +307,12 @@ func newIncomingTrack(
 		}
 
 	case strings.ToLower(webrtc.MimeTypeG722):
+		t.typ = description.MediaTypeAudio
 		t.format = &format.G722{}
 
 	case strings.ToLower(webrtc.MimeTypePCMU):
+		t.typ = description.MediaTypeAudio
+
 		channels := track.Codec().Channels
 		if channels == 0 {
 			channels = 1
@@ -313,6 +331,8 @@ func newIncomingTrack(
 		}
 
 	case strings.ToLower(webrtc.MimeTypePCMA):
+		t.typ = description.MediaTypeAudio
+
 		channels := track.Codec().Channels
 		if channels == 0 {
 			channels = 1
@@ -331,6 +351,7 @@ func newIncomingTrack(
 		}
 
 	case strings.ToLower(mimeTypeL16):
+		t.typ = description.MediaTypeAudio
 		t.format = &format.LPCM{
 			PayloadTyp:   uint8(track.PayloadType()),
 			BitDepth:     16,
@@ -339,7 +360,7 @@ func newIncomingTrack(
 		}
 
 	default:
-		return nil, fmt.Errorf("unsupported codec: %+v", track.Codec())
+		return nil, fmt.Errorf("unsupported codec: %+v", track.Codec().RTPCodecCapability)
 	}
 
 	// read incoming RTCP packets to make interceptors work
@@ -354,7 +375,7 @@ func newIncomingTrack(
 	}()
 
 	// send period key frame requests
-	if isVideo {
+	if t.typ == description.MediaTypeVideo {
 		go func() {
 			keyframeTicker := time.NewTicker(keyFrameInterval)
 			defer keyframeTicker.Stop()
diff --git a/internal/protocols/webrtc/outgoing_track.go b/internal/protocols/webrtc/outgoing_track.go
index 3f72c1cc..adc60e51 100644
--- a/internal/protocols/webrtc/outgoing_track.go
+++ b/internal/protocols/webrtc/outgoing_track.go
@@ -54,6 +54,15 @@ func (t *OutgoingTrack) codecParameters() (webrtc.RTPCodecParameters, error) {
 			PayloadType: 96,
 		}, nil
 
+	case *format.H265:
+		return webrtc.RTPCodecParameters{
+			RTPCodecCapability: webrtc.RTPCodecCapability{
+				MimeType:  webrtc.MimeTypeH265,
+				ClockRate: 90000,
+			},
+			PayloadType: 96,
+		}, nil
+
 	case *format.H264:
 		return webrtc.RTPCodecParameters{
 			RTPCodecCapability: webrtc.RTPCodecCapability{
@@ -205,6 +214,7 @@ func (t *OutgoingTrack) isVideo() bool {
 	case *format.AV1,
 		*format.VP9,
 		*format.VP8,
+		*format.H265,
 		*format.H264:
 		return true
 	}
diff --git a/internal/protocols/webrtc/peer_connection_test.go b/internal/protocols/webrtc/peer_connection_test.go
index 1927786f..60454133 100644
--- a/internal/protocols/webrtc/peer_connection_test.go
+++ b/internal/protocols/webrtc/peer_connection_test.go
@@ -83,6 +83,17 @@ func TestPeerConnectionPublishRead(t *testing.T) {
 				PayloadTyp: 96,
 			},
 		},
+		{
+			"h265",
+			test.FormatH265,
+			webrtc.RTPCodecCapability{
+				MimeType:  "video/H265",
+				ClockRate: 90000,
+			},
+			&format.H265{
+				PayloadTyp: 96,
+			},
+		},
 		{
 			"h264",
 			test.FormatH264,
diff --git a/internal/protocols/webrtc/tracks_to_medias.go b/internal/protocols/webrtc/tracks_to_medias.go
index 809eeec4..81ea0850 100644
--- a/internal/protocols/webrtc/tracks_to_medias.go
+++ b/internal/protocols/webrtc/tracks_to_medias.go
@@ -10,21 +10,9 @@ func TracksToMedias(tracks []*IncomingTrack) []*description.Media {
 	ret := make([]*description.Media, len(tracks))
 
 	for i, track := range tracks {
-		forma := track.Format()
-
-		var mediaType description.MediaType
-
-		switch forma.(type) {
-		case *format.AV1, *format.VP9, *format.VP8, *format.H264:
-			mediaType = description.MediaTypeVideo
-
-		default:
-			mediaType = description.MediaTypeAudio
-		}
-
 		ret[i] = &description.Media{
-			Type:    mediaType,
-			Formats: []format.Format{forma},
+			Type:    track.typ,
+			Formats: []format.Format{track.format},
 		}
 	}