diff --git a/rtmp/rtmp_publisher.go b/rtmp/rtmp_publisher.go
index 9f4c23f..56c93dd 100644
--- a/rtmp/rtmp_publisher.go
+++ b/rtmp/rtmp_publisher.go
@@ -75,13 +75,7 @@ func (p *publisher) OnDiscardPacket(pkt interface{}) {
 }
 
 func (p *publisher) OnDeMuxStream(stream_ utils.AVStream) {
-	//AVStream的Data单独拷贝出来
 	//释放掉内存池中最新分配的内存
-	tmp := stream_.Extra()
-	bytes := make([]byte, len(tmp))
-	copy(bytes, tmp)
-	stream_.SetExtraData(bytes)
-
 	if utils.AVMediaTypeAudio == stream_.Type() {
 		p.audioMemoryPool.FreeTail()
 	} else if utils.AVMediaTypeVideo == stream_.Type() {
@@ -100,10 +94,11 @@ func (p *publisher) OnDeMuxPacket(packet utils.AVPacket) {
 		return
 	}
 
+	//未开启GOP缓存，释放掉内存
 	if utils.AVMediaTypeAudio == packet.MediaType() {
-		p.audioMemoryPool.FreeHead()
+		p.audioMemoryPool.FreeTail()
 	} else if utils.AVMediaTypeVideo == packet.MediaType() {
-		p.videoMemoryPool.FreeHead()
+		p.videoMemoryPool.FreeTail()
 	}
 }
 
diff --git a/rtsp/rtsp_session.go b/rtsp/rtsp_session.go
index 20cdfcb..64426f3 100644
--- a/rtsp/rtsp_session.go
+++ b/rtsp/rtsp_session.go
@@ -271,6 +271,7 @@ func (s *session) Input(method string, url_ *url.URL, headers textproto.MIMEHead
 	var err error
 	split := strings.Split(url_.Path, "/")
 	source := split[len(split)-1]
+
 	if MethodOptions == method {
 		err = s.onOptions(source, headers)
 	} else if MethodDescribe == method {
diff --git a/rtsp/rtsp_stream.go b/rtsp/rtsp_stream.go
index 442a561..95d8833 100644
--- a/rtsp/rtsp_stream.go
+++ b/rtsp/rtsp_stream.go
@@ -49,10 +49,13 @@ func (t *tranStream) onAllocBuffer(params interface{}) []byte {
 	return t.rtpTracks[params.(int)].buffer[OverTcpHeaderSize:]
 }
 
+// onRtpPacket 所有封装后的RTP流都将回调于此
 func (t *tranStream) onRtpPacket(data []byte, timestamp uint32, params interface{}) {
+	//params传递track索引
 	index := params.(int)
 	track := t.rtpTracks[index]
 
+	//保存sps和ssp等数据
 	if track.cache && track.header == nil {
 		bytes := make([]byte, OverTcpHeaderSize+len(data))
 		copy(bytes[OverTcpHeaderSize:], data)
@@ -155,11 +158,16 @@ func (t *tranStream) AddTrack(stream utils.AVStream) error {
 
 	//创建RTP封装
 	var muxer librtp.Muxer
-	if utils.AVCodecIdH264 == stream.CodecId() || utils.AVCodecIdH265 == stream.CodecId() {
+	if utils.AVCodecIdH264 == stream.CodecId() {
 		muxer = librtp.NewH264Muxer(payloadType.Pt, 0, 0xFFFFFFFF)
+	} else if utils.AVCodecIdH265 == stream.CodecId() {
+		muxer = librtp.NewH265Muxer(payloadType.Pt, 0, 0xFFFFFFFF)
 	} else if utils.AVCodecIdAAC == stream.CodecId() {
 		muxer = librtp.NewAACMuxer(payloadType.Pt, 0, 0xFFFFFFFF)
+	} else if utils.AVCodecIdPCMALAW == stream.CodecId() || utils.AVCodecIdPCMMULAW == stream.CodecId() {
+		muxer = librtp.NewMuxer(payloadType.Pt, 0, 0xFFFFFFFF)
 	}
+
 	muxer.SetAllocHandler(t.onAllocBuffer)
 	muxer.SetWriteHandler(t.onRtpPacket)