diff --git a/README.md b/README.md
index 79511a1260..c957836f3e 100755
--- a/README.md
+++ b/README.md
@@ -176,6 +176,7 @@ The ports used by SRS:
 
 ## V4 changes
 
+* v4.0, 2021-05-15, Rename SrsRtpPacket2 to SrsRtpPacket. 4.0.111
 * v4.0, 2021-05-14, RTC: Remove [Object Cache Pool](https://github.com/ossrs/srs/commit/14bfc98122bba369572417c19ebb2a61b373fc45#commitcomment-47655008), no effect. 4.0.110
 * v4.0, 2021-05-14, Change virtual public to public. 4.0.109
 * v4.0, 2021-05-14, Refine id and vid for statistic. 4.0.108
diff --git a/trunk/src/app/srs_app_gb28181.hpp b/trunk/src/app/srs_app_gb28181.hpp
index fdc421fbeb..4d3e37fa10 100644
--- a/trunk/src/app/srs_app_gb28181.hpp
+++ b/trunk/src/app/srs_app_gb28181.hpp
@@ -77,7 +77,7 @@
 #define STREAM_TYPE_AUDIO_PCM       0x9c
 
 class SrsConfDirective;
-class SrsRtpPacket;
+class SrsRtspPacket;
 class SrsRtmpClient;
 class SrsRawH264Stream;
 class SrsRawAacStream;
@@ -108,7 +108,7 @@ class SrsGb28181Caster;
 
 //ps rtp header packet parse
 
-class SrsPsRtpPacket: public SrsRtpPacket
+class SrsPsRtpPacket: public SrsRtspPacket
 {
 public:
     SrsPsRtpPacket();
diff --git a/trunk/src/app/srs_app_rtc_conn.cpp b/trunk/src/app/srs_app_rtc_conn.cpp
index 240f8c1864..c910730fca 100644
--- a/trunk/src/app/srs_app_rtc_conn.cpp
+++ b/trunk/src/app/srs_app_rtc_conn.cpp
@@ -594,7 +594,7 @@ srs_error_t SrsRtcPlayStream::cycle()
         }
 
         // Wait for amount of packets.
-        SrsRtpPacket2* pkt = NULL;
+        SrsRtpPacket* pkt = NULL;
         consumer->dump_packet(&pkt);
         if (!pkt) {
             // TODO: FIXME: We should check the quit event.
@@ -618,7 +618,7 @@ srs_error_t SrsRtcPlayStream::cycle()
     }
 }
 
-srs_error_t SrsRtcPlayStream::send_packet(SrsRtpPacket2*& pkt)
+srs_error_t SrsRtcPlayStream::send_packet(SrsRtpPacket*& pkt)
 {
     srs_error_t err = srs_success;
 
@@ -1287,7 +1287,7 @@ srs_error_t SrsRtcPublishStream::on_rtp_plaintext(char* plaintext, int nb_plaint
     }
 
     // Allocate packet form cache.
-    SrsRtpPacket2* pkt = new SrsRtpPacket2();
+    SrsRtpPacket* pkt = new SrsRtpPacket();
 
     // Copy the packet body.
     char* p = pkt->wrap(plaintext, nb_plaintext);
@@ -1305,7 +1305,7 @@ srs_error_t SrsRtcPublishStream::on_rtp_plaintext(char* plaintext, int nb_plaint
     return err;
 }
 
-srs_error_t SrsRtcPublishStream::do_on_rtp_plaintext(SrsRtpPacket2*& pkt, SrsBuffer* buf)
+srs_error_t SrsRtcPublishStream::do_on_rtp_plaintext(SrsRtpPacket*& pkt, SrsBuffer* buf)
 {
     srs_error_t err = srs_success;
 
@@ -1389,7 +1389,7 @@ srs_error_t SrsRtcPublishStream::check_send_nacks()
     return err;
 }
 
-void SrsRtcPublishStream::on_before_decode_payload(SrsRtpPacket2* pkt, SrsBuffer* buf, ISrsRtpPayloader** ppayload, SrsRtpPacketPayloadType* ppt)
+void SrsRtcPublishStream::on_before_decode_payload(SrsRtpPacket* pkt, SrsBuffer* buf, ISrsRtpPayloader** ppayload, SrsRtspPacketPayloadType* ppt)
 {
     // No payload, ignore.
     if (buf->empty()) {
@@ -2528,7 +2528,7 @@ void SrsRtcConnection::simulate_player_drop_packet(SrsRtpHeader* h, int nn_bytes
     nn_simulate_player_nack_drop--;
 }
 
-srs_error_t SrsRtcConnection::do_send_packet(SrsRtpPacket2* pkt)
+srs_error_t SrsRtcConnection::do_send_packet(SrsRtpPacket* pkt)
 {
     srs_error_t err = srs_success;
 
diff --git a/trunk/src/app/srs_app_rtc_conn.hpp b/trunk/src/app/srs_app_rtc_conn.hpp
index 9010f87aed..5c7bcd2eff 100644
--- a/trunk/src/app/srs_app_rtc_conn.hpp
+++ b/trunk/src/app/srs_app_rtc_conn.hpp
@@ -53,7 +53,7 @@ class SrsRtcServer;
 class SrsRtcConnection;
 class SrsSharedPtrMessage;
 class SrsRtcStream;
-class SrsRtpPacket2;
+class SrsRtpPacket;
 class ISrsCodec;
 class SrsRtpNackForReceiver;
 class SrsRtpIncommingVideoFrame;
@@ -266,7 +266,7 @@ class SrsRtcPlayStream : public ISrsCoroutineHandler, public ISrsReloadHandler
 public:
     virtual srs_error_t cycle();
 private:
-    srs_error_t send_packet(SrsRtpPacket2*& pkt);
+    srs_error_t send_packet(SrsRtpPacket*& pkt);
 public:
     // Directly set the status of track, generally for init to set the default value.
     void set_all_tracks_status(bool status);
@@ -310,7 +310,7 @@ class SrsRtcPublishTwccTimer : public ISrsFastTimer
 };
 
 // A RTC publish stream, client push and publish stream to SRS.
-class SrsRtcPublishStream : public ISrsRtpPacketDecodeHandler
+class SrsRtcPublishStream : public ISrsRtspPacketDecodeHandler
     , public ISrsRtcPublishStream, public ISrsRtcPLIWorkerHandler
 {
 private:
@@ -367,11 +367,11 @@ class SrsRtcPublishStream : public ISrsRtpPacketDecodeHandler
     // @remark We copy the plaintext, user should free it.
     srs_error_t on_rtp_plaintext(char* plaintext, int nb_plaintext);
 private:
-    srs_error_t do_on_rtp_plaintext(SrsRtpPacket2*& pkt, SrsBuffer* buf);
+    srs_error_t do_on_rtp_plaintext(SrsRtpPacket*& pkt, SrsBuffer* buf);
 public:
     srs_error_t check_send_nacks();
 public:
-    virtual void on_before_decode_payload(SrsRtpPacket2* pkt, SrsBuffer* buf, ISrsRtpPayloader** ppayload, SrsRtpPacketPayloadType* ppt);
+    virtual void on_before_decode_payload(SrsRtpPacket* pkt, SrsBuffer* buf, ISrsRtpPayloader** ppayload, SrsRtspPacketPayloadType* ppt);
 private:
     srs_error_t send_periodic_twcc();
 public:
@@ -547,7 +547,7 @@ class SrsRtcConnection : public ISrsResource, public ISrsDisposingHandler
     // Simulate the NACK to drop nn packets.
     void simulate_nack_drop(int nn);
     void simulate_player_drop_packet(SrsRtpHeader* h, int nn_bytes);
-    srs_error_t do_send_packet(SrsRtpPacket2* pkt);
+    srs_error_t do_send_packet(SrsRtpPacket* pkt);
     // Directly set the status of play track, generally for init to set the default value.
     void set_all_tracks_status(std::string stream_uri, bool is_publish, bool status);
 private:
@@ -579,7 +579,7 @@ class ISrsRtcHijacker
     // When stop publish by RTC.
     virtual void on_stop_publish(SrsRtcConnection* session, SrsRtcPublishStream* publisher, SrsRequest* req) = 0;
     // When got RTP plaintext packet.
-    virtual srs_error_t on_rtp_packet(SrsRtcConnection* session, SrsRtcPublishStream* publisher, SrsRequest* req, SrsRtpPacket2* pkt) = 0;
+    virtual srs_error_t on_rtp_packet(SrsRtcConnection* session, SrsRtcPublishStream* publisher, SrsRequest* req, SrsRtpPacket* pkt) = 0;
     // When before play by RTC. (wait source to ready in cascade scenario)
     virtual srs_error_t on_before_play(SrsRtcConnection* session, SrsRequest* req) = 0;
     // When start player by RTC.
diff --git a/trunk/src/app/srs_app_rtc_jitbuffer.hpp b/trunk/src/app/srs_app_rtc_jitbuffer.hpp
index d0521c2f82..eef613c08c 100644
--- a/trunk/src/app/srs_app_rtc_jitbuffer.hpp
+++ b/trunk/src/app/srs_app_rtc_jitbuffer.hpp
@@ -44,7 +44,7 @@ class SrsRtpFrameBuffer;
 class SrsRtpDecodingState;
 class SrsGb28181RtmpMuxer;
 class VCMPacket;
-class SrsRtpPacket2;
+class SrsRtpPacket;
 
 ///jittbuffer
 
diff --git a/trunk/src/app/srs_app_rtc_queue.cpp b/trunk/src/app/srs_app_rtc_queue.cpp
index fe3b40a669..06dba7f2d5 100644
--- a/trunk/src/app/srs_app_rtc_queue.cpp
+++ b/trunk/src/app/srs_app_rtc_queue.cpp
@@ -47,14 +47,14 @@ SrsRtpRingBuffer::SrsRtpRingBuffer(int capacity)
     capacity_ = (uint16_t)capacity;
     initialized_ = false;
 
-    queue_ = new SrsRtpPacket2*[capacity_];
-    memset(queue_, 0, sizeof(SrsRtpPacket2*) * capacity);
+    queue_ = new SrsRtpPacket*[capacity_];
+    memset(queue_, 0, sizeof(SrsRtpPacket*) * capacity);
 }
 
 SrsRtpRingBuffer::~SrsRtpRingBuffer()
 {
     for (int i = 0; i < capacity_; ++i) {
-        SrsRtpPacket2* pkt = queue_[i];
+        SrsRtpPacket* pkt = queue_[i];
         srs_freep(pkt);
     }
     srs_freepa(queue_);
@@ -77,9 +77,9 @@ void SrsRtpRingBuffer::advance_to(uint16_t seq)
     begin = seq;
 }
 
-void SrsRtpRingBuffer::set(uint16_t at, SrsRtpPacket2* pkt)
+void SrsRtpRingBuffer::set(uint16_t at, SrsRtpPacket* pkt)
 {
-    SrsRtpPacket2* p = queue_[at % capacity_];
+    SrsRtpPacket* p = queue_[at % capacity_];
     srs_freep(p);
 
     queue_[at % capacity_] = pkt;
@@ -143,7 +143,7 @@ bool SrsRtpRingBuffer::update(uint16_t seq, uint16_t& nack_first, uint16_t& nack
     return true;
 }
 
-SrsRtpPacket2* SrsRtpRingBuffer::at(uint16_t seq) {
+SrsRtpPacket* SrsRtpRingBuffer::at(uint16_t seq) {
     return queue_[seq % capacity_];
 }
 
@@ -165,7 +165,7 @@ void SrsRtpRingBuffer::clear_histroy(uint16_t seq)
 {
     // TODO FIXME Did not consider loopback
     for (uint16_t i = 0; i < capacity_; i++) {
-        SrsRtpPacket2* p = queue_[i];
+        SrsRtpPacket* p = queue_[i];
         if (p && p->header.get_sequence() < seq) {
             srs_freep(p);
             queue_[i] = NULL;
@@ -176,7 +176,7 @@ void SrsRtpRingBuffer::clear_histroy(uint16_t seq)
 void SrsRtpRingBuffer::clear_all_histroy()
 {
     for (uint16_t i = 0; i < capacity_; i++) {
-        SrsRtpPacket2* p = queue_[i];
+        SrsRtpPacket* p = queue_[i];
         if (p) {
             srs_freep(p);
             queue_[i] = NULL;
diff --git a/trunk/src/app/srs_app_rtc_queue.hpp b/trunk/src/app/srs_app_rtc_queue.hpp
index 5cd6444428..37b43d6432 100644
--- a/trunk/src/app/srs_app_rtc_queue.hpp
+++ b/trunk/src/app/srs_app_rtc_queue.hpp
@@ -33,7 +33,7 @@
 #include <srs_kernel_rtc_rtp.hpp>
 #include <srs_kernel_rtc_rtcp.hpp>
 
-class SrsRtpPacket2;
+class SrsRtpPacket;
 class SrsRtpQueue;
 class SrsRtpRingBuffer;
 
@@ -51,7 +51,7 @@ class SrsRtpRingBuffer
     // Capacity of the ring-buffer.
     uint16_t capacity_;
     // Ring bufer.
-    SrsRtpPacket2** queue_;
+    SrsRtpPacket** queue_;
     // Increase one when uint16 flip back, for get_extended_highest_sequence.
     uint64_t nn_seq_flip_backs;
     // Whether initialized, because we use uint16 so we can't use -1.
@@ -74,7 +74,7 @@ class SrsRtpRingBuffer
     // Move the low position of buffer to seq.
     void advance_to(uint16_t seq);
     // Free the packet at position.
-    void set(uint16_t at, SrsRtpPacket2* pkt);
+    void set(uint16_t at, SrsRtpPacket* pkt);
     void remove(uint16_t at);
     // The highest sequence number, calculate the flip back base.
     uint32_t get_extended_highest_sequence();
@@ -82,7 +82,7 @@ class SrsRtpRingBuffer
     // @return If false, the seq is too old.
     bool update(uint16_t seq, uint16_t& nack_first, uint16_t& nack_last);
     // Get the packet by seq.
-    SrsRtpPacket2* at(uint16_t seq);
+    SrsRtpPacket* at(uint16_t seq);
 public:
     // TODO: FIXME: Refine it?
     void notify_nack_list_full();
diff --git a/trunk/src/app/srs_app_rtc_source.cpp b/trunk/src/app/srs_app_rtc_source.cpp
index 6b789a55f2..b068b2a703 100644
--- a/trunk/src/app/srs_app_rtc_source.cpp
+++ b/trunk/src/app/srs_app_rtc_source.cpp
@@ -175,9 +175,9 @@ SrsRtcConsumer::~SrsRtcConsumer()
 {
     source->on_consumer_destroy(this);
 
-    vector<SrsRtpPacket2*>::iterator it;
+    vector<SrsRtpPacket*>::iterator it;
     for (it = queue.begin(); it != queue.end(); ++it) {
-        SrsRtpPacket2* pkt = *it;
+        SrsRtpPacket* pkt = *it;
         srs_freep(pkt);
     }
 
@@ -189,7 +189,7 @@ void SrsRtcConsumer::update_source_id()
     should_update_source_id = true;
 }
 
-srs_error_t SrsRtcConsumer::enqueue(SrsRtpPacket2* pkt)
+srs_error_t SrsRtcConsumer::enqueue(SrsRtpPacket* pkt)
 {
     srs_error_t err = srs_success;
 
@@ -206,7 +206,7 @@ srs_error_t SrsRtcConsumer::enqueue(SrsRtpPacket2* pkt)
     return err;
 }
 
-srs_error_t SrsRtcConsumer::dump_packet(SrsRtpPacket2** ppkt)
+srs_error_t SrsRtcConsumer::dump_packet(SrsRtpPacket** ppkt)
 {
     srs_error_t err = srs_success;
 
@@ -574,7 +574,7 @@ void SrsRtcStream::set_publish_stream(ISrsRtcPublishStream* v)
     publish_stream_ = v;
 }
 
-srs_error_t SrsRtcStream::on_rtp(SrsRtpPacket2* pkt)
+srs_error_t SrsRtcStream::on_rtp(SrsRtpPacket* pkt)
 {
     srs_error_t err = srs_success;
 
@@ -848,8 +848,8 @@ srs_error_t SrsRtcFromRtmpBridger::transcode(SrsAudioFrame* audio)
     for (std::vector<SrsAudioFrame*>::iterator it = out_audios.begin(); it != out_audios.end(); ++it) {
         SrsAudioFrame* out_audio = *it;
 
-        SrsRtpPacket2* pkt = new SrsRtpPacket2();
-        SrsAutoFree(SrsRtpPacket2, pkt);
+        SrsRtpPacket* pkt = new SrsRtpPacket();
+        SrsAutoFree(SrsRtpPacket, pkt);
 
         if ((err = package_opus(out_audio, pkt)) != srs_success) {
             err = srs_error_wrap(err, "package opus");
@@ -867,7 +867,7 @@ srs_error_t SrsRtcFromRtmpBridger::transcode(SrsAudioFrame* audio)
     return err;
 }
 
-srs_error_t SrsRtcFromRtmpBridger::package_opus(SrsAudioFrame* audio, SrsRtpPacket2* pkt)
+srs_error_t SrsRtcFromRtmpBridger::package_opus(SrsAudioFrame* audio, SrsRtpPacket* pkt)
 {
     srs_error_t err = srs_success;
 
@@ -879,7 +879,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_opus(SrsAudioFrame* audio, SrsRtpPack
     pkt->header.set_timestamp(audio->dts * 48);
 
     SrsRtpRawPayload* raw = new SrsRtpRawPayload();
-    pkt->set_payload(raw, SrsRtpPacketPayloadTypeRaw);
+    pkt->set_payload(raw, SrsRtspPacketPayloadTypeRaw);
 
     srs_assert(audio->nb_samples == 1);
     raw->payload = pkt->wrap(audio->samples[0].bytes, audio->samples[0].size);
@@ -911,8 +911,8 @@ srs_error_t SrsRtcFromRtmpBridger::on_video(SrsSharedPtrMessage* msg)
 
     // Well, for each IDR, we append a SPS/PPS before it, which is packaged in STAP-A.
     if (has_idr) {
-        SrsRtpPacket2* pkt = new SrsRtpPacket2();
-        SrsAutoFree(SrsRtpPacket2, pkt);
+        SrsRtpPacket* pkt = new SrsRtpPacket();
+        SrsAutoFree(SrsRtpPacket, pkt);
 
         if ((err = package_stap_a(source_, msg, pkt)) != srs_success) {
             return srs_error_wrap(err, "package stap-a");
@@ -924,7 +924,7 @@ srs_error_t SrsRtcFromRtmpBridger::on_video(SrsSharedPtrMessage* msg)
     }
 
     // If merge Nalus, we pcakges all NALUs(samples) as one NALU, in a RTP or FUA packet.
-    vector<SrsRtpPacket2*> pkts;
+    vector<SrsRtpPacket*> pkts;
     if (merge_nalus && nn_samples > 1) {
         if ((err = package_nalus(msg, samples, pkts)) != srs_success) {
             return srs_error_wrap(err, "package nalus as one");
@@ -989,7 +989,7 @@ srs_error_t SrsRtcFromRtmpBridger::filter(SrsSharedPtrMessage* msg, SrsFormat* f
     return err;
 }
 
-srs_error_t SrsRtcFromRtmpBridger::package_stap_a(SrsRtcStream* source, SrsSharedPtrMessage* msg, SrsRtpPacket2* pkt)
+srs_error_t SrsRtcFromRtmpBridger::package_stap_a(SrsRtcStream* source, SrsSharedPtrMessage* msg, SrsRtpPacket* pkt)
 {
     srs_error_t err = srs_success;
 
@@ -1014,7 +1014,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_stap_a(SrsRtcStream* source, SrsShare
     pkt->header.set_timestamp(msg->timestamp * 90);
 
     SrsRtpSTAPPayload* stap = new SrsRtpSTAPPayload();
-    pkt->set_payload(stap, SrsRtpPacketPayloadTypeSTAP);
+    pkt->set_payload(stap, SrsRtspPacketPayloadTypeSTAP);
 
     uint8_t header = sps[0];
     stap->nri = (SrsAvcNaluType)header;
@@ -1048,7 +1048,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_stap_a(SrsRtcStream* source, SrsShare
     return err;
 }
 
-srs_error_t SrsRtcFromRtmpBridger::package_nalus(SrsSharedPtrMessage* msg, const vector<SrsSample*>& samples, vector<SrsRtpPacket2*>& pkts)
+srs_error_t SrsRtcFromRtmpBridger::package_nalus(SrsSharedPtrMessage* msg, const vector<SrsSample*>& samples, vector<SrsRtpPacket*>& pkts)
 {
     srs_error_t err = srs_success;
 
@@ -1084,7 +1084,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_nalus(SrsSharedPtrMessage* msg, const
 
     if (nn_bytes < kRtpMaxPayloadSize) {
         // Package NALUs in a single RTP packet.
-        SrsRtpPacket2* pkt = new SrsRtpPacket2();
+        SrsRtpPacket* pkt = new SrsRtpPacket();
         pkts.push_back(pkt);
 
         pkt->header.set_payload_type(kVideoPayloadType);
@@ -1093,7 +1093,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_nalus(SrsSharedPtrMessage* msg, const
         pkt->nalu_type = (SrsAvcNaluType)first_nalu_type;
         pkt->header.set_sequence(video_sequence++);
         pkt->header.set_timestamp(msg->timestamp * 90);
-        pkt->set_payload(raw, SrsRtpPacketPayloadTypeNALU);
+        pkt->set_payload(raw, SrsRtspPacketPayloadTypeNALU);
         pkt->wrap(msg);
     } else {
         // We must free it, should never use RTP packets to free it,
@@ -1118,7 +1118,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_nalus(SrsSharedPtrMessage* msg, const
                 return srs_error_wrap(err, "read samples %d bytes, left %d, total %d", packet_size, nb_left, nn_bytes);
             }
 
-            SrsRtpPacket2* pkt = new SrsRtpPacket2();
+            SrsRtpPacket* pkt = new SrsRtpPacket();
             pkts.push_back(pkt);
 
             pkt->header.set_payload_type(kVideoPayloadType);
@@ -1133,7 +1133,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_nalus(SrsSharedPtrMessage* msg, const
             fua->start = bool(i == 0);
             fua->end = bool(i == num_of_packet - 1);
 
-            pkt->set_payload(fua, SrsRtpPacketPayloadTypeFUA);
+            pkt->set_payload(fua, SrsRtspPacketPayloadTypeFUA);
             pkt->wrap(msg);
 
             nb_left -= packet_size;
@@ -1144,11 +1144,11 @@ srs_error_t SrsRtcFromRtmpBridger::package_nalus(SrsSharedPtrMessage* msg, const
 }
 
 // Single NAL Unit Packet @see https://tools.ietf.org/html/rfc6184#section-5.6
-srs_error_t SrsRtcFromRtmpBridger::package_single_nalu(SrsSharedPtrMessage* msg, SrsSample* sample, vector<SrsRtpPacket2*>& pkts)
+srs_error_t SrsRtcFromRtmpBridger::package_single_nalu(SrsSharedPtrMessage* msg, SrsSample* sample, vector<SrsRtpPacket*>& pkts)
 {
     srs_error_t err = srs_success;
 
-    SrsRtpPacket2* pkt = new SrsRtpPacket2();
+    SrsRtpPacket* pkt = new SrsRtpPacket();
     pkts.push_back(pkt);
 
     pkt->header.set_payload_type(kVideoPayloadType);
@@ -1158,7 +1158,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_single_nalu(SrsSharedPtrMessage* msg,
     pkt->header.set_timestamp(msg->timestamp * 90);
 
     SrsRtpRawPayload* raw = new SrsRtpRawPayload();
-    pkt->set_payload(raw, SrsRtpPacketPayloadTypeRaw);
+    pkt->set_payload(raw, SrsRtspPacketPayloadTypeRaw);
 
     raw->payload = sample->bytes;
     raw->nn_payload = sample->size;
@@ -1168,7 +1168,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_single_nalu(SrsSharedPtrMessage* msg,
     return err;
 }
 
-srs_error_t SrsRtcFromRtmpBridger::package_fu_a(SrsSharedPtrMessage* msg, SrsSample* sample, int fu_payload_size, vector<SrsRtpPacket2*>& pkts)
+srs_error_t SrsRtcFromRtmpBridger::package_fu_a(SrsSharedPtrMessage* msg, SrsSample* sample, int fu_payload_size, vector<SrsRtpPacket*>& pkts)
 {
     srs_error_t err = srs_success;
 
@@ -1181,7 +1181,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_fu_a(SrsSharedPtrMessage* msg, SrsSam
     for (int i = 0; i < num_of_packet; ++i) {
         int packet_size = srs_min(nb_left, fu_payload_size);
 
-        SrsRtpPacket2* pkt = new SrsRtpPacket2();
+        SrsRtpPacket* pkt = new SrsRtpPacket();
         pkts.push_back(pkt);
 
         pkt->header.set_payload_type(kVideoPayloadType);
@@ -1191,7 +1191,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_fu_a(SrsSharedPtrMessage* msg, SrsSam
         pkt->header.set_timestamp(msg->timestamp * 90);
 
         SrsRtpFUAPayload2* fua = new SrsRtpFUAPayload2();
-        pkt->set_payload(fua, SrsRtpPacketPayloadTypeFUA2);
+        pkt->set_payload(fua, SrsRtspPacketPayloadTypeFUA2);
 
         fua->nri = (SrsAvcNaluType)header;
         fua->nalu_type = (SrsAvcNaluType)nal_type;
@@ -1210,13 +1210,13 @@ srs_error_t SrsRtcFromRtmpBridger::package_fu_a(SrsSharedPtrMessage* msg, SrsSam
     return err;
 }
 
-srs_error_t SrsRtcFromRtmpBridger::consume_packets(vector<SrsRtpPacket2*>& pkts)
+srs_error_t SrsRtcFromRtmpBridger::consume_packets(vector<SrsRtpPacket*>& pkts)
 {
     srs_error_t err = srs_success;
 
     // TODO: FIXME: Consume a range of packets.
     for (int i = 0; i < (int)pkts.size(); i++) {
-        SrsRtpPacket2* pkt = pkts[i];
+        SrsRtpPacket* pkt = pkts[i];
         if ((err = source_->on_rtp(pkt)) != srs_success) {
             err = srs_error_wrap(err, "consume sps/pps");
             break;
@@ -1224,7 +1224,7 @@ srs_error_t SrsRtcFromRtmpBridger::consume_packets(vector<SrsRtpPacket2*>& pkts)
     }
 
     for (int i = 0; i < (int)pkts.size(); i++) {
-        SrsRtpPacket2* pkt = pkts[i];
+        SrsRtpPacket* pkt = pkts[i];
         srs_freep(pkt);
     }
 
@@ -1288,7 +1288,7 @@ srs_error_t SrsRtmpFromRtcBridger::on_publish()
     return err;
 }
 
-srs_error_t SrsRtmpFromRtcBridger::on_rtp(SrsRtpPacket2 *pkt)
+srs_error_t SrsRtmpFromRtcBridger::on_rtp(SrsRtpPacket *pkt)
 {
     srs_error_t err = srs_success;
 
@@ -1311,7 +1311,7 @@ void SrsRtmpFromRtcBridger::on_unpublish()
     source_->on_unpublish();
 }
 
-srs_error_t SrsRtmpFromRtcBridger::trancode_audio(SrsRtpPacket2 *pkt)
+srs_error_t SrsRtmpFromRtcBridger::trancode_audio(SrsRtpPacket *pkt)
 {
     srs_error_t err = srs_success;
 
@@ -1377,12 +1377,12 @@ void SrsRtmpFromRtcBridger::packet_aac(SrsCommonMessage* audio, char* data, int
     audio->size = rtmp_len;
 }
 
-srs_error_t SrsRtmpFromRtcBridger::packet_video(SrsRtpPacket2* src)
+srs_error_t SrsRtmpFromRtcBridger::packet_video(SrsRtpPacket* src)
 {
     srs_error_t err = srs_success;
 
     // TODO: Only copy when need
-    SrsRtpPacket2* pkt = src->copy();
+    SrsRtpPacket* pkt = src->copy();
 
     if (pkt->is_keyframe()) {
         return packet_video_key_frame(pkt);
@@ -1415,7 +1415,7 @@ srs_error_t SrsRtmpFromRtcBridger::packet_video(SrsRtpPacket2* src)
     return err;
 }
 
-srs_error_t SrsRtmpFromRtcBridger::packet_video_key_frame(SrsRtpPacket2* pkt)
+srs_error_t SrsRtmpFromRtcBridger::packet_video_key_frame(SrsRtpPacket* pkt)
 {
     srs_error_t err = srs_success;
 
@@ -1519,7 +1519,7 @@ srs_error_t SrsRtmpFromRtcBridger::packet_video_rtmp(const uint16_t start, const
     for (uint16_t i = 0; i < cnt; ++i) {
         uint16_t sn = start + i;
         uint16_t index = cache_index(sn);
-        SrsRtpPacket2* pkt = cache_video_pkts_[index].pkt;
+        SrsRtpPacket* pkt = cache_video_pkts_[index].pkt;
         // calculate nalu len
         SrsRtpFUAPayload2* fua_payload = dynamic_cast<SrsRtpFUAPayload2*>(pkt->payload());
         if (fua_payload) {
@@ -1547,7 +1547,7 @@ srs_error_t SrsRtmpFromRtcBridger::packet_video_rtmp(const uint16_t start, const
     }
 
     SrsCommonMessage rtmp;
-    SrsRtpPacket2* header = cache_video_pkts_[cache_index(start)].pkt;
+    SrsRtpPacket* header = cache_video_pkts_[cache_index(start)].pkt;
     rtmp.header.initialize_video(nb_payload, header->header.get_timestamp() / 90, 1);
     rtmp.create_payload(nb_payload);
     rtmp.size = nb_payload;
@@ -1566,7 +1566,7 @@ srs_error_t SrsRtmpFromRtcBridger::packet_video_rtmp(const uint16_t start, const
     int nalu_len = 0;
     for (uint16_t i = 0; i < cnt; ++i) {
         uint16_t index = cache_index((start + i));
-        SrsRtpPacket2* pkt = cache_video_pkts_[index].pkt;
+        SrsRtpPacket* pkt = cache_video_pkts_[index].pkt;
         cache_video_pkts_[index].in_use = false;
         cache_video_pkts_[index].pkt = NULL;
         cache_video_pkts_[index].ts = 0;
@@ -1682,7 +1682,7 @@ bool SrsRtmpFromRtcBridger::check_frame_complete(const uint16_t start, const uin
     uint16_t fu_e_c = 0;
     for (uint16_t i = 0; i < cnt; ++i) {
         int index = cache_index((start + i));
-        SrsRtpPacket2* pkt = cache_video_pkts_[index].pkt;
+        SrsRtpPacket* pkt = cache_video_pkts_[index].pkt;
         SrsRtpFUAPayload2* fua_payload = dynamic_cast<SrsRtpFUAPayload2*>(pkt->payload());
         if (fua_payload) {
             if (fua_payload->start) {
@@ -2262,11 +2262,11 @@ std::string SrsRtcRecvTrack::get_track_id()
     return track_desc_->id_;
 }
 
-srs_error_t SrsRtcRecvTrack::on_nack(SrsRtpPacket2** ppkt)
+srs_error_t SrsRtcRecvTrack::on_nack(SrsRtpPacket** ppkt)
 {
     srs_error_t err = srs_success;
 
-    SrsRtpPacket2* pkt = *ppkt;
+    SrsRtpPacket* pkt = *ppkt;
     uint16_t seq = pkt->header.get_sequence();
     SrsRtpNackInfo* nack_info = nack_receiver_->find(seq);
     if (nack_info) {
@@ -2317,7 +2317,7 @@ SrsRtcAudioRecvTrack::~SrsRtcAudioRecvTrack()
 {
 }
 
-void SrsRtcAudioRecvTrack::on_before_decode_payload(SrsRtpPacket2* pkt, SrsBuffer* buf, ISrsRtpPayloader** ppayload, SrsRtpPacketPayloadType* ppt)
+void SrsRtcAudioRecvTrack::on_before_decode_payload(SrsRtpPacket* pkt, SrsBuffer* buf, ISrsRtpPayloader** ppayload, SrsRtspPacketPayloadType* ppt)
 {
     // No payload, ignore.
     if (buf->empty()) {
@@ -2325,10 +2325,10 @@ void SrsRtcAudioRecvTrack::on_before_decode_payload(SrsRtpPacket2* pkt, SrsBuffe
     }
 
     *ppayload = new SrsRtpRawPayload();
-    *ppt = SrsRtpPacketPayloadTypeRaw;
+    *ppt = SrsRtspPacketPayloadTypeRaw;
 }
 
-srs_error_t SrsRtcAudioRecvTrack::on_rtp(SrsRtcStream* source, SrsRtpPacket2* pkt)
+srs_error_t SrsRtcAudioRecvTrack::on_rtp(SrsRtcStream* source, SrsRtpPacket* pkt)
 {
     srs_error_t err = srs_success;
 
@@ -2362,7 +2362,7 @@ SrsRtcVideoRecvTrack::~SrsRtcVideoRecvTrack()
 {
 }
 
-void SrsRtcVideoRecvTrack::on_before_decode_payload(SrsRtpPacket2* pkt, SrsBuffer* buf, ISrsRtpPayloader** ppayload, SrsRtpPacketPayloadType* ppt)
+void SrsRtcVideoRecvTrack::on_before_decode_payload(SrsRtpPacket* pkt, SrsBuffer* buf, ISrsRtpPayloader** ppayload, SrsRtspPacketPayloadType* ppt)
 {
     // No payload, ignore.
     if (buf->empty()) {
@@ -2374,17 +2374,17 @@ void SrsRtcVideoRecvTrack::on_before_decode_payload(SrsRtpPacket2* pkt, SrsBuffe
 
     if (v == kStapA) {
         *ppayload = new SrsRtpSTAPPayload();
-        *ppt = SrsRtpPacketPayloadTypeSTAP;
+        *ppt = SrsRtspPacketPayloadTypeSTAP;
     } else if (v == kFuA) {
         *ppayload = new SrsRtpFUAPayload2();
-        *ppt = SrsRtpPacketPayloadTypeFUA2;
+        *ppt = SrsRtspPacketPayloadTypeFUA2;
     } else {
         *ppayload = new SrsRtpRawPayload();
-        *ppt = SrsRtpPacketPayloadTypeRaw;
+        *ppt = SrsRtspPacketPayloadTypeRaw;
     }
 }
 
-srs_error_t SrsRtcVideoRecvTrack::on_rtp(SrsRtcStream* source, SrsRtpPacket2* pkt)
+srs_error_t SrsRtcVideoRecvTrack::on_rtp(SrsRtcStream* source, SrsRtpPacket* pkt)
 {
     srs_error_t err = srs_success;
 
@@ -2446,9 +2446,9 @@ bool SrsRtcSendTrack::has_ssrc(uint32_t ssrc)
     return track_desc_->has_ssrc(ssrc);
 }
 
-SrsRtpPacket2* SrsRtcSendTrack::fetch_rtp_packet(uint16_t seq)
+SrsRtpPacket* SrsRtcSendTrack::fetch_rtp_packet(uint16_t seq)
 {
-    SrsRtpPacket2* pkt = rtp_queue_->at(seq);
+    SrsRtpPacket* pkt = rtp_queue_->at(seq);
 
     if (pkt == NULL) {
         return pkt;
@@ -2489,11 +2489,11 @@ std::string SrsRtcSendTrack::get_track_id()
     return track_desc_->id_;
 }
 
-srs_error_t SrsRtcSendTrack::on_nack(SrsRtpPacket2** ppkt)
+srs_error_t SrsRtcSendTrack::on_nack(SrsRtpPacket** ppkt)
 {
     srs_error_t err = srs_success;
 
-    SrsRtpPacket2* pkt = *ppkt;
+    SrsRtpPacket* pkt = *ppkt;
     uint16_t seq = pkt->header.get_sequence();
 
     // insert into video_queue and audio_queue
@@ -2516,7 +2516,7 @@ srs_error_t SrsRtcSendTrack::on_recv_nack(const vector<uint16_t>& lost_seqs)
 
     for(int i = 0; i < (int)lost_seqs.size(); ++i) {
         uint16_t seq = lost_seqs.at(i);
-        SrsRtpPacket2* pkt = fetch_rtp_packet(seq);
+        SrsRtpPacket* pkt = fetch_rtp_packet(seq);
         if (pkt == NULL) {
             continue;
         }
@@ -2545,7 +2545,7 @@ SrsRtcAudioSendTrack::~SrsRtcAudioSendTrack()
 {
 }
 
-srs_error_t SrsRtcAudioSendTrack::on_rtp(SrsRtpPacket2* pkt)
+srs_error_t SrsRtcAudioSendTrack::on_rtp(SrsRtpPacket* pkt)
 {
     srs_error_t err = srs_success;
 
@@ -2573,7 +2573,7 @@ srs_error_t SrsRtcAudioSendTrack::on_rtp(SrsRtpPacket2* pkt)
     return err;
 }
 
-srs_error_t SrsRtcAudioSendTrack::on_rtcp(SrsRtpPacket2* pkt)
+srs_error_t SrsRtcAudioSendTrack::on_rtcp(SrsRtpPacket* pkt)
 {
     srs_error_t err = srs_success;
     // process rtcp
@@ -2589,7 +2589,7 @@ SrsRtcVideoSendTrack::~SrsRtcVideoSendTrack()
 {
 }
 
-srs_error_t SrsRtcVideoSendTrack::on_rtp(SrsRtpPacket2* pkt)
+srs_error_t SrsRtcVideoSendTrack::on_rtp(SrsRtpPacket* pkt)
 {
     srs_error_t err = srs_success;
 
@@ -2617,7 +2617,7 @@ srs_error_t SrsRtcVideoSendTrack::on_rtp(SrsRtpPacket2* pkt)
     return err;
 }
 
-srs_error_t SrsRtcVideoSendTrack::on_rtcp(SrsRtpPacket2* pkt)
+srs_error_t SrsRtcVideoSendTrack::on_rtcp(SrsRtpPacket* pkt)
 {
     srs_error_t err = srs_success;
     // process rtcp
diff --git a/trunk/src/app/srs_app_rtc_source.hpp b/trunk/src/app/srs_app_rtc_source.hpp
index 7aab6896ef..77cc744419 100644
--- a/trunk/src/app/srs_app_rtc_source.hpp
+++ b/trunk/src/app/srs_app_rtc_source.hpp
@@ -46,7 +46,7 @@ class SrsMessageArray;
 class SrsRtcStream;
 class SrsRtcFromRtmpBridger;
 class SrsAudioTranscoder;
-class SrsRtpPacket2;
+class SrsRtpPacket;
 class SrsSample;
 class SrsRtcStreamDescription;
 class SrsRtcTrackDescription;
@@ -88,7 +88,7 @@ class SrsRtcConsumer
 {
 private:
     SrsRtcStream* source;
-    std::vector<SrsRtpPacket2*> queue;
+    std::vector<SrsRtpPacket*> queue;
     // when source id changed, notice all consumers
     bool should_update_source_id;
     // The cond wait for mw.
@@ -107,9 +107,9 @@ class SrsRtcConsumer
     virtual void update_source_id();
     // Put RTP packet into queue.
     // @note We do not drop packet here, but drop it in sender.
-    srs_error_t enqueue(SrsRtpPacket2* pkt);
+    srs_error_t enqueue(SrsRtpPacket* pkt);
     // For RTC, we only got one packet, because there is not many packets in queue.
-    virtual srs_error_t dump_packet(SrsRtpPacket2** ppkt);
+    virtual srs_error_t dump_packet(SrsRtpPacket** ppkt);
     // Wait for at-least some messages incoming in queue.
     virtual void wait(int nb_msgs);
 public:
@@ -170,7 +170,7 @@ class ISrsRtcSourceBridger
     virtual ~ISrsRtcSourceBridger();
 public:
     virtual srs_error_t on_publish() = 0;
-    virtual srs_error_t on_rtp(SrsRtpPacket2 *pkt) = 0;
+    virtual srs_error_t on_rtp(SrsRtpPacket *pkt) = 0;
     virtual void on_unpublish() = 0;
 };
 
@@ -248,7 +248,7 @@ class SrsRtcStream : public ISrsFastTimer
     ISrsRtcPublishStream* publish_stream();
     void set_publish_stream(ISrsRtcPublishStream* v);
     // Consume the shared RTP packet, user must free it.
-    srs_error_t on_rtp(SrsRtpPacket2* pkt);
+    srs_error_t on_rtp(SrsRtpPacket* pkt);
     // Set and get stream description for souce
     bool has_stream_desc();
     void set_stream_desc(SrsRtcStreamDescription* stream_desc);
@@ -287,16 +287,16 @@ class SrsRtcFromRtmpBridger : public ISrsSourceBridger
     virtual srs_error_t on_audio(SrsSharedPtrMessage* msg);
 private:
     srs_error_t transcode(SrsAudioFrame* audio);
-    srs_error_t package_opus(SrsAudioFrame* audio, SrsRtpPacket2* pkt);
+    srs_error_t package_opus(SrsAudioFrame* audio, SrsRtpPacket* pkt);
 public:
     virtual srs_error_t on_video(SrsSharedPtrMessage* msg);
 private:
     srs_error_t filter(SrsSharedPtrMessage* msg, SrsFormat* format, bool& has_idr, std::vector<SrsSample*>& samples);
-    srs_error_t package_stap_a(SrsRtcStream* source, SrsSharedPtrMessage* msg, SrsRtpPacket2* pkt);
-    srs_error_t package_nalus(SrsSharedPtrMessage* msg, const std::vector<SrsSample*>& samples, std::vector<SrsRtpPacket2*>& pkts);
-    srs_error_t package_single_nalu(SrsSharedPtrMessage* msg, SrsSample* sample, std::vector<SrsRtpPacket2*>& pkts);
-    srs_error_t package_fu_a(SrsSharedPtrMessage* msg, SrsSample* sample, int fu_payload_size, std::vector<SrsRtpPacket2*>& pkts);
-    srs_error_t consume_packets(std::vector<SrsRtpPacket2*>& pkts);
+    srs_error_t package_stap_a(SrsRtcStream* source, SrsSharedPtrMessage* msg, SrsRtpPacket* pkt);
+    srs_error_t package_nalus(SrsSharedPtrMessage* msg, const std::vector<SrsSample*>& samples, std::vector<SrsRtpPacket*>& pkts);
+    srs_error_t package_single_nalu(SrsSharedPtrMessage* msg, SrsSample* sample, std::vector<SrsRtpPacket*>& pkts);
+    srs_error_t package_fu_a(SrsSharedPtrMessage* msg, SrsSample* sample, int fu_payload_size, std::vector<SrsRtpPacket*>& pkts);
+    srs_error_t consume_packets(std::vector<SrsRtpPacket*>& pkts);
 };
 
 class SrsRtmpFromRtcBridger : public ISrsRtcSourceBridger
@@ -315,7 +315,7 @@ class SrsRtmpFromRtcBridger : public ISrsRtcSourceBridger
         bool in_use;
         uint16_t sn;
         uint32_t ts;
-        SrsRtpPacket2* pkt;
+        SrsRtpPacket* pkt;
     };
     const static uint16_t s_cache_size = 512;
     RtcPacketCache cache_video_pkts_[s_cache_size];
@@ -329,13 +329,13 @@ class SrsRtmpFromRtcBridger : public ISrsRtcSourceBridger
     srs_error_t initialize(SrsRequest* r);
 public:
     virtual srs_error_t on_publish();
-    virtual srs_error_t on_rtp(SrsRtpPacket2 *pkt);
+    virtual srs_error_t on_rtp(SrsRtpPacket *pkt);
     virtual void on_unpublish();
 private:
-    srs_error_t trancode_audio(SrsRtpPacket2 *pkt);
+    srs_error_t trancode_audio(SrsRtpPacket *pkt);
     void packet_aac(SrsCommonMessage* audio, char* data, int len, uint32_t pts, bool is_header);
-    srs_error_t packet_video(SrsRtpPacket2* pkt);
-    srs_error_t packet_video_key_frame(SrsRtpPacket2* pkt);
+    srs_error_t packet_video(SrsRtpPacket* pkt);
+    srs_error_t packet_video_key_frame(SrsRtpPacket* pkt);
     srs_error_t packet_video_rtmp(const uint16_t start, const uint16_t end);
     int32_t find_next_lost_sn(uint16_t current_sn, uint16_t& end_sn);
     void clear_cached_video();
@@ -554,35 +554,35 @@ class SrsRtcRecvTrack
 public:
     // Note that we can set the pkt to NULL to avoid copy, for example, if the NACK cache the pkt and
     // set to NULL, nack nerver copy it but set the pkt to NULL.
-    srs_error_t on_nack(SrsRtpPacket2** ppkt);
+    srs_error_t on_nack(SrsRtpPacket** ppkt);
 public:
-    virtual srs_error_t on_rtp(SrsRtcStream* source, SrsRtpPacket2* pkt) = 0;
+    virtual srs_error_t on_rtp(SrsRtcStream* source, SrsRtpPacket* pkt) = 0;
     virtual srs_error_t check_send_nacks() = 0;
 protected:
     virtual srs_error_t do_check_send_nacks(uint32_t& timeout_nacks);
 };
 
-class SrsRtcAudioRecvTrack : public SrsRtcRecvTrack, public ISrsRtpPacketDecodeHandler
+class SrsRtcAudioRecvTrack : public SrsRtcRecvTrack, public ISrsRtspPacketDecodeHandler
 {
 public:
     SrsRtcAudioRecvTrack(SrsRtcConnection* session, SrsRtcTrackDescription* track_desc);
     virtual ~SrsRtcAudioRecvTrack();
 public:
-    virtual void on_before_decode_payload(SrsRtpPacket2* pkt, SrsBuffer* buf, ISrsRtpPayloader** ppayload, SrsRtpPacketPayloadType* ppt);
+    virtual void on_before_decode_payload(SrsRtpPacket* pkt, SrsBuffer* buf, ISrsRtpPayloader** ppayload, SrsRtspPacketPayloadType* ppt);
 public:
-    virtual srs_error_t on_rtp(SrsRtcStream* source, SrsRtpPacket2* pkt);
+    virtual srs_error_t on_rtp(SrsRtcStream* source, SrsRtpPacket* pkt);
     virtual srs_error_t check_send_nacks();
 };
 
-class SrsRtcVideoRecvTrack : public SrsRtcRecvTrack, public ISrsRtpPacketDecodeHandler
+class SrsRtcVideoRecvTrack : public SrsRtcRecvTrack, public ISrsRtspPacketDecodeHandler
 {
 public:
     SrsRtcVideoRecvTrack(SrsRtcConnection* session, SrsRtcTrackDescription* stream_descs);
     virtual ~SrsRtcVideoRecvTrack();
 public:
-    virtual void on_before_decode_payload(SrsRtpPacket2* pkt, SrsBuffer* buf, ISrsRtpPayloader** ppayload, SrsRtpPacketPayloadType* ppt);
+    virtual void on_before_decode_payload(SrsRtpPacket* pkt, SrsBuffer* buf, ISrsRtpPayloader** ppayload, SrsRtspPacketPayloadType* ppt);
 public:
-    virtual srs_error_t on_rtp(SrsRtcStream* source, SrsRtpPacket2* pkt);
+    virtual srs_error_t on_rtp(SrsRtcStream* source, SrsRtpPacket* pkt);
     virtual srs_error_t check_send_nacks();
 };
 
@@ -608,17 +608,17 @@ class SrsRtcSendTrack
     // SrsRtcSendTrack::set_nack_no_copy
     void set_nack_no_copy(bool v) { nack_no_copy_ = v; }
     bool has_ssrc(uint32_t ssrc);
-    SrsRtpPacket2* fetch_rtp_packet(uint16_t seq);
+    SrsRtpPacket* fetch_rtp_packet(uint16_t seq);
     bool set_track_status(bool active);
     bool get_track_status();
     std::string get_track_id();
 public:
     // Note that we can set the pkt to NULL to avoid copy, for example, if the NACK cache the pkt and
     // set to NULL, nack nerver copy it but set the pkt to NULL.
-    srs_error_t on_nack(SrsRtpPacket2** ppkt);
+    srs_error_t on_nack(SrsRtpPacket** ppkt);
 public:
-    virtual srs_error_t on_rtp(SrsRtpPacket2* pkt) = 0;
-    virtual srs_error_t on_rtcp(SrsRtpPacket2* pkt) = 0;
+    virtual srs_error_t on_rtp(SrsRtpPacket* pkt) = 0;
+    virtual srs_error_t on_rtcp(SrsRtpPacket* pkt) = 0;
     virtual srs_error_t on_recv_nack(const std::vector<uint16_t>& lost_seqs);
 };
 
@@ -628,8 +628,8 @@ class SrsRtcAudioSendTrack : public SrsRtcSendTrack
     SrsRtcAudioSendTrack(SrsRtcConnection* session, SrsRtcTrackDescription* track_desc);
     virtual ~SrsRtcAudioSendTrack();
 public:
-    virtual srs_error_t on_rtp(SrsRtpPacket2* pkt);
-    virtual srs_error_t on_rtcp(SrsRtpPacket2* pkt);
+    virtual srs_error_t on_rtp(SrsRtpPacket* pkt);
+    virtual srs_error_t on_rtcp(SrsRtpPacket* pkt);
 };
 
 class SrsRtcVideoSendTrack : public SrsRtcSendTrack
@@ -638,8 +638,8 @@ class SrsRtcVideoSendTrack : public SrsRtcSendTrack
     SrsRtcVideoSendTrack(SrsRtcConnection* session, SrsRtcTrackDescription* track_desc);
     virtual ~SrsRtcVideoSendTrack();
 public:
-    virtual srs_error_t on_rtp(SrsRtpPacket2* pkt);
-    virtual srs_error_t on_rtcp(SrsRtpPacket2* pkt);
+    virtual srs_error_t on_rtp(SrsRtpPacket* pkt);
+    virtual srs_error_t on_rtcp(SrsRtpPacket* pkt);
 };
 
 class SrsRtcSSRCGenerator
diff --git a/trunk/src/app/srs_app_rtsp.cpp b/trunk/src/app/srs_app_rtsp.cpp
index 829bd16571..3f17c6cb25 100644
--- a/trunk/src/app/srs_app_rtsp.cpp
+++ b/trunk/src/app/srs_app_rtsp.cpp
@@ -53,7 +53,7 @@ SrsRtpConn::SrsRtpConn(SrsRtspConn* r, int p, int sid)
     stream_id = sid;
     // TODO: support listen at <[ip:]port>
     listener = new SrsUdpListener(this, srs_any_address_for_listener(), p);
-    cache = new SrsRtpPacket();
+    cache = new SrsRtspPacket();
     pprint = SrsPithyPrint::create_caster();
 }
 
@@ -83,14 +83,14 @@ srs_error_t SrsRtpConn::on_udp_packet(const sockaddr* from, const int fromlen, c
     if (true) {
         SrsBuffer stream(buf, nb_buf);
         
-        SrsRtpPacket pkt;
+        SrsRtspPacket pkt;
         if ((err = pkt.decode(&stream)) != srs_success) {
             return srs_error_wrap(err, "decode");
         }
         
         if (pkt.chunked) {
             if (!cache) {
-                cache = new SrsRtpPacket();
+                cache = new SrsRtspPacket();
             }
             cache->copy(&pkt);
             cache->payload->append(pkt.payload->bytes(), pkt.payload->length());
@@ -106,7 +106,7 @@ srs_error_t SrsRtpConn::on_udp_packet(const sockaddr* from, const int fromlen, c
             }
         } else {
             srs_freep(cache);
-            cache = new SrsRtpPacket();
+            cache = new SrsRtspPacket();
             cache->reap(&pkt);
         }
     }
@@ -119,7 +119,7 @@ srs_error_t SrsRtpConn::on_udp_packet(const sockaddr* from, const int fromlen, c
     }
     
     // always free it.
-    SrsAutoFree(SrsRtpPacket, cache);
+    SrsAutoFree(SrsRtspPacket, cache);
 
     err = rtsp->on_rtp_packet(cache, stream_id);
     if (err != srs_success) {
@@ -380,7 +380,7 @@ srs_error_t SrsRtspConn::do_cycle()
     return err;
 }
 
-srs_error_t SrsRtspConn::on_rtp_packet(SrsRtpPacket* pkt, int stream_id)
+srs_error_t SrsRtspConn::on_rtp_packet(SrsRtspPacket* pkt, int stream_id)
 {
     srs_error_t err = srs_success;
     
@@ -438,7 +438,7 @@ srs_error_t SrsRtspConn::cycle()
     return err;
 }
 
-srs_error_t SrsRtspConn::on_rtp_video(SrsRtpPacket* pkt, int64_t dts, int64_t pts)
+srs_error_t SrsRtspConn::on_rtp_video(SrsRtspPacket* pkt, int64_t dts, int64_t pts)
 {
     srs_error_t err = srs_success;
     
@@ -457,7 +457,7 @@ srs_error_t SrsRtspConn::on_rtp_video(SrsRtpPacket* pkt, int64_t dts, int64_t pt
     return err;
 }
 
-srs_error_t SrsRtspConn::on_rtp_audio(SrsRtpPacket* pkt, int64_t dts)
+srs_error_t SrsRtspConn::on_rtp_audio(SrsRtspPacket* pkt, int64_t dts)
 {
     srs_error_t err = srs_success;
     
@@ -476,7 +476,7 @@ srs_error_t SrsRtspConn::on_rtp_audio(SrsRtpPacket* pkt, int64_t dts)
     return err;
 }
 
-srs_error_t SrsRtspConn::kickoff_audio_cache(SrsRtpPacket* pkt, int64_t dts)
+srs_error_t SrsRtspConn::kickoff_audio_cache(SrsRtspPacket* pkt, int64_t dts)
 {
     srs_error_t err = srs_success;
     
diff --git a/trunk/src/app/srs_app_rtsp.hpp b/trunk/src/app/srs_app_rtsp.hpp
index 7009751848..a41dd0c662 100644
--- a/trunk/src/app/srs_app_rtsp.hpp
+++ b/trunk/src/app/srs_app_rtsp.hpp
@@ -39,7 +39,7 @@ class SrsRtspConn;
 class SrsRtspStack;
 class SrsRtspCaster;
 class SrsConfDirective;
-class SrsRtpPacket;
+class SrsRtspPacket;
 class SrsRequest;
 class SrsStSocket;
 class SrsRtmpClient;
@@ -60,7 +60,7 @@ class SrsRtpConn: public ISrsUdpHandler
     SrsPithyPrint* pprint;
     SrsUdpListener* listener;
     SrsRtspConn* rtsp;
-    SrsRtpPacket* cache;
+    SrsRtspPacket* cache;
     int stream_id;
     int _port;
 public:
@@ -153,14 +153,14 @@ class SrsRtspConn : public ISrsCoroutineHandler, public ISrsConnection
     virtual srs_error_t do_cycle();
 // internal methods
 public:
-    virtual srs_error_t on_rtp_packet(SrsRtpPacket* pkt, int stream_id);
+    virtual srs_error_t on_rtp_packet(SrsRtspPacket* pkt, int stream_id);
 // Interface ISrsOneCycleThreadHandler
 public:
     virtual srs_error_t cycle();
 private:
-    virtual srs_error_t on_rtp_video(SrsRtpPacket* pkt, int64_t dts, int64_t pts);
-    virtual srs_error_t on_rtp_audio(SrsRtpPacket* pkt, int64_t dts);
-    virtual srs_error_t kickoff_audio_cache(SrsRtpPacket* pkt, int64_t dts);
+    virtual srs_error_t on_rtp_video(SrsRtspPacket* pkt, int64_t dts, int64_t pts);
+    virtual srs_error_t on_rtp_audio(SrsRtspPacket* pkt, int64_t dts);
+    virtual srs_error_t kickoff_audio_cache(SrsRtspPacket* pkt, int64_t dts);
 private:
     virtual srs_error_t write_sequence_header();
     virtual srs_error_t write_h264_sps_pps(uint32_t dts, uint32_t pts);
diff --git a/trunk/src/core/srs_core_version4.hpp b/trunk/src/core/srs_core_version4.hpp
index be8eb9c81a..47bad09fc8 100644
--- a/trunk/src/core/srs_core_version4.hpp
+++ b/trunk/src/core/srs_core_version4.hpp
@@ -26,6 +26,6 @@
 
 #define VERSION_MAJOR       4
 #define VERSION_MINOR       0
-#define VERSION_REVISION    110
+#define VERSION_REVISION    111
 
 #endif
diff --git a/trunk/src/kernel/srs_kernel_rtc_rtp.cpp b/trunk/src/kernel/srs_kernel_rtc_rtp.cpp
index 88cacc5afb..c3dbc7b947 100644
--- a/trunk/src/kernel/srs_kernel_rtc_rtp.cpp
+++ b/trunk/src/kernel/srs_kernel_rtc_rtp.cpp
@@ -760,18 +760,18 @@ ISrsRtpPayloader::~ISrsRtpPayloader()
 {
 }
 
-ISrsRtpPacketDecodeHandler::ISrsRtpPacketDecodeHandler()
+ISrsRtspPacketDecodeHandler::ISrsRtspPacketDecodeHandler()
 {
 }
 
-ISrsRtpPacketDecodeHandler::~ISrsRtpPacketDecodeHandler()
+ISrsRtspPacketDecodeHandler::~ISrsRtspPacketDecodeHandler()
 {
 }
 
-SrsRtpPacket2::SrsRtpPacket2()
+SrsRtpPacket::SrsRtpPacket()
 {
     payload_ = NULL;
-    payload_type_ = SrsRtpPacketPayloadTypeUnknown;
+    payload_type_ = SrsRtspPacketPayloadTypeUnknown;
     shared_buffer_ = NULL;
     actual_buffer_size_ = 0;
 
@@ -783,13 +783,13 @@ SrsRtpPacket2::SrsRtpPacket2()
     ++_srs_pps_objs_rtps->sugar;
 }
 
-SrsRtpPacket2::~SrsRtpPacket2()
+SrsRtpPacket::~SrsRtpPacket()
 {
     srs_freep(payload_);
     srs_freep(shared_buffer_);
 }
 
-char* SrsRtpPacket2::wrap(int size)
+char* SrsRtpPacket::wrap(int size)
 {
     // The buffer size is larger or equals to the size of packet.
     actual_buffer_size_ = size;
@@ -814,14 +814,14 @@ char* SrsRtpPacket2::wrap(int size)
     return shared_buffer_->payload;
 }
 
-char* SrsRtpPacket2::wrap(char* data, int size)
+char* SrsRtpPacket::wrap(char* data, int size)
 {
     char* buf = wrap(size);
     memcpy(buf, data, size);
     return buf;
 }
 
-char* SrsRtpPacket2::wrap(SrsSharedPtrMessage* msg)
+char* SrsRtpPacket::wrap(SrsSharedPtrMessage* msg)
 {
     // Generally, the wrap(msg) is used for RTMP to RTC, which is not generated by RTC,
     // so we do not recycle the msg. It's ok to directly free the msg, event the msg is
@@ -836,9 +836,9 @@ char* SrsRtpPacket2::wrap(SrsSharedPtrMessage* msg)
     return msg->payload;
 }
 
-SrsRtpPacket2* SrsRtpPacket2::copy()
+SrsRtpPacket* SrsRtpPacket::copy()
 {
-    SrsRtpPacket2* cp = new SrsRtpPacket2();
+    SrsRtpPacket* cp = new SrsRtpPacket();
 
     // We got packet from cache, the payload and message MUST be NULL,
     // because we had clear it in recycle.
@@ -861,7 +861,7 @@ SrsRtpPacket2* SrsRtpPacket2::copy()
     return cp;
 }
 
-void SrsRtpPacket2::set_padding(int size)
+void SrsRtpPacket::set_padding(int size)
 {
     header.set_padding(size);
     if (cached_payload_size) {
@@ -869,7 +869,7 @@ void SrsRtpPacket2::set_padding(int size)
     }
 }
 
-void SrsRtpPacket2::add_padding(int size)
+void SrsRtpPacket::add_padding(int size)
 {
     header.set_padding(header.get_padding() + size);
     if (cached_payload_size) {
@@ -877,22 +877,22 @@ void SrsRtpPacket2::add_padding(int size)
     }
 }
 
-void SrsRtpPacket2::set_decode_handler(ISrsRtpPacketDecodeHandler* h)
+void SrsRtpPacket::set_decode_handler(ISrsRtspPacketDecodeHandler* h)
 {
     decode_handler = h;
 }
 
-bool SrsRtpPacket2::is_audio()
+bool SrsRtpPacket::is_audio()
 {
     return frame_type == SrsFrameTypeAudio;
 }
 
-void SrsRtpPacket2::set_extension_types(SrsRtpExtensionTypes* v)
+void SrsRtpPacket::set_extension_types(SrsRtpExtensionTypes* v)
 {
     return header.set_extensions(v);
 }
 
-uint64_t SrsRtpPacket2::nb_bytes()
+uint64_t SrsRtpPacket::nb_bytes()
 {
     if (!cached_payload_size) {
         int nn_payload = (payload_? payload_->nb_bytes():0);
@@ -901,7 +901,7 @@ uint64_t SrsRtpPacket2::nb_bytes()
     return cached_payload_size;
 }
 
-srs_error_t SrsRtpPacket2::encode(SrsBuffer* buf)
+srs_error_t SrsRtpPacket::encode(SrsBuffer* buf)
 {
     srs_error_t err = srs_success;
 
@@ -925,7 +925,7 @@ srs_error_t SrsRtpPacket2::encode(SrsBuffer* buf)
     return err;
 }
 
-srs_error_t SrsRtpPacket2::decode(SrsBuffer* buf)
+srs_error_t SrsRtpPacket::decode(SrsBuffer* buf)
 {
     srs_error_t err = srs_success;
 
@@ -949,7 +949,7 @@ srs_error_t SrsRtpPacket2::decode(SrsBuffer* buf)
     // By default, we always use the RAW payload.
     if (!payload_) {
         payload_ = new SrsRtpRawPayload();
-        payload_type_ = SrsRtpPacketPayloadTypeRaw;
+        payload_type_ = SrsRtspPacketPayloadTypeRaw;
     }
 
     if ((err = payload_->decode(buf)) != srs_success) {
@@ -959,7 +959,7 @@ srs_error_t SrsRtpPacket2::decode(SrsBuffer* buf)
     return err;
 }
 
-bool SrsRtpPacket2::is_keyframe()
+bool SrsRtpPacket::is_keyframe()
 {
     // False if audio packet
     if(SrsFrameTypeAudio == frame_type) {
diff --git a/trunk/src/kernel/srs_kernel_rtc_rtp.hpp b/trunk/src/kernel/srs_kernel_rtc_rtp.hpp
index 99589d723c..44db2ed2e2 100644
--- a/trunk/src/kernel/srs_kernel_rtc_rtp.hpp
+++ b/trunk/src/kernel/srs_kernel_rtc_rtp.hpp
@@ -33,7 +33,7 @@
 #include <list>
 #include <vector>
 
-class SrsRtpPacket2;
+class SrsRtpPacket;
 
 // The RTP packet max size, should never exceed this size.
 const int kRtpPacketSize        = 1500;
@@ -262,35 +262,35 @@ class ISrsRtpPayloader : public ISrsCodec
 };
 
 // The payload type, for performance to avoid dynamic cast.
-enum SrsRtpPacketPayloadType
+enum SrsRtspPacketPayloadType
 {
-    SrsRtpPacketPayloadTypeRaw,
-    SrsRtpPacketPayloadTypeFUA2,
-    SrsRtpPacketPayloadTypeFUA,
-    SrsRtpPacketPayloadTypeNALU,
-    SrsRtpPacketPayloadTypeSTAP,
-    SrsRtpPacketPayloadTypeUnknown,
+    SrsRtspPacketPayloadTypeRaw,
+    SrsRtspPacketPayloadTypeFUA2,
+    SrsRtspPacketPayloadTypeFUA,
+    SrsRtspPacketPayloadTypeNALU,
+    SrsRtspPacketPayloadTypeSTAP,
+    SrsRtspPacketPayloadTypeUnknown,
 };
 
-class ISrsRtpPacketDecodeHandler
+class ISrsRtspPacketDecodeHandler
 {
 public:
-    ISrsRtpPacketDecodeHandler();
-    virtual ~ISrsRtpPacketDecodeHandler();
+    ISrsRtspPacketDecodeHandler();
+    virtual ~ISrsRtspPacketDecodeHandler();
 public:
     // We don't know the actual payload, so we depends on external handler.
-    virtual void on_before_decode_payload(SrsRtpPacket2* pkt, SrsBuffer* buf, ISrsRtpPayloader** ppayload, SrsRtpPacketPayloadType* ppt) = 0;
+    virtual void on_before_decode_payload(SrsRtpPacket* pkt, SrsBuffer* buf, ISrsRtpPayloader** ppayload, SrsRtspPacketPayloadType* ppt) = 0;
 };
 
 // The RTP packet with cached shared message.
-class SrsRtpPacket2
+class SrsRtpPacket
 {
 // RTP packet fields.
 public:
     SrsRtpHeader header;
 private:
     ISrsRtpPayloader* payload_;
-    SrsRtpPacketPayloadType payload_type_;
+    SrsRtspPacketPayloadType payload_type_;
 private:
     // The original shared message, all RTP packets can refer to its data.
     // Note that the size of shared msg, is not the packet size, it's a larger aligned buffer.
@@ -310,10 +310,10 @@ class SrsRtpPacket2
     // The cached payload size for packet.
     int cached_payload_size;
     // The helper handler for decoder, use RAW payload if NULL.
-    ISrsRtpPacketDecodeHandler* decode_handler;
+    ISrsRtspPacketDecodeHandler* decode_handler;
 public:
-    SrsRtpPacket2();
-    virtual ~SrsRtpPacket2();
+    SrsRtpPacket();
+    virtual ~SrsRtpPacket();
 public:
     // Wrap buffer to shared_message, which is managed by us.
     char* wrap(int size);
@@ -321,20 +321,20 @@ class SrsRtpPacket2
     // Wrap the shared message, we copy it.
     char* wrap(SrsSharedPtrMessage* msg);
     // Copy the RTP packet.
-    virtual SrsRtpPacket2* copy();
+    virtual SrsRtpPacket* copy();
 public:
     // Parse the TWCC extension, ignore by default.
-    void enable_twcc_decode() { header.enable_twcc_decode(); } // SrsRtpPacket2::enable_twcc_decode
+    void enable_twcc_decode() { header.enable_twcc_decode(); } // SrsRtpPacket::enable_twcc_decode
     // Get and set the payload of packet.
     // @remark Note that return NULL if no payload.
-    void set_payload(ISrsRtpPayloader* p, SrsRtpPacketPayloadType pt) { payload_ = p; payload_type_ = pt; }
+    void set_payload(ISrsRtpPayloader* p, SrsRtspPacketPayloadType pt) { payload_ = p; payload_type_ = pt; }
     ISrsRtpPayloader* payload() { return payload_; }
     // Set the padding of RTP packet.
     void set_padding(int size);
     // Increase the padding of RTP packet.
     void add_padding(int size);
     // Set the decode handler.
-    void set_decode_handler(ISrsRtpPacketDecodeHandler* h);
+    void set_decode_handler(ISrsRtspPacketDecodeHandler* h);
     // Whether the packet is Audio packet.
     bool is_audio();
     // Set RTP header extensions for encoding or decoding header extension
diff --git a/trunk/src/protocol/srs_rtsp_stack.cpp b/trunk/src/protocol/srs_rtsp_stack.cpp
index 8bc1937d5e..8755dcea93 100644
--- a/trunk/src/protocol/srs_rtsp_stack.cpp
+++ b/trunk/src/protocol/srs_rtsp_stack.cpp
@@ -120,7 +120,7 @@ std::string srs_generate_rtsp_method_str(SrsRtspMethod method)
     }
 }
 
-SrsRtpPacket::SrsRtpPacket()
+SrsRtspPacket::SrsRtspPacket()
 {
     version = 2;
     padding = 0;
@@ -139,13 +139,13 @@ SrsRtpPacket::SrsRtpPacket()
     completed = false;
 }
 
-SrsRtpPacket::~SrsRtpPacket()
+SrsRtspPacket::~SrsRtspPacket()
 {
     srs_freep(payload);
     srs_freep(audio);
 }
 
-void SrsRtpPacket::copy(SrsRtpPacket* src)
+void SrsRtspPacket::copy(SrsRtspPacket* src)
 {
     version = src->version;
     padding = src->padding;
@@ -164,7 +164,7 @@ void SrsRtpPacket::copy(SrsRtpPacket* src)
     audio = new SrsAudioFrame();
 }
 
-void SrsRtpPacket::reap(SrsRtpPacket* src)
+void SrsRtspPacket::reap(SrsRtspPacket* src)
 {
     copy(src);
     
@@ -177,7 +177,7 @@ void SrsRtpPacket::reap(SrsRtpPacket* src)
     src->audio = NULL;
 }
 
-srs_error_t SrsRtpPacket::decode(SrsBuffer* stream)
+srs_error_t SrsRtspPacket::decode(SrsBuffer* stream)
 {
     srs_error_t err = srs_success;
     
@@ -212,7 +212,7 @@ srs_error_t SrsRtpPacket::decode(SrsBuffer* stream)
     return err;
 }
 
-srs_error_t SrsRtpPacket::decode_97(SrsBuffer* stream)
+srs_error_t SrsRtspPacket::decode_97(SrsBuffer* stream)
 {
     srs_error_t err = srs_success;
     
@@ -264,7 +264,7 @@ srs_error_t SrsRtpPacket::decode_97(SrsBuffer* stream)
     return err;
 }
 
-srs_error_t SrsRtpPacket::decode_96(SrsBuffer* stream)
+srs_error_t SrsRtspPacket::decode_96(SrsBuffer* stream)
 {
     srs_error_t err = srs_success;
     
diff --git a/trunk/src/protocol/srs_rtsp_stack.hpp b/trunk/src/protocol/srs_rtsp_stack.hpp
index 1000793f50..0f8359f10c 100644
--- a/trunk/src/protocol/srs_rtsp_stack.hpp
+++ b/trunk/src/protocol/srs_rtsp_stack.hpp
@@ -128,7 +128,7 @@ enum SrsRtspTokenState
 
 // The rtp packet.
 // 5. RTP Data Transfer Protocol, @see rfc3550-2003-rtp.pdf, page 12
-class SrsRtpPacket
+class SrsRtspPacket
 {
 public:
     // The version (V): 2 bits
@@ -255,13 +255,13 @@ class SrsRtpPacket
     // The audio samples, one rtp packets may contains multiple audio samples.
     SrsAudioFrame* audio;
 public:
-    SrsRtpPacket();
-    virtual ~SrsRtpPacket();
+    SrsRtspPacket();
+    virtual ~SrsRtspPacket();
 public:
     // copy the header from src.
-    virtual void copy(SrsRtpPacket* src);
+    virtual void copy(SrsRtspPacket* src);
     // reap the src to this packet, reap the payload.
-    virtual void reap(SrsRtpPacket* src);
+    virtual void reap(SrsRtspPacket* src);
     // decode rtp packet from stream.
     virtual srs_error_t decode(SrsBuffer* stream);
 private:
diff --git a/trunk/src/utest/srs_utest_rtc.cpp b/trunk/src/utest/srs_utest_rtc.cpp
index fb56237f14..77107ad391 100644
--- a/trunk/src/utest/srs_utest_rtc.cpp
+++ b/trunk/src/utest/srs_utest_rtc.cpp
@@ -731,20 +731,20 @@ VOID TEST(KernelRTCTest, NACKFetchRTPPacket)
 
     // The RTP queue will free the packet.
     if (true) {
-        SrsRtpPacket2* pkt = new SrsRtpPacket2();
+        SrsRtpPacket* pkt = new SrsRtpPacket();
         pkt->header.set_sequence(100);
         track->rtp_queue_->set(pkt->header.get_sequence(), pkt);
     }
 
     // If sequence not match, packet not found.
     if (true) {
-        SrsRtpPacket2* pkt = track->fetch_rtp_packet(10);
+        SrsRtpPacket* pkt = track->fetch_rtp_packet(10);
         EXPECT_TRUE(pkt == NULL);
     }
 
     // The sequence matched, we got the packet.
     if (true) {
-        SrsRtpPacket2* pkt = track->fetch_rtp_packet(100);
+        SrsRtpPacket* pkt = track->fetch_rtp_packet(100);
         EXPECT_TRUE(pkt != NULL);
     }
 
@@ -752,7 +752,7 @@ VOID TEST(KernelRTCTest, NACKFetchRTPPacket)
     if (true) {
         // The sequence is the "same", 1100%1000 is 100,
         // so we can also get it from the RTP queue.
-        SrsRtpPacket2* pkt = track->rtp_queue_->at(1100);
+        SrsRtpPacket* pkt = track->rtp_queue_->at(1100);
         EXPECT_TRUE(pkt != NULL);
 
         // But the track requires exactly match, so it returns NULL.