diff --git a/trunk/src/app/srs_app_rtc_conn.cpp b/trunk/src/app/srs_app_rtc_conn.cpp
index 13f0d4301..ea55bf6b8 100644
--- a/trunk/src/app/srs_app_rtc_conn.cpp
+++ b/trunk/src/app/srs_app_rtc_conn.cpp
@@ -1653,6 +1653,8 @@ srs_error_t SrsRtcPublisher::on_rtp(char* buf, int nb_buf)
 
     // Decode the RTP packet from buffer.
     SrsRtpPacket2* pkt = new SrsRtpPacket2();
+    SrsAutoFree(SrsRtpPacket2, pkt);
+
     if (true) {
         pkt->set_decode_handler(this);
         pkt->shared_msg = new SrsSharedPtrMessage();
@@ -1667,7 +1669,6 @@ srs_error_t SrsRtcPublisher::on_rtp(char* buf, int nb_buf)
     // For NACK simulator, drop packet.
     if (nn_simulate_nack_drop) {
         simulate_drop_packet(&pkt->rtp_header, nb_buf);
-        srs_freep(pkt);
         return err;
     }
 
@@ -1682,7 +1683,6 @@ srs_error_t SrsRtcPublisher::on_rtp(char* buf, int nb_buf)
             return srs_error_wrap(err, "on video");
         }
     } else {
-        srs_freep(pkt);
         return srs_error_new(ERROR_RTC_RTP, "unknown ssrc=%u", ssrc);
     }
 
diff --git a/trunk/src/app/srs_app_rtc_queue.cpp b/trunk/src/app/srs_app_rtc_queue.cpp
index 21ba6c781..1f0e51cf4 100644
--- a/trunk/src/app/srs_app_rtc_queue.cpp
+++ b/trunk/src/app/srs_app_rtc_queue.cpp
@@ -83,11 +83,6 @@ void SrsRtpRingBuffer::remove(uint16_t at)
     set(at, NULL);
 }
 
-bool SrsRtpRingBuffer::overflow()
-{
-    return srs_rtp_seq_distance(begin, end) >= capacity_;
-}
-
 uint32_t SrsRtpRingBuffer::get_extended_highest_sequence()
 {
     return nn_seq_flip_backs * 65536 + end - 1;
diff --git a/trunk/src/app/srs_app_rtc_queue.hpp b/trunk/src/app/srs_app_rtc_queue.hpp
index ecda676c9..55cea88a6 100644
--- a/trunk/src/app/srs_app_rtc_queue.hpp
+++ b/trunk/src/app/srs_app_rtc_queue.hpp
@@ -84,8 +84,6 @@ public:
     // Free the packet at position.
     void set(uint16_t at, SrsRtpPacket2* pkt);
     void remove(uint16_t at);
-    // Whether queue overflow or heavy(too many packets and need clear).
-    bool overflow();
     // The highest sequence number, calculate the flip back base.
     uint32_t get_extended_highest_sequence();
     // Update the sequence, got the nack range by [first, last).
diff --git a/trunk/src/app/srs_app_rtc_source.cpp b/trunk/src/app/srs_app_rtc_source.cpp
index e37685bf5..e31824d4a 100644
--- a/trunk/src/app/srs_app_rtc_source.cpp
+++ b/trunk/src/app/srs_app_rtc_source.cpp
@@ -424,8 +424,6 @@ srs_error_t SrsRtcSource::on_rtp(SrsRtpPacket2* pkt)
 {
     srs_error_t err = srs_success;
 
-    SrsAutoFree(SrsRtpPacket2, pkt);
-
     for (int i = 0; i < (int)consumers.size(); i++) {
         SrsRtcConsumer* consumer = consumers.at(i);
         if ((err = consumer->enqueue2(pkt->copy())) != srs_success) {
@@ -679,6 +677,8 @@ srs_error_t SrsRtcFromRtmpBridger::transcode(char* adts_audio, int nn_adts_audio
         nn_max_extra_payload = srs_max(nn_max_extra_payload, size);
 
         SrsRtpPacket2* pkt = NULL;
+        SrsAutoFree(SrsRtpPacket2, pkt);
+
         if ((err = package_opus(data, size, &pkt)) != srs_success) {
             return srs_error_wrap(err, "package opus");
         }
@@ -772,6 +772,8 @@ srs_error_t SrsRtcFromRtmpBridger::filter(SrsSharedPtrMessage* msg, SrsFormat* f
     // Well, for each IDR, we append a SPS/PPS before it, which is packaged in STAP-A.
     if (msg->has_idr()) {
         SrsRtpPacket2* pkt = NULL;
+        SrsAutoFree(SrsRtpPacket2, pkt);
+
         if ((err = package_stap_a(source_, msg, &pkt)) != srs_success) {
             return srs_error_wrap(err, "package stap-a");
         }
@@ -1021,17 +1023,15 @@ srs_error_t SrsRtcFromRtmpBridger::consume_packets(vector<SrsRtpPacket2*>& pkts)
     srs_error_t err = srs_success;
 
     // TODO: FIXME: Consume a range of packets.
-    int i = 0;
-    for (; i < (int)pkts.size(); i++) {
+    for (int i = 0; i < (int)pkts.size(); i++) {
         SrsRtpPacket2* pkt = pkts[i];
-
         if ((err = source_->on_rtp(pkt)) != srs_success) {
             err = srs_error_wrap(err, "consume sps/pps");
             break;
         }
     }
 
-    for (; i < (int)pkts.size(); i++) {
+    for (int i = 0; i < (int)pkts.size(); i++) {
         SrsRtpPacket2* pkt = pkts[i];
         srs_freep(pkt);
     }
diff --git a/trunk/src/app/srs_app_rtc_source.hpp b/trunk/src/app/srs_app_rtc_source.hpp
index ea1c5ac48..6e8608690 100644
--- a/trunk/src/app/srs_app_rtc_source.hpp
+++ b/trunk/src/app/srs_app_rtc_source.hpp
@@ -150,6 +150,7 @@ public:
     // Get and set the publisher, passed to consumer to process requests such as PLI.
     SrsRtcPublisher* rtc_publisher();
     void set_rtc_publisher(SrsRtcPublisher* v);
+    // Consume the shared RTP packet, user must free it.
     srs_error_t on_rtp(SrsRtpPacket2* pkt);
     virtual srs_error_t on_audio_imp(SrsSharedPtrMessage* audio);
     // When got RTC audio message, which is encoded in opus.