diff --git a/trunk/src/app/srs_app_rtc_conn.cpp b/trunk/src/app/srs_app_rtc_conn.cpp
index 13a03e266..85fcacd6b 100644
--- a/trunk/src/app/srs_app_rtc_conn.cpp
+++ b/trunk/src/app/srs_app_rtc_conn.cpp
@@ -494,11 +494,8 @@ SrsRtcPlayer::SrsRtcPlayer(SrsRtcSession* s, int parent_cid)
 
     session_ = s;
 
-    audio_timestamp = 0;
     audio_sequence = 0;
-
     video_sequence = 0;
-
     mw_msgs = 0;
     realtime = true;
 
@@ -711,15 +708,12 @@ srs_error_t SrsRtcPlayer::send_packets(SrsRtcSource* source, const vector<SrsRtp
         if (pkt->is_audio()) {
             info.nn_audios++;
 
-            pkt->header.set_timestamp(audio_timestamp);
             pkt->header.set_sequence(audio_sequence++);
             pkt->header.set_ssrc(audio_ssrc);
             pkt->header.set_payload_type(audio_payload_type);
 
             // TODO: FIXME: Padding audio to the max payload in RTP packets.
 
-            // TODO: FIXME: Why 960? Need Refactoring?
-            audio_timestamp += 960;
             continue;
         }
 
diff --git a/trunk/src/app/srs_app_rtc_conn.hpp b/trunk/src/app/srs_app_rtc_conn.hpp
index 583e7f8e1..fac2775b2 100644
--- a/trunk/src/app/srs_app_rtc_conn.hpp
+++ b/trunk/src/app/srs_app_rtc_conn.hpp
@@ -195,7 +195,6 @@ protected:
 private:
     // TODO: FIXME: How to handle timestamp overflow?
     // Information for audio.
-    uint32_t audio_timestamp;
     uint16_t audio_sequence;
     uint32_t audio_ssrc;
     uint16_t audio_payload_type;
diff --git a/trunk/src/app/srs_app_rtc_source.cpp b/trunk/src/app/srs_app_rtc_source.cpp
index 11a57bca5..0e5675a7d 100644
--- a/trunk/src/app/srs_app_rtc_source.cpp
+++ b/trunk/src/app/srs_app_rtc_source.cpp
@@ -424,6 +424,7 @@ SrsRtcFromRtmpBridger::SrsRtcFromRtmpBridger(SrsRtcSource* source)
     discard_bframe = false;
     merge_nalus = false;
     meta = new SrsMetaCache();
+    audio_timestamp = 0;
 }
 
 SrsRtcFromRtmpBridger::~SrsRtcFromRtmpBridger()
@@ -592,6 +593,10 @@ srs_error_t SrsRtcFromRtmpBridger::package_opus(char* data, int size, SrsRtpPack
     SrsRtpPacket2* pkt = new SrsRtpPacket2();
     pkt->frame_type = SrsFrameTypeAudio;
     pkt->header.set_marker(true);
+    pkt->header.set_timestamp(audio_timestamp);
+
+    // TODO: FIXME: Why 960? Need Refactoring?
+    audio_timestamp += 960;
 
     SrsRtpRawPayload* raw = new SrsRtpRawPayload();
     pkt->payload = raw;
diff --git a/trunk/src/app/srs_app_rtc_source.hpp b/trunk/src/app/srs_app_rtc_source.hpp
index b93a8676f..e21f6c828 100644
--- a/trunk/src/app/srs_app_rtc_source.hpp
+++ b/trunk/src/app/srs_app_rtc_source.hpp
@@ -173,6 +173,7 @@ private:
     SrsAudioRecode* codec;
     bool discard_bframe;
     bool merge_nalus;
+    uint32_t audio_timestamp;
 public:
     SrsRtcFromRtmpBridger(SrsRtcSource* source);
     virtual ~SrsRtcFromRtmpBridger();