diff --git a/trunk/research/players/rtc_player.html b/trunk/research/players/rtc_player.html
index 03d23bd13..4f0db27ec 100644
--- a/trunk/research/players/rtc_player.html
+++ b/trunk/research/players/rtc_player.html
@@ -48,6 +48,9 @@
     <label></label>
     <video id="rtc_media_player" controls autoplay></video>
 
+    <label></label>
+    SessionID: <span id='sessionid'></span>
+
     <footer>
         <p></p>
         <p><a href="https://github.com/ossrs/srs">SRS Team &copy; 2020</a></p>
@@ -115,6 +118,7 @@
                         if (data.code) {
                             reject(data); return;
                         }
+                        $('#sessionid').html(data.sessionid);
                         resolve(data.sdp);
                     }).fail(function(reason){
                         reject(reason);
diff --git a/trunk/research/players/rtc_publisher.html b/trunk/research/players/rtc_publisher.html
index 7945c7044..d8bce7de8 100644
--- a/trunk/research/players/rtc_publisher.html
+++ b/trunk/research/players/rtc_publisher.html
@@ -48,6 +48,9 @@
     <label></label>
     <video id="rtc_media_player" width="320" autoplay muted></video>
 
+    <label></label>
+    SessionID: <span id='sessionid'></span>
+
     <footer>
         <p></p>
         <p><a href="https://github.com/ossrs/srs">SRS Team &copy; 2020</a></p>
@@ -125,6 +128,7 @@
                         if (data.code) {
                             reject(data); return;
                         }
+                        $('#sessionid').html(data.sessionid);
                         resolve(data.sdp);
                     }).fail(function(reason){
                         reject(reason);
diff --git a/trunk/src/app/srs_app_http_api.cpp b/trunk/src/app/srs_app_http_api.cpp
index 96a04171f..51db3bbec 100644
--- a/trunk/src/app/srs_app_http_api.cpp
+++ b/trunk/src/app/srs_app_http_api.cpp
@@ -967,7 +967,7 @@ srs_error_t SrsGoApiRtcPlay::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMe
     // TODO: add candidates in response json?
 
     res->set("sdp", SrsJsonAny::str(local_sdp_str.c_str()));
-    res->set("sessionid", SrsJsonAny::str(session->id().c_str()));
+    res->set("sessionid", SrsJsonAny::str(session->username().c_str()));
 
     srs_trace("RTC username=%s, offer=%dB, answer=%dB", session->username().c_str(),
         remote_sdp_str.length(), local_sdp_str.length());
@@ -1309,7 +1309,7 @@ srs_error_t SrsGoApiRtcPublish::do_serve_http(ISrsHttpResponseWriter* w, ISrsHtt
     // TODO: add candidates in response json?
 
     res->set("sdp", SrsJsonAny::str(local_sdp_str.c_str()));
-    res->set("sessionid", SrsJsonAny::str(session->id().c_str()));
+    res->set("sessionid", SrsJsonAny::str(session->username().c_str()));
 
     srs_trace("RTC username=%s, offer=%dB, answer=%dB", session->username().c_str(),
         remote_sdp_str.length(), local_sdp_str.length());
diff --git a/trunk/src/app/srs_app_rtc_conn.cpp b/trunk/src/app/srs_app_rtc_conn.cpp
index c439fedb4..fef896939 100644
--- a/trunk/src/app/srs_app_rtc_conn.cpp
+++ b/trunk/src/app/srs_app_rtc_conn.cpp
@@ -1553,6 +1553,7 @@ SrsRtcPublisher::SrsRtcPublisher(SrsRtcSession* session)
     audio_nack_ = new SrsRtpNackForReceiver(audio_queue_, 100 * 2 / 3);
 
     source = NULL;
+    nn_simulate_nack_drop = 0;
 }
 
 SrsRtcPublisher::~SrsRtcPublisher()
@@ -1967,6 +1968,17 @@ srs_error_t SrsRtcPublisher::on_rtp(char* buf, int nb_buf)
         return srs_error_wrap(err, "decode rtp packet");
     }
 
+    // For NACK simulator, drop packet.
+    if (nn_simulate_nack_drop) {
+        SrsRtpHeader* h = &pkt->rtp_header;
+        srs_warn("RTC NACK simulator #%d drop seq=%u, ssrc=%u/%s, ts=%u, %d bytes", nn_simulate_nack_drop,
+            h->get_sequence(), h->get_ssrc(), (h->get_ssrc()==video_ssrc? "Video":"Audio"), h->get_timestamp(),
+            (int)nb_buf);
+        nn_simulate_nack_drop--;
+        srs_freep(pkt);
+        return err;
+    }
+
     uint32_t ssrc = pkt->rtp_header.get_ssrc();
     if (ssrc == audio_ssrc) {
         return on_audio(pkt);
@@ -2217,7 +2229,7 @@ srs_error_t SrsRtcPublisher::notify(int type, srs_utime_t interval, srs_utime_t
 
 void SrsRtcPublisher::simulate_nack_drop(int nn)
 {
-    // TODO: FIXME: Implements it.
+    nn_simulate_nack_drop = nn;
 }
 
 SrsRtcSession::SrsRtcSession(SrsRtcServer* s)
@@ -3359,16 +3371,17 @@ srs_error_t SrsRtcServer::create_session(
         return srs_error_new(ERROR_RTC_SOURCE_BUSY, "stream %s busy", req->get_stream_url().c_str());
     }
 
-    // TODO: FIXME: Seems not random, please check it.
     std::string local_pwd = gen_random_str(32);
     std::string local_ufrag = "";
+    // TODO: FIXME: Rename for a better name, it's not an username.
     std::string username = "";
     while (true) {
         local_ufrag = gen_random_str(8);
 
         username = local_ufrag + ":" + remote_sdp.get_ice_ufrag();
-        if (!map_username_session.count(username))
+        if (!map_username_session.count(username)) {
             break;
+        }
     }
 
     int cid = _srs_context->get_id();
diff --git a/trunk/src/app/srs_app_rtc_conn.hpp b/trunk/src/app/srs_app_rtc_conn.hpp
index 2e48e84c5..073f0f982 100644
--- a/trunk/src/app/srs_app_rtc_conn.hpp
+++ b/trunk/src/app/srs_app_rtc_conn.hpp
@@ -278,6 +278,8 @@ private:
 private:
     SrsRequest* req;
     SrsSource* source;
+    // Simulators.
+    int nn_simulate_nack_drop;
 private:
     std::map<uint32_t, uint64_t> last_sender_report_sys_time;
     std::map<uint32_t, SrsNtp> last_sender_report_ntp;
diff --git a/trunk/src/main/srs_main_server.cpp b/trunk/src/main/srs_main_server.cpp
index 307476752..fc5b19c32 100644
--- a/trunk/src/main/srs_main_server.cpp
+++ b/trunk/src/main/srs_main_server.cpp
@@ -85,6 +85,9 @@ srs_error_t do_main(int argc, char** argv)
     
     // TODO: support both little and big endian.
     srs_assert(srs_is_little_endian());
+
+    // For RTC to generating random ICE username.
+    ::srandom((unsigned long)(srs_update_system_time() | (::getpid()<<13)));
     
     // for gperf gmp or gcp,
     // should never enable it when not enabled for performance issue.