diff --git a/trunk/conf/full.conf b/trunk/conf/full.conf
index 581e4daa3..66a8a27ab 100644
--- a/trunk/conf/full.conf
+++ b/trunk/conf/full.conf
@@ -443,6 +443,10 @@ rtc_server {
     # @remark Linux 4.18+ only, for other OS always disabled.
     # default: on
     gso             on;
+    # Whether pad first packet for GSO for padding bytes.
+    # If 0, disable padding for GSO.
+    # default: 0
+    padding         0;
 }
 
 vhost rtc.vhost.srs.com {
diff --git a/trunk/src/app/srs_app_config.cpp b/trunk/src/app/srs_app_config.cpp
index 956bc99b0..85c70f87b 100644
--- a/trunk/src/app/srs_app_config.cpp
+++ b/trunk/src/app/srs_app_config.cpp
@@ -3618,7 +3618,8 @@ srs_error_t SrsConfig::check_normal_config()
         for (int i = 0; conf && i < (int)conf->directives.size(); i++) {
             string n = conf->at(i)->name;
             if (n != "enabled" && n != "listen" && n != "dir" && n != "candidate" && n != "ecdsa"
-                && n != "sendmmsg" && n != "encrypt" && n != "reuseport" && n != "gso" && n != "merge_nalus") {
+                && n != "sendmmsg" && n != "encrypt" && n != "reuseport" && n != "gso" && n != "merge_nalus"
+                && n != "padding") {
                 return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal rtc_server.%s", n.c_str());
             }
         }
@@ -4840,6 +4841,23 @@ bool SrsConfig::get_rtc_server_gso2()
     return SRS_CONF_PERFER_TRUE(conf->arg0());
 }
 
+int SrsConfig::get_rtc_server_padding()
+{
+    static int DEFAULT = 0;
+
+    SrsConfDirective* conf = root->get("rtc_server");
+    if (!conf) {
+        return DEFAULT;
+    }
+
+    conf = conf->get("padding");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+
+    return srs_min(16, ::atoi(conf->arg0().c_str()));
+}
+
 SrsConfDirective* SrsConfig::get_rtc(string vhost)
 {
     SrsConfDirective* conf = get_vhost(vhost);
diff --git a/trunk/src/app/srs_app_config.hpp b/trunk/src/app/srs_app_config.hpp
index 96524c33a..8d31ef86e 100644
--- a/trunk/src/app/srs_app_config.hpp
+++ b/trunk/src/app/srs_app_config.hpp
@@ -535,6 +535,9 @@ public:
     virtual bool get_rtc_server_gso();
 private:
     virtual bool get_rtc_server_gso2();
+public:
+    virtual int get_rtc_server_padding();
+
 public:
     SrsConfDirective* get_rtc(std::string vhost);
     bool get_rtc_enabled(std::string vhost);
diff --git a/trunk/src/app/srs_app_rtc_conn.cpp b/trunk/src/app/srs_app_rtc_conn.cpp
index c890d5866..c6e585665 100644
--- a/trunk/src/app/srs_app_rtc_conn.cpp
+++ b/trunk/src/app/srs_app_rtc_conn.cpp
@@ -468,6 +468,7 @@ SrsRtcPackets::SrsRtcPackets(bool gso, bool merge_nalus)
     nn_rtp_pkts = 0;
     nn_audios = nn_extras = 0;
     nn_videos = nn_samples = 0;
+    nn_paddings = 0;
 }
 
 SrsRtcPackets::~SrsRtcPackets()
@@ -490,6 +491,7 @@ SrsRtcSenderThread::SrsRtcSenderThread(SrsRtcSession* s, SrsUdpMuxSocket* u, int
     sendonly_ukt = u->copy_sendonly();
     gso = false;
     merge_nalus = false;
+    max_padding = 0;
 
     audio_timestamp = 0;
     audio_sequence = 0;
@@ -519,8 +521,9 @@ srs_error_t SrsRtcSenderThread::initialize(const uint32_t& vssrc, const uint32_t
 
     gso = _srs_config->get_rtc_server_gso();
     merge_nalus = _srs_config->get_rtc_server_merge_nalus();
-    srs_trace("RTC sender video(ssrc=%d, pt=%d), audio(ssrc=%d, pt=%d), package(gso=%d, merge_nalus=%d)",
-        video_ssrc, video_payload_type, audio_ssrc, audio_payload_type, gso, merge_nalus);
+    max_padding = _srs_config->get_rtc_server_padding();
+    srs_trace("RTC sender video(ssrc=%d, pt=%d), audio(ssrc=%d, pt=%d), package(gso=%d, merge_nalus=%d), padding=%d",
+        video_ssrc, video_payload_type, audio_ssrc, audio_payload_type, gso, merge_nalus, max_padding);
 
     return err;
 }
@@ -543,6 +546,14 @@ srs_error_t SrsRtcSenderThread::on_reload_rtc_server()
         }
     }
 
+    if (true) {
+        bool v = _srs_config->get_rtc_server_padding();
+        if (max_padding != v) {
+            srs_trace("Reload padding %d=>%d", max_padding, v);
+            max_padding = v;
+        }
+    }
+
     return srs_success;
 }
 
@@ -665,7 +676,7 @@ srs_error_t SrsRtcSenderThread::cycle()
         // Stat the RTP packets going into kernel.
         stat->perf_on_gso_packets(pkts.nn_rtp_pkts);
 #if defined(SRS_DEBUG)
-        srs_trace("RTC PLAY packets, msgs %d/%d, rtp %d, gso %d, %d audios, %d extras, %d videos, %d samples, %d bytes",
+        srs_trace("RTC PLAY done, msgs %d/%d, rtp %d, gso %d, %d audios, %d extras, %d videos, %d samples, %d bytes",
             msg_count, nn_rtc_packets, pkts.packets.size(), pkts.nn_rtp_pkts, pkts.nn_audios, pkts.nn_extras, pkts.nn_videos,
             pkts.nn_samples, pkts.nn_bytes);
 #endif
@@ -673,9 +684,9 @@ srs_error_t SrsRtcSenderThread::cycle()
         pprint->elapse();
         if (pprint->can_print()) {
             // TODO: FIXME: Print stat like frame/s, packet/s, loss_packets.
-            srs_trace("-> RTC PLAY %d msgs, %d/%d packets, %d audios, %d extras, %d videos, %d samples, %d bytes",
+            srs_trace("-> RTC PLAY %d msgs, %d/%d packets, %d audios, %d extras, %d videos, %d samples, %d bytes, %d pad",
                 msg_count, pkts.packets.size(), pkts.nn_rtp_pkts, pkts.nn_audios, pkts.nn_extras, pkts.nn_videos,
-                pkts.nn_samples, pkts.nn_bytes);
+                pkts.nn_samples, pkts.nn_bytes, pkts.nn_paddings);
         }
     }
 }
@@ -862,13 +873,28 @@ srs_error_t SrsRtcSenderThread::send_packets_gso(SrsUdpMuxSocket* skt, SrsRtcPac
     int nn_packets = (int)packets.packets.size();
     for (int i = 0; i < nn_packets; i++) {
         SrsRtpPacket2* packet = packets.packets[i];
-
-        // The handler to send message.
-        mmsghdr* mhdr = NULL;
-
-        // Check whether we can use GSO to send it.
         int nn_packet = packet->nb_bytes();
 
+        SrsRtpPacket2* next_packet = NULL;
+        int nn_next_packet = 0;
+        if (i < nn_packets - 1) {
+            next_packet = (i < nn_packets - 1)? packets.packets[i + 1]:NULL;
+            nn_next_packet = next_packet? next_packet->nb_bytes() : 0;
+        }
+
+        // Padding the first packet if size is similar to the next one.
+        if (i == 0 && max_padding > 0 && next_packet && nn_packet < nn_next_packet && nn_next_packet - nn_packet < max_padding) {
+#if defined(SRS_DEBUG)
+            srs_trace("Padding %d bytes %d=>%d, packets %d, max_padding %d", nn_next_packet - nn_packet,
+                nn_packet, nn_next_packet, nn_packets, max_padding);
+#endif
+            packet->set_padding(nn_next_packet - nn_packet);
+            nn_packet = nn_next_packet;
+            packets.nn_paddings++;
+        }
+
+        // Check whether we can use GSO to send it.
+        mmsghdr* mhdr = NULL;
         if ((gso_size && gso_size == nn_packet) || (use_gso && !gso_final)) {
             use_gso = true;
             gso_final = (gso_size && gso_size != nn_packet);
@@ -893,10 +919,7 @@ srs_error_t SrsRtcSenderThread::send_packets_gso(SrsUdpMuxSocket* skt, SrsRtcPac
         }
 
         // Change the state according to the next packet.
-        if (i < nn_packets - 1) {
-            SrsRtpPacket2* next_packet = (i < nn_packets - 1)? packets.packets[i + 1]:NULL;
-            int nn_next_packet = next_packet? next_packet->nb_bytes() : 0;
-
+        if (next_packet) {
             // If GSO, but next is bigger than this one, we must enter the final state.
             if (use_gso && !gso_final) {
                 gso_final = (nn_packet < nn_next_packet);
@@ -1025,8 +1048,8 @@ srs_error_t SrsRtcSenderThread::send_packets_gso(SrsUdpMuxSocket* skt, SrsRtcPac
     }
 
 #if defined(SRS_DEBUG)
-    srs_trace("RTC PLAY summary, rtp %d/%d, videos %d/%d, audios %d/%d", packets.packets.size(),
-        packets.nn_rtp_pkts, packets.nn_videos, packets.nn_samples, packets.nn_audios, packets.nn_extras);
+    srs_trace("RTC PLAY summary, rtp %d/%d, videos %d/%d, audios %d/%d, pad %d", packets.packets.size(), packets.nn_rtp_pkts,
+        packets.nn_videos, packets.nn_samples, packets.nn_audios, packets.nn_extras, packets.nn_paddings);
 #endif
 
     return err;
diff --git a/trunk/src/app/srs_app_rtc_conn.hpp b/trunk/src/app/srs_app_rtc_conn.hpp
index 2ffe75441..978681ccb 100644
--- a/trunk/src/app/srs_app_rtc_conn.hpp
+++ b/trunk/src/app/srs_app_rtc_conn.hpp
@@ -138,6 +138,8 @@ public:
     int nn_audios;
     // The original video messages.
     int nn_videos;
+    // The number of padded packet.
+    int nn_paddings;
 public:
     std::vector<SrsRtpPacket2*> packets;
 public:
@@ -166,6 +168,7 @@ public:
     SrsUdpMuxSocket* sendonly_ukt;
     bool merge_nalus;
     bool gso;
+    int max_padding;
 public:
     SrsRtcSenderThread(SrsRtcSession* s, SrsUdpMuxSocket* u, int parent_cid);
     virtual ~SrsRtcSenderThread();