diff --git a/trunk/conf/full.conf b/trunk/conf/full.conf
index a4a901ab9..0f45df751 100644
--- a/trunk/conf/full.conf
+++ b/trunk/conf/full.conf
@@ -438,6 +438,9 @@ vhost rtc.vhost.srs.com {
         # Client will send ping(STUN binding request) to server, we use it as heartbeat.
         # default: 30
         stun_timeout    30;
+        # The strick check when process stun.
+        # default: off
+        stun_strict_check on;
     }
     # whether enable min delay mode for vhost.
     # For RTC, we recommend to set to on.
diff --git a/trunk/src/app/srs_app_config.cpp b/trunk/src/app/srs_app_config.cpp
index ecb2b4f42..6868adfed 100644
--- a/trunk/src/app/srs_app_config.cpp
+++ b/trunk/src/app/srs_app_config.cpp
@@ -3876,7 +3876,7 @@ srs_error_t SrsConfig::check_normal_config()
             } else if (n == "rtc") {
                 for (int j = 0; j < (int)conf->directives.size(); j++) {
                     string m = conf->at(j)->name;
-                    if (m != "enabled" && m != "bframe" && m != "aac" && m != "stun_timeout") {
+                    if (m != "enabled" && m != "bframe" && m != "aac" && m != "stun_timeout" && m != "stun_strict_check") {
                         return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal vhost.rtc.%s of %s", m.c_str(), vhost->arg0().c_str());
                     }
                 }
@@ -4798,6 +4798,24 @@ srs_utime_t SrsConfig::get_rtc_stun_timeout(string vhost)
     return (srs_utime_t)(::atoi(conf->arg0().c_str()) * SRS_UTIME_SECONDS);
 }
 
+bool SrsConfig::get_rtc_stun_strict_check(string vhost)
+{
+    static bool DEFAULT = false;
+
+    SrsConfDirective* conf = get_rtc(vhost);
+
+    if (!conf) {
+        return DEFAULT;
+    }
+
+    conf = conf->get("stun_strict_check");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+
+    return SRS_CONF_PERFER_FALSE(conf->arg0());
+}
+
 SrsConfDirective* SrsConfig::get_vhost(string vhost, bool try_default_vhost)
 {
     srs_assert(root);
diff --git a/trunk/src/app/srs_app_config.hpp b/trunk/src/app/srs_app_config.hpp
index bf5fa6dc7..3f01c0031 100644
--- a/trunk/src/app/srs_app_config.hpp
+++ b/trunk/src/app/srs_app_config.hpp
@@ -531,6 +531,7 @@ public:
     bool get_rtc_bframe_discard(std::string vhost);
     bool get_rtc_aac_discard(std::string vhost);
     srs_utime_t get_rtc_stun_timeout(std::string vhost);
+    bool get_rtc_stun_strict_check(std::string vhost);
 
 // vhost specified section
 public:
diff --git a/trunk/src/app/srs_app_http_api.cpp b/trunk/src/app/srs_app_http_api.cpp
index 096cec496..050c40a09 100644
--- a/trunk/src/app/srs_app_http_api.cpp
+++ b/trunk/src/app/srs_app_http_api.cpp
@@ -1011,26 +1011,40 @@ srs_error_t SrsGoApiRtcPlay::exchange_sdp(const std::string& app, const std::str
             }
 
             if (local_media_desc.payload_types_.empty()) {
-                return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no valid found opus payload type");
+                return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no found valid opus payload type");
             }
 
         } else if (remote_media_desc.is_video()) {
+            std::deque<SrsMediaPayloadType> backup_payloads;
             std::vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("H264");
             for (std::vector<SrsMediaPayloadType>::iterator iter = payloads.begin(); iter != payloads.end(); ++iter) {
+                if (iter->format_specific_param_.empty()) {
+                    backup_payloads.push_front(*iter);
+                    continue;
+                }
                 H264SpecificParam h264_param;
                 if ((err = parse_h264_fmtp(iter->format_specific_param_, h264_param)) != srs_success) {
                     srs_error_reset(err); continue;
                 }
 
-                if (h264_param.packetization_mode == 1 && h264_param.level_asymmerty_allow == 1) {
+                // Try to pick the "best match" H.264 payload type.
+                if (h264_param.packetization_mode == "1" && h264_param.level_asymmerty_allow == "1") {
                     // Only choose first match H.264 payload type.
                     local_media_desc.payload_types_.push_back(*iter);
                     break;
                 }
+
+                backup_payloads.push_back(*iter);
+            }
+
+            // Try my best to pick at least one media payload type.
+            if (local_media_desc.payload_types_.empty() && ! backup_payloads.empty()) {
+                srs_warn("choose backup H.264 payload type=%d", backup_payloads.front().payload_type_);
+                local_media_desc.payload_types_.push_back(backup_payloads.front());
             }
 
             if (local_media_desc.payload_types_.empty()) {
-                return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no valid found H.264 payload type");
+                return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no found valid H.264 payload type");
             }
         }
 
diff --git a/trunk/src/app/srs_app_pithy_print.cpp b/trunk/src/app/srs_app_pithy_print.cpp
index 9aacf3bfe..cef07f1de 100644
--- a/trunk/src/app/srs_app_pithy_print.cpp
+++ b/trunk/src/app/srs_app_pithy_print.cpp
@@ -110,6 +110,8 @@ SrsPithyPrint::SrsPithyPrint(int _stage_id)
 #define SRS_CONSTS_STAGE_HTTP_STREAM_CACHE 10
 // for the ng-exec stage.
 #define SRS_CONSTS_STAGE_EXEC 11
+// for the rtc play
+#define SRS_CONSTS_STAGE_RTC_PLAY 12
 
 SrsPithyPrint* SrsPithyPrint::create_rtmp_play()
 {
@@ -166,6 +168,11 @@ SrsPithyPrint* SrsPithyPrint::create_http_stream_cache()
     return new SrsPithyPrint(SRS_CONSTS_STAGE_HTTP_STREAM_CACHE);
 }
 
+SrsPithyPrint* SrsPithyPrint::create_rtc_play()
+{
+    return new SrsPithyPrint(SRS_CONSTS_STAGE_RTC_PLAY);
+}
+
 SrsPithyPrint::~SrsPithyPrint()
 {
     leave_stage();
diff --git a/trunk/src/app/srs_app_pithy_print.hpp b/trunk/src/app/srs_app_pithy_print.hpp
index b2494b926..c3b9f1219 100644
--- a/trunk/src/app/srs_app_pithy_print.hpp
+++ b/trunk/src/app/srs_app_pithy_print.hpp
@@ -87,6 +87,7 @@ public:
     static SrsPithyPrint* create_caster();
     static SrsPithyPrint* create_http_stream();
     static SrsPithyPrint* create_http_stream_cache();
+    static SrsPithyPrint* create_rtc_play();
     virtual ~SrsPithyPrint();
 private:
     // Enter the specified stage, return the client id.
diff --git a/trunk/src/app/srs_app_rtc_conn.cpp b/trunk/src/app/srs_app_rtc_conn.cpp
index 18d3a2dba..96a4fd0d8 100644
--- a/trunk/src/app/srs_app_rtc_conn.cpp
+++ b/trunk/src/app/srs_app_rtc_conn.cpp
@@ -53,6 +53,7 @@ using namespace std;
 #include <srs_http_stack.hpp>
 #include <srs_app_http_api.hpp>
 #include <srs_app_statistic.hpp>
+#include <srs_app_pithy_print.hpp>
 
 static bool is_stun(const uint8_t* data, const int size) 
 {
@@ -520,11 +521,21 @@ srs_error_t SrsRtcSenderThread::cycle()
 
     SrsMessageArray msgs(SRS_PERF_MW_MSGS);
 
+    SrsPithyPrint* pprint = SrsPithyPrint::create_rtc_play();
+    SrsAutoFree(SrsPithyPrint, pprint);
+
     while (true) {
 		if ((err = trd->pull()) != srs_success) {
             return srs_error_wrap(err, "rtc sender thread");
         }
 
+        pprint->elapse();
+
+		if (pprint->can_print()) {
+            // TODO: FIXME:
+            // Print stat like frame/s, packet/s, loss_packets.
+        } 
+
 #ifdef SRS_PERF_QUEUE_COND_WAIT
         if (realtime) {
             // for realtime, min required msgs is 0, send when got one+ msgs.
@@ -709,7 +720,10 @@ srs_error_t SrsRtcSession::on_binding_request(SrsUdpMuxSocket* udp_mux_skt, SrsS
 {
     srs_error_t err = srs_success;
 
-    if (stun_req->get_ice_controlled()) {
+    bool strict_check = _srs_config->get_rtc_stun_strict_check(request.vhost);
+    if (strict_check && stun_req->get_ice_controlled()) {
+        // @see: https://tools.ietf.org/html/draft-ietf-ice-rfc5245bis-00#section-6.1.3.1
+        // TODO: Send 487 (Role Conflict) error response.
         return srs_error_new(ERROR_RTC_STUN, "Peer must not in ice-controlled role in ice-lite mode.");
     }
 
@@ -1381,8 +1395,8 @@ srs_error_t SrsRtcServer::cycle()
             srs_cond_wait(cond);
         }
 
-        vector<mmsghdr> mhdrs = mmhdrs;
-        mmhdrs.clear();
+        vector<mmsghdr> mhdrs;
+        mmhdrs.swap(mhdrs);
 
         mmsghdr* p = &mhdrs[0];
         for (mmsghdr* end = p + mhdrs.size(); p < end; p += max_sendmmsg) {
diff --git a/trunk/src/app/srs_app_sdp.cpp b/trunk/src/app/srs_app_sdp.cpp
index f9ec31b13..f5f4ee47f 100644
--- a/trunk/src/app/srs_app_sdp.cpp
+++ b/trunk/src/app/srs_app_sdp.cpp
@@ -76,11 +76,11 @@ srs_error_t parse_h264_fmtp(const std::string& fmtp, H264SpecificParam& h264_par
         std::vector<std::string> kv = split_str(vec[i], "=");
         if (kv.size() == 2) {
             if (kv[0] == "profile-level-id") {
-                h264_param.profile_level_id = atoi(kv[1].c_str());
+                h264_param.profile_level_id = kv[1];
             } else if (kv[0] == "packetization-mode") {
-                h264_param.packetization_mode = atoi(kv[1].c_str());
+                h264_param.packetization_mode = kv[1];
             } else if (kv[0] == "level-asymmetry-allowed") {
-                h264_param.level_asymmerty_allow = atoi(kv[1].c_str());
+                h264_param.level_asymmerty_allow = kv[1];
             } else {
                 return srs_error_new(ERROR_RTC_SDP_DECODE, "invalid h264 param=%s", kv[0].c_str());
             }
diff --git a/trunk/src/app/srs_app_sdp.hpp b/trunk/src/app/srs_app_sdp.hpp
index 839f9cb5c..c0fb95321 100644
--- a/trunk/src/app/srs_app_sdp.hpp
+++ b/trunk/src/app/srs_app_sdp.hpp
@@ -73,9 +73,9 @@ class SrsSSRCGroup
 
 struct H264SpecificParam
 {
-    int profile_level_id;
-    int packetization_mode;
-    int level_asymmerty_allow;
+    std::string profile_level_id;
+    std::string packetization_mode;
+    std::string level_asymmerty_allow;
 };
 
 extern srs_error_t parse_h264_fmtp(const std::string& fmtp, H264SpecificParam& h264_param);
diff --git a/trunk/src/kernel/srs_kernel_consts.hpp b/trunk/src/kernel/srs_kernel_consts.hpp
index 208765ae7..dc6449045 100644
--- a/trunk/src/kernel/srs_kernel_consts.hpp
+++ b/trunk/src/kernel/srs_kernel_consts.hpp
@@ -176,6 +176,8 @@
 #define SRS_CONSTS_LOG_STREAM_CASTER "SCS"
 // The nginx exec log id.
 #define SRS_CONSTS_LOG_EXEC "EXE"
+// The rtc.
+#define SRS_CONSTS_LOG_RTC "RTC"
 
 ///////////////////////////////////////////////////////////
 ///////////////////////////////////////////////////////////