diff --git a/conf/config.ini b/conf/config.ini
index 46bdb452..ae0ce077 100644
--- a/conf/config.ini
+++ b/conf/config.ini
@@ -313,6 +313,10 @@ externIP=
 #该端口是多线程的，同时支持客户端网络切换导致的连接迁移
 #需要注意的是，如果服务器在nat内，需要做端口映射时，必须确保外网映射端口跟该端口一致
 port=8000
+#rtc tcp服务器监听端口号，在udp 不通的情况下，会使用tcp传输数据
+#该端口是多线程的，同时支持客户端网络切换导致的连接迁移
+#需要注意的是，如果服务器在nat内，需要做端口映射时，必须确保外网映射端口跟该端口一致
+tcpPort = 8000
 #设置remb比特率，非0时关闭twcc并开启remb。该设置在rtc推流时有效，可以控制推流画质
 #目前已经实现twcc自动调整码率，关闭remb根据真实网络状况调整码率
 rembBitRate=0
diff --git a/server/main.cpp b/server/main.cpp
index f41d1d10..8d3b0353 100644
--- a/server/main.cpp
+++ b/server/main.cpp
@@ -292,6 +292,7 @@ int start_main(int argc,char *argv[]) {
             return Socket::createSocket(new_poller, false);
         });
         uint16_t rtcPort = mINI::Instance()[Rtc::kPort];
+        uint16_t rtcTcpPort = mINI::Instance()[Rtc::kTcpPort];
 #endif//defined(ENABLE_WEBRTC)
 
 
@@ -338,7 +339,10 @@ int start_main(int argc,char *argv[]) {
 
 #if defined(ENABLE_WEBRTC)
             //webrtc udp服务器
-            if (rtcPort) { rtcSrv_udp->start<WebRtcSession>(rtcPort); rtcSrv_tcp->start<WebRtcSession>(rtcPort); }
+            if (rtcPort) { rtcSrv_udp->start<WebRtcSession>(rtcPort);}
+
+            if (rtcTcpPort) { rtcSrv_tcp->start<WebRtcSession>(rtcTcpPort);}
+             
 #endif//defined(ENABLE_WEBRTC)
 
 #if defined(ENABLE_SRT)
diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp
index d3d3d397..2466d9e8 100644
--- a/webrtc/Sdp.cpp
+++ b/webrtc/Sdp.cpp
@@ -1204,7 +1204,9 @@ RtcSessionSdp::Ptr RtcSession::toRtcSessionSdp() const{
         }
 
         for (auto &cand : m.candidate) {
-            sdp_media.addAttr(std::make_shared<SdpAttrCandidate>(cand));
+            if(cand.port){
+                sdp_media.addAttr(std::make_shared<SdpAttrCandidate>(cand));
+            }
         }
     }
     return ret;
diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp
index d345fddc..e2be1628 100644
--- a/webrtc/WebRtcTransport.cpp
+++ b/webrtc/WebRtcTransport.cpp
@@ -44,11 +44,14 @@ const string kRembBitRate = RTC_FIELD "rembBitRate";
 // webrtc单端口udp服务器
 const string kPort = RTC_FIELD "port";
 
+const string kTcpPort = RTC_FIELD "tcpPort";
+
 static onceToken token([]() {
     mINI::Instance()[kTimeOutSec] = 15;
     mINI::Instance()[kExternIP] = "";
     mINI::Instance()[kRembBitRate] = 0;
     mINI::Instance()[kPort] = 8000;
+    mINI::Instance()[kTcpPort] = 8000;
 });
 
 } // namespace RTC
@@ -612,6 +615,7 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
     WebRtcTransport::onRtcConfigure(configure);
 
     GET_CONFIG(uint16_t, local_port, Rtc::kPort);
+    GET_CONFIG(uint16_t, local_tcp_port, Rtc::kTcpPort);
     // 添加接收端口candidate信息
     GET_CONFIG_FUNC(std::vector<std::string>, extern_ips, Rtc::kExternIP, [](string str) {
         std::vector<std::string> ret;
@@ -624,13 +628,13 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
     if (extern_ips.empty()) {
         std::string localIp = SockUtil::get_local_ip();
         configure.addCandidate(*makeIceCandidate(localIp, local_port, 120, "udp"));
-        configure.addCandidate(*makeIceCandidate(localIp, local_port, 110, "tcp"));
+        configure.addCandidate(*makeIceCandidate(localIp, local_tcp_port, 110, "tcp"));
     } else {
         const uint32_t delta = 10;
         uint32_t priority = 100 + delta * extern_ips.size();
         for (auto ip : extern_ips) {
             configure.addCandidate(*makeIceCandidate(ip, local_port, priority + 5, "udp"));
-            configure.addCandidate(*makeIceCandidate(ip, local_port, priority, "tcp"));
+            configure.addCandidate(*makeIceCandidate(ip, local_tcp_port, priority, "tcp"));
             priority -= delta;
         }
     }
diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h
index d3a492cf..14615146 100644
--- a/webrtc/WebRtcTransport.h
+++ b/webrtc/WebRtcTransport.h
@@ -32,6 +32,7 @@ namespace mediakit {
 //RTC配置项目
 namespace Rtc {
 extern const std::string kPort;
+extern const std::string kTcpPort;
 extern const std::string kTimeOutSec;
 }//namespace RTC