diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp
index a0562e7b..b8eb3304 100644
--- a/webrtc/Sdp.cpp
+++ b/webrtc/Sdp.cpp
@@ -1311,6 +1311,10 @@ void RtcSession::checkValid() const{
     bool have_active_media = false;
     for (auto &item : media) {
         item.checkValid();
+
+        if (TrackApplication == item.type) {
+            have_active_media = true;
+        }
         switch (item.direction) {
             case RtpDirection::sendrecv:
             case RtpDirection::sendonly:
@@ -1348,6 +1352,10 @@ bool RtcSession::supportSimulcast() const {
     return false;
 }
 
+bool RtcSession::isOnlyDatachannel() const {
+    return 1 == media.size() && TrackApplication == media[0].type;
+}
+
 string const SdpConst::kTWCCRtcpFb = "transport-cc";
 string const SdpConst::kRembRtcpFb = "goog-remb";
 
@@ -1596,6 +1604,10 @@ RETRY:
 #ifdef ENABLE_SCTP
         answer_media.direction = matchDirection(offer_media.direction, configure.direction);
         answer_media.candidate = configure.candidate;
+        answer_media.ice_ufrag = configure.ice_ufrag;
+        answer_media.ice_pwd = configure.ice_pwd;
+        answer_media.fingerprint = configure.fingerprint;
+        answer_media.ice_lite = configure.ice_lite;
 #else
         answer_media.direction = RtpDirection::inactive;
 #endif
diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h
index 41e1d5d3..240a6a07 100644
--- a/webrtc/Sdp.h
+++ b/webrtc/Sdp.h
@@ -678,6 +678,7 @@ public:
     const  RtcMedia *getMedia(mediakit::TrackType type) const;
     bool supportRtcpFb(const std::string &name, mediakit::TrackType type = mediakit::TrackType::TrackVideo) const;
     bool supportSimulcast() const;
+    bool isOnlyDatachannel() const;
 
 private:
     RtcSessionSdp::Ptr toRtcSessionSdp() const;
diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp
index 1819c347..9f1dc284 100644
--- a/webrtc/WebRtcTransport.cpp
+++ b/webrtc/WebRtcTransport.cpp
@@ -377,7 +377,7 @@ void WebRtcTransportImp::onCreate() {
                 return false;
             }
             if (strong_self->_alive_ticker.elapsedTime() > timeoutSec * 1000) {
-                strong_self->onShutdown(SockException(Err_timeout, "接受rtp和rtcp超时"));
+                strong_self->onShutdown(SockException(Err_timeout, "接受rtp/rtcp/datachannel超时"));
             }
             return true;
         },
@@ -386,6 +386,15 @@ void WebRtcTransportImp::onCreate() {
     _twcc_ctx.setOnSendTwccCB([this](uint32_t ssrc, string fci) { onSendTwcc(ssrc, fci); });
 }
 
+void WebRtcTransportImp::OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) {
+    WebRtcTransport::OnDtlsTransportApplicationDataReceived(dtlsTransport, data, len);
+#ifdef ENABLE_SCTP
+    if (_answer_sdp->isOnlyDatachannel()) {
+        _alive_ticker.resetTime();
+    }
+#endif
+}
+
 WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller)
     : WebRtcTransport(poller) {
     InfoL << getIdentifier();
diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h
index c6570226..23f5880b 100644
--- a/webrtc/WebRtcTransport.h
+++ b/webrtc/WebRtcTransport.h
@@ -248,6 +248,7 @@ public:
 
 protected:
     WebRtcTransportImp(const EventPoller::Ptr &poller);
+    void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override;
     void onStartWebRTC() override;
     void onSendSockData(Buffer::Ptr buf, bool flush = true, RTC::TransportTuple *tuple = nullptr) override;
     void onCheckSdp(SdpType type, RtcSession &sdp) override;