diff --git a/server/Process.cpp b/server/Process.cpp
index fab4c515..9a75ee58 100644
--- a/server/Process.cpp
+++ b/server/Process.cpp
@@ -108,10 +108,17 @@ void Process::run(const string &cmd, const string &log_file_tmp) {
         }
         fprintf(stderr, "\r\n\r\n#### pid=%d,cmd=%s #####\r\n\r\n", getpid(), cmd.data());
 
+#ifndef ANDROID
         //关闭父进程继承的fd
         for (int i = 3; i < getdtablesize(); i++) {
             ::close(i);
         }
+#else
+        //关闭父进程继承的fd
+        for (int i = 3; i < 1024; i++) {
+            ::close(i);
+        }
+#endif
 
         auto params = split(cmd, " ");
         // memory leak in child process, it's ok.
diff --git a/src/Extension/G711.h b/src/Extension/G711.h
index bcf5f677..550acbc4 100644
--- a/src/Extension/G711.h
+++ b/src/Extension/G711.h
@@ -22,7 +22,7 @@ namespace mediakit{
 class G711Track : public AudioTrackImp{
 public:
     using Ptr = std::shared_ptr<G711Track>;
-    G711Track(CodecId codecId,int sample_rate, int channels, int sample_bit) : AudioTrackImp(codecId,sample_rate,channels,sample_bit){}
+    G711Track(CodecId codecId, int sample_rate, int channels, int sample_bit) : AudioTrackImp(codecId, 8000, 1, 16) {}
 
 private:
     Sdp::Ptr getSdp() override;
diff --git a/src/Rtmp/RtmpSession.cpp b/src/Rtmp/RtmpSession.cpp
index 61c844a1..7187226d 100644
--- a/src/Rtmp/RtmpSession.cpp
+++ b/src/Rtmp/RtmpSession.cpp
@@ -415,17 +415,11 @@ void RtmpSession::onCmd_pause(AMFDecoder &dec) {
 }
 
 void RtmpSession::setMetaData(AMFDecoder &dec) {
-    if (!_publisher_src) {
-        throw std::runtime_error("not a publisher");
-    }
     std::string type = dec.load<std::string>();
     if (type != "onMetaData") {
         throw std::runtime_error("can only set metadata");
     }
-    auto metadata = dec.load<AMFValue>();
-//    dumpMetadata(metadata);
-    _publisher_src->setMetaData(metadata);
-    _set_meta_data = true;
+    _publisher_metadata = dec.load<AMFValue>();
 }
 
 void RtmpSession::onProcessCmd(AMFDecoder &dec) {
@@ -478,7 +472,8 @@ void RtmpSession::onRtmpChunk(RtmpPacket::Ptr packet) {
     case MSG_AUDIO:
     case MSG_VIDEO: {
         if (!_publisher_src) {
-            throw std::runtime_error("Not a rtmp publisher!");
+            WarnL << "Not a rtmp publisher!";
+            return;
         }
         GET_CONFIG(bool, rtmp_modify_stamp, Rtmp::kModifyStamp);
         if (rtmp_modify_stamp) {
@@ -487,9 +482,9 @@ void RtmpSession::onRtmpChunk(RtmpPacket::Ptr packet) {
             chunk_data.time_stamp = (uint32_t)dts_out;
         }
 
-        if (!_set_meta_data && !chunk_data.isCfgFrame()) {
+        if (!_set_meta_data) {
             _set_meta_data = true;
-            _publisher_src->setMetaData(TitleMeta().getMetadata());
+            _publisher_src->setMetaData(_publisher_metadata ? _publisher_metadata : TitleMeta().getMetadata());
         }
         _publisher_src->onWrite(std::move(packet));
         break;
diff --git a/src/Rtmp/RtmpSession.h b/src/Rtmp/RtmpSession.h
index 0939ee80..02f2d623 100644
--- a/src/Rtmp/RtmpSession.h
+++ b/src/Rtmp/RtmpSession.h
@@ -98,8 +98,8 @@ private:
     //数据接收超时计时器
     Ticker _ticker;
     MediaInfo _media_info;
-
     std::weak_ptr<RtmpMediaSource> _player_src;
+    AMFValue _publisher_metadata;
     std::shared_ptr<RtmpMediaSourceImp> _publisher_src;
     RtmpMediaSource::RingType::RingReader::Ptr _ring_reader;
 };
diff --git a/src/Rtp/RtpProcess.cpp b/src/Rtp/RtpProcess.cpp
index 5c888a19..a50a8ecb 100644
--- a/src/Rtp/RtpProcess.cpp
+++ b/src/Rtp/RtpProcess.cpp
@@ -145,6 +145,9 @@ bool RtpProcess::alive() {
 
 void RtpProcess::setStopCheckRtp(bool is_check){
     _stop_rtp_check = is_check;
+    if (!is_check) {
+        _last_frame_time.resetTime();
+    }
 }
 
 void RtpProcess::onDetach() {
diff --git a/webrtc/Sdp.cpp b/webrtc/Sdp.cpp
index 1fe3f987..010aa5af 100644
--- a/webrtc/Sdp.cpp
+++ b/webrtc/Sdp.cpp
@@ -1150,11 +1150,9 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){
     rtcp_mux = true;
     rtcp_rsize = false;
     group_bundle = true;
-    unified_plan = false;
     support_rtx = true;
     support_red = false;
     support_ulpfec = false;
-    support_simulcast = false;
     ice_lite = true;
     ice_trickle = true;
     ice_renomination = false;
@@ -1226,26 +1224,19 @@ void RtcConfigure::addCandidate(const SdpAttrCandidate &candidate, TrackType typ
     }
 }
 
-shared_ptr<RtcSession> RtcConfigure::createOffer(){
-    shared_ptr<RtcSession> ret = std::make_shared<RtcSession>();
-    ret->version = 0;
-    ret->origin.parse("- 0 0 IN IP4 0.0.0.0");
-    ret->session_name = "zlmediakit_webrtc_session";
-    ret->session_info = "zlmediakit_webrtc_session";
-    ret->connection.parse("IN IP4 0.0.0.0");
-    ret->msid_semantic.parse("WMS *");
-    return nullptr;
-}
-
 shared_ptr<RtcSession> RtcConfigure::createAnswer(const RtcSession &offer){
     shared_ptr<RtcSession> ret = std::make_shared<RtcSession>();
     ret->version = offer.version;
+    //todo 此处设置会话id与会话地址，貌似没什么作用
     ret->origin = offer.origin;
     ret->session_name = offer.session_name;
     ret->msid_semantic = offer.msid_semantic;
     matchMedia(ret, TrackAudio, offer.media, audio);
     matchMedia(ret, TrackVideo, offer.media, video);
     matchMedia(ret, TrackApplication, offer.media, application);
+    if (ret->media.empty()) {
+        throw std::invalid_argument("生成的answer sdp中媒体个数为0");
+    }
     return ret;
 }
 
@@ -1259,7 +1250,7 @@ void RtcConfigure::matchMedia(shared_ptr<RtcSession> &ret, TrackType type, const
                 continue;
             }
             if (offer_media.ice_lite && configure.ice_lite) {
-                WarnL << "offer sdp开启了ice_lite模式，但是answer sdp配置为不支持";
+                WarnL << "answer sdp配置为ice_lite模式，与offer sdp中的ice_lite模式冲突";
                 continue;
             }
             const RtcCodecPlan *offer_plan_ptr = nullptr;
@@ -1283,16 +1274,21 @@ void RtcConfigure::matchMedia(shared_ptr<RtcSession> &ret, TrackType type, const
             answer_media.type = offer_media.type;
             answer_media.mid = offer_media.mid;
             answer_media.proto = offer_media.proto;
+            //todo(此处设置rtp端口，貌似没什么作用)
             answer_media.port = offer_media.port;
+            //todo(此处设置rtp的ip地址，貌似没什么作用)
+            answer_media.addr = offer_media.addr;
+            //todo(此处设置rtcp地址，貌似没什么作用)
+            answer_media.rtcp_addr = offer_media.rtcp_addr;
             answer_media.rtcp_mux = offer_media.rtcp_mux && configure.rtcp_mux;
             answer_media.rtcp_rsize = offer_media.rtcp_rsize && configure.rtcp_rsize;
-            answer_media.rtcp_addr = offer_media.rtcp_addr;
             answer_media.ice_trickle = offer_media.ice_trickle && configure.ice_trickle;
             answer_media.ice_renomination = offer_media.ice_renomination && configure.ice_renomination;
             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;
+            answer_media.candidate = configure.candidate;
             switch (offer_media.role) {
                 case DtlsRole::actpass :
                 case DtlsRole::active : {
@@ -1332,7 +1328,11 @@ void RtcConfigure::matchMedia(shared_ptr<RtcSession> &ret, TrackType type, const
                 }
                 default: continue;
             }
+
+            //添加媒体plan
             answer_media.plan.emplace_back(*offer_plan_ptr);
+
+            //添加rtx,red,ulpfec plan
             if (configure.support_red || configure.support_rtx || configure.support_ulpfec) {
                 for (auto &plan : offer_media.plan) {
                     if (!strcasecmp(plan.codec.data(), "rtx")) {
@@ -1382,6 +1382,8 @@ void RtcConfigure::matchMedia(shared_ptr<RtcSession> &ret, TrackType type, const
                     offer_rtcp_fb.emplace_back(fp);
                 }
             }
+
+            //修改为我们支持的rtcp-fb类型
             answer_media.plan[0].rtcp_fb.swap(offer_rtcp_fb);
             ret->media.emplace_back(answer_media);
             return;
diff --git a/webrtc/Sdp.h b/webrtc/Sdp.h
index 895610ff..4d81c8fb 100644
--- a/webrtc/Sdp.h
+++ b/webrtc/Sdp.h
@@ -599,7 +599,6 @@ public:
     RtcSSRC rtx_ssrc;
 
     //////// simulcast ////////
-    bool simulcast{false};
     RtcSSRC rtp_ssrc_low;
     RtcSSRC rtp_ssrc_mid;
     RtcSSRC rtp_ssrc_high;
@@ -663,12 +662,9 @@ public:
         bool rtcp_mux;
         bool rtcp_rsize;
         bool group_bundle;
-        bool unified_plan;
         bool support_rtx;
         bool support_red;
         bool support_ulpfec;
-
-        bool support_simulcast;
         bool ice_lite;
         bool ice_trickle;
         bool ice_renomination;
@@ -696,7 +692,6 @@ public:
                            const SdpAttrFingerprint &fingerprint);
     void addCandidate(const SdpAttrCandidate &candidate, TrackType type = TrackInvalid);
 
-    shared_ptr<RtcSession> createOffer();
     shared_ptr<RtcSession> createAnswer(const RtcSession &offer);
 
 private:
diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp
index e82029f9..9468e3d1 100644
--- a/webrtc/WebRtcTransport.cpp
+++ b/webrtc/WebRtcTransport.cpp
@@ -88,6 +88,17 @@ std::string WebRtcTransport::getAnswerSdp(const string &offer){
 
     RtcConfigure configure;
     configure.setDefaultSetting(ice_server_->GetUsernameFragment(), ice_server_->GetPassword(), RtpDirection::recvonly, fingerprint);
+
+    SdpAttrCandidate candidate;
+    candidate.foundation = "udpcandidate";
+    candidate.component = 1;
+    candidate.transport = "udp";
+    candidate.priority = getSSRC();
+    candidate.address = getIP();
+    candidate.port = getPort();
+    candidate.type = "host";
+    configure.addCandidate(candidate);
+
     _answer_sdp = configure.createAnswer(*_offer_sdp);
 
     //设置远端dtls签名
@@ -104,23 +115,6 @@ std::string WebRtcTransport::getAnswerSdp(const string &offer){
         throw std::invalid_argument("支持group BUNDLE模式");
     }
 
-    SdpAttrCandidate candidate;
-    candidate.foundation = "udpcandidate";
-    candidate.component = 1;
-    candidate.transport = "udp";
-    candidate.priority = getSSRC();
-    candidate.address = getIP();
-    candidate.port = getPort();
-    candidate.type = "host";
-    for (auto &m :_answer_sdp->media) {
-        m.candidate.emplace_back(candidate);
-        m.port = getPort();
-        m.rtcp_addr.address = getIP();
-        m.rtcp_addr.port = getPort();
-        m.addr.address = getIP();
-    }
-    _answer_sdp->connection.address = getIP();
-
     auto str = _answer_sdp->toString();
     InfoL << "\r\n" << str;
     return str;
@@ -156,12 +150,10 @@ void WebRtcTransport::OnInputDataPacket(char *buf, size_t len, RTC::TransportTup
     }
     if (is_rtp(buf)) {
         RtpHeader *header = (RtpHeader *) buf;
-        InfoL << "rtp:" << header->dumpString(len);
         return;
     }
     if (is_rtcp(buf)) {
         RtcpHeader *header = (RtcpHeader *) buf;
-//        InfoL << "rtcp:" << header->dumpString();
         return;
     }
 }