diff --git a/CMakeLists.txt b/CMakeLists.txt
index adc5f768..f562d96d 100644
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -50,6 +50,7 @@ option(ENABLE_SERVER "Enable Server" ON)
 option(ENABLE_SERVER_LIB "Enable server as android static library" OFF)
 option(ENABLE_SRT "Enable SRT" ON)
 option(ENABLE_TESTS "Enable Tests" ON)
+option(ENABLE_SCTP "Enable SCTP" ON)
 option(ENABLE_WEBRTC "Enable WebRTC" ON)
 option(ENABLE_X264 "Enable x264" OFF)
 option(ENABLE_WEPOLL "Enable wepoll" ON)
diff --git a/src/Record/MP4Muxer.cpp b/src/Record/MP4Muxer.cpp
index 1dfa3a95..51270919 100644
--- a/src/Record/MP4Muxer.cpp
+++ b/src/Record/MP4Muxer.cpp
@@ -15,6 +15,7 @@
 #include "Extension/G711.h"
 #include "Extension/H264.h"
 #include "Extension/H265.h"
+#include "Extension/JPEG.h"
 #include "Common/config.h"
 
 using namespace std;
@@ -145,6 +146,7 @@ static uint8_t getObject(CodecId codecId) {
         case CodecAAC : return MOV_OBJECT_AAC;
         case CodecH264 : return MOV_OBJECT_H264;
         case CodecH265 : return MOV_OBJECT_HEVC;
+        case CodecJPEG : return MOV_OBJECT_JPEG;
         default : return 0;
     }
 }
@@ -302,6 +304,28 @@ bool MP4MuxerInterface::addTrack(const Track::Ptr &track) {
             break;
         }
 
+    case CodecJPEG: {
+            auto jpeg_track = dynamic_pointer_cast<JPEGTrack>(track);
+            if (!jpeg_track) {
+                WarnL << "不是JPEG Track";
+                return false;
+            }
+
+            auto track_id = mp4_writer_add_video(_mov_writter.get(),
+                                                 mp4_object,
+                                                 jpeg_track->getVideoWidth(),
+                                                 jpeg_track->getVideoHeight(),
+                                                 nullptr,
+                                                 0);
+            if (track_id < 0) {
+                WarnL << "添加JPEG Track失败:" << track_id;
+                return false;
+            }
+            _codec_to_trackid[track->getCodecId()].track_id = track_id;
+            _have_video = true;
+            break;
+        }
+
         default: WarnL << "MP4录制不支持该编码格式:" << track->getCodecName(); return false;
     }
 
diff --git a/src/Rtsp/RtpMultiCaster.cpp b/src/Rtsp/RtpMultiCaster.cpp
index 8504b83a..e4aaefa1 100644
--- a/src/Rtsp/RtpMultiCaster.cpp
+++ b/src/Rtsp/RtpMultiCaster.cpp
@@ -99,13 +99,13 @@ RtpMultiCaster::~RtpMultiCaster() {
     DebugL;
 }
 
-RtpMultiCaster::RtpMultiCaster(SocketHelper &helper, const string &local_ip, const string &vhost, const string &app, const string &stream) {
+RtpMultiCaster::RtpMultiCaster(SocketHelper &helper, const string &local_ip, const string &vhost, const string &app, const string &stream, uint32_t multicast_ip, uint16_t video_port, uint16_t audio_port) {
     auto src = dynamic_pointer_cast<RtspMediaSource>(MediaSource::find(RTSP_SCHEMA, vhost, app, stream));
     if (!src) {
         auto err = StrPrinter << "未找到媒体源:" << vhost << " " << app << " " << stream << endl;
         throw std::runtime_error(err);
     }
-    _multicast_ip = MultiCastAddressMaker::Instance().obtain();
+    _multicast_ip = (multicast_ip) ? make_shared<uint32_t>(multicast_ip) : MultiCastAddressMaker::Instance().obtain();
     if (!_multicast_ip) {
         throw std::runtime_error("获取组播地址失败");
     }
@@ -113,7 +113,7 @@ RtpMultiCaster::RtpMultiCaster(SocketHelper &helper, const string &local_ip, con
     for (auto i = 0; i < 2; ++i) {
         //创建udp socket, 数组下标为TrackType
         _udp_sock[i] = helper.createSocket();
-        if (!_udp_sock[i]->bindUdpSock(0, local_ip.data())) {
+        if (!_udp_sock[i]->bindUdpSock((i == TrackVideo) ? video_port : audio_port, local_ip.data())) {
             auto err = StrPrinter << "绑定UDP端口失败:" << local_ip << endl;
             throw std::runtime_error(err);
         }
@@ -171,11 +171,11 @@ string RtpMultiCaster::getMultiCasterIP() {
     return SockUtil::inet_ntoa(addr);
 }
 
-RtpMultiCaster::Ptr RtpMultiCaster::get(SocketHelper &helper, const string &local_ip, const string &vhost, const string &app, const string &stream) {
-    static auto on_create = [](SocketHelper &helper, const string &local_ip, const string &vhost, const string &app, const string &stream){
+RtpMultiCaster::Ptr RtpMultiCaster::get(SocketHelper &helper, const string &local_ip, const string &vhost, const string &app, const string &stream, uint32_t multicast_ip, uint16_t video_port, uint16_t audio_port) {
+    static auto on_create = [](SocketHelper &helper, const string &local_ip, const string &vhost, const string &app, const string &stream, uint32_t multicast_ip, uint16_t video_port, uint16_t audio_port){
         try {
             auto poller = helper.getPoller();
-            auto ret = RtpMultiCaster::Ptr(new RtpMultiCaster(helper, local_ip, vhost, app, stream), [poller](RtpMultiCaster *ptr) {
+            auto ret = RtpMultiCaster::Ptr(new RtpMultiCaster(helper, local_ip, vhost, app, stream, multicast_ip, video_port, audio_port), [poller](RtpMultiCaster *ptr) {
                 poller->async([ptr]() {
                     delete ptr;
                 });
@@ -194,12 +194,12 @@ RtpMultiCaster::Ptr RtpMultiCaster::get(SocketHelper &helper, const string &loca
     lock_guard<recursive_mutex> lck(g_mtx);
     auto it = g_multi_caster_map.find(strKey);
     if (it == g_multi_caster_map.end()) {
-        return on_create(helper, local_ip, vhost, app, stream);
+        return on_create(helper, local_ip, vhost, app, stream, multicast_ip, video_port, audio_port);
     }
     auto ret = it->second.lock();
     if (!ret) {
         g_multi_caster_map.erase(it);
-        return on_create(helper, local_ip, vhost, app, stream);
+        return on_create(helper, local_ip, vhost, app, stream, multicast_ip, video_port, audio_port);
     }
     return ret;
 }
diff --git a/src/Rtsp/RtpMultiCaster.h b/src/Rtsp/RtpMultiCaster.h
index 022d9d83..98ce46fe 100644
--- a/src/Rtsp/RtpMultiCaster.h
+++ b/src/Rtsp/RtpMultiCaster.h
@@ -46,14 +46,14 @@ public:
 
     ~RtpMultiCaster();
 
-    static Ptr get(toolkit::SocketHelper &helper, const std::string &local_ip, const std::string &vhost, const std::string &app, const std::string &stream);
+    static Ptr get(toolkit::SocketHelper &helper, const std::string &local_ip, const std::string &vhost, const std::string &app, const std::string &stream, uint32_t multicast_ip = 0, uint16_t video_port = 0, uint16_t audio_port = 0);
     void setDetachCB(void *listener,const onDetach &cb);
 
     std::string getMultiCasterIP();
     uint16_t getMultiCasterPort(TrackType trackType);
 
 private:
-    RtpMultiCaster(toolkit::SocketHelper &helper, const std::string &local_ip, const std::string &vhost, const std::string &app, const std::string &stream);
+    RtpMultiCaster(toolkit::SocketHelper &helper, const std::string &local_ip, const std::string &vhost, const std::string &app, const std::string &stream, uint32_t multicast_ip, uint16_t video_port, uint16_t audio_port);
 
 private:
     std::recursive_mutex _mtx;
diff --git a/src/Rtsp/RtspSession.cpp b/src/Rtsp/RtspSession.cpp
index 75f64c0e..fa1bf993 100644
--- a/src/Rtsp/RtspSession.cpp
+++ b/src/Rtsp/RtspSession.cpp
@@ -721,7 +721,7 @@ void RtspSession::handleReq_Setup(const Parser &parser) {
         break;
     case Rtsp::RTP_MULTICAST: {
         if(!_multicaster){
-            _multicaster = RtpMultiCaster::get(*this, get_local_ip(), _media_info._vhost, _media_info._app, _media_info._streamid);
+            _multicaster = RtpMultiCaster::get(*this, get_local_ip(), _media_info._vhost, _media_info._app, _media_info._streamid, _multicast_ip, _multicast_video_port, _multicast_audio_port);
             if (!_multicaster) {
                 send_NotAcceptable();
                 throw SockException(Err_shutdown, "can not get a available udp multicast socket");
diff --git a/src/Rtsp/RtspSession.h b/src/Rtsp/RtspSession.h
index df2be74f..e6831b7a 100644
--- a/src/Rtsp/RtspSession.h
+++ b/src/Rtsp/RtspSession.h
@@ -72,6 +72,22 @@ protected:
     //收到RTCP包回调
     virtual void onRtcpPacket(int track_idx, SdpTrack::Ptr &track, const char *data, size_t len);
 
+    //回复客户端
+    virtual bool sendRtspResponse(const std::string &res_code, const StrCaseMap &header = StrCaseMap(), const std::string &sdp = "", const char *protocol = "RTSP/1.0");
+    //url解析后保存的相关信息
+    MediaInfo _media_info;
+
+    ////////RTP over udp_multicast////////
+    //共享的rtp组播对象
+    RtpMultiCaster::Ptr _multicaster;
+
+    //Session号
+    std::string _sessionid;
+
+    uint32_t _multicast_ip = 0;
+    uint16_t _multicast_video_port = 0;
+    uint16_t _multicast_audio_port = 0;
+
 private:
     //处理options方法,获取服务器能力
     void handleReq_Options(const Parser &parser);
@@ -130,7 +146,6 @@ private:
     void updateRtcpContext(const RtpPacket::Ptr &rtp);
     //回复客户端
     bool sendRtspResponse(const std::string &res_code, const std::initializer_list<std::string> &header, const std::string &sdp = "", const char *protocol = "RTSP/1.0");
-    bool sendRtspResponse(const std::string &res_code, const StrCaseMap &header = StrCaseMap(), const std::string &sdp = "", const char *protocol = "RTSP/1.0");
 
     //设置socket标志
     void setSocketFlags();
@@ -149,8 +164,6 @@ private:
     uint64_t _bytes_usage = 0;
     //ContentBase
     std::string _content_base;
-    //Session号
-    std::string _sessionid;
     //记录是否需要rtsp专属鉴权，防止重复触发事件
     std::string _rtsp_realm;
     //登录认证
@@ -158,8 +171,6 @@ private:
     //用于判断客户端是否超时
     toolkit::Ticker _alive_ticker;
 
-    //url解析后保存的相关信息
-    MediaInfo _media_info;
     //rtsp推流相关绑定的源
     RtspMediaSourceImp::Ptr _push_src;
     //推流器所有权
@@ -180,9 +191,6 @@ private:
     toolkit::Socket::Ptr _rtcp_socks[2];
     //标记是否收到播放的udp打洞包,收到播放的udp打洞包后才能知道其外网udp端口号
     std::unordered_set<int> _udp_connected_flags;
-    ////////RTP over udp_multicast////////
-    //共享的rtp组播对象
-    RtpMultiCaster::Ptr _multicaster;
     ////////RTSP over HTTP  ////////
     //quicktime 请求rtsp会产生两次tcp连接，
     //一次发送 get 一次发送post，需要通过x-sessioncookie关联起来
diff --git a/webrtc/CMakeLists.txt b/webrtc/CMakeLists.txt
index f2c4f65d..a00c0d36 100644
--- a/webrtc/CMakeLists.txt
+++ b/webrtc/CMakeLists.txt
@@ -38,6 +38,7 @@ set(LINK_LIBRARIES ${SRTP_LIBRARIES})
 set(COMPILE_DEFINITIONS)
 set(INCLUDE_DIRECTORIES)
 
+if(ENABLE_SCTP)
 find_package(SCTP QUIET)
 if(SCTP_FOUND)
   message(STATUS "found library: ${SCTP_INCLUDE_DIRS} ${SCTP_LIBRARIES}")
@@ -48,6 +49,7 @@ if(SCTP_FOUND)
   list(APPEND LINK_LIBRARIES ${SCTP_LIBRARIES})
   message(STATUS "WebRTC datachannel 功能已打开")
 endif()
+endif()
 
 file(GLOB WEBRTC_SRC_LIST
   ${CMAKE_CURRENT_SOURCE_DIR}/*.cpp