Merge branch 'feature/rtc' into develop

README.md

@@ -159,6 +159,8 @@ For previous versions, please read:
 
 ## V4 changes
 
+* v4.0, 2020-06-24, Support static link c++ libraries. 4.0.32
+* v4.0, 2020-06-23, Change log cid from int to string. 4.0.31
 * v4.0, 2020-06-13, GB28181 with JitterBuffer support. 4.0.30
 * v4.0, 2020-06-03, Support enable C++11. 4.0.29
 * v4.0, 2020-05-31, Remove [srs-librtmp](https://github.com/ossrs/srs/issues/1535#issuecomment-633907655). 4.0.28

trunk/auto/auto_headers.sh

@@ -87,6 +87,12 @@ else
     srs_undefine_macro "SRS_RTC" $SRS_AUTO_HEADERS_H
 fi
 
+if [ $SRS_FFMPEG_FIT = YES ]; then
+    srs_define_macro "SRS_FFMPEG_FIT" $SRS_AUTO_HEADERS_H
+else
+    srs_undefine_macro "SRS_FFMPEG_FIT" $SRS_AUTO_HEADERS_H
+fi
+
 if [ $SRS_SIMULATOR = YES ]; then
     srs_define_macro "SRS_SIMULATOR" $SRS_AUTO_HEADERS_H
 else

trunk/auto/depends.sh

@@ -595,7 +595,7 @@ fi
 #####################################################################################
 # ffmpeg-fix, for WebRTC to transcode AAC with Opus.
 #####################################################################################
-if [[ $SRS_RTC == YES ]]; then
+if [[ $SRS_FFMPEG_FIT == YES ]]; then
     FFMPEG_OPTIONS=""
 
     # If disable nasm, disable all ASMs.

trunk/auto/options.sh

@@ -22,7 +22,6 @@ SRS_GB28181=NO
 SRS_CXX11=NO
 SRS_CXX14=NO
 SRS_NGINX=NO
-SRS_FFMPEG_TOOL=NO
 SRS_LIBRTMP=NO
 SRS_RESEARCH=NO
 SRS_UTEST=NO
@@ -46,8 +45,12 @@ SRS_HLS=YES
 SRS_DVR=YES
 # 
 ################################################################
-# libraries
+# FFmpeg stub is the stub code in SRS for ingester or encoder.
 SRS_FFMPEG_STUB=NO
+# FFmpeg tool is the binary for FFmpeg tool, to exec ingest or transcode.
+SRS_FFMPEG_TOOL=NO
+# FFmpeg fit is the source code for RTC, to transcode audio or video in SRS.
+SRS_FFMPEG_FIT=RESERVED
 # arguments
 SRS_PREFIX=/usr/local/srs
 SRS_JOBS=1
@@ -153,6 +156,7 @@ Features:
   --gb28181=on|off          Whether build the GB28181 support for SRS.
   --cxx11=on|off            Whether enable the C++11 support for SRS.
   --cxx14=on|off            Whether enable the C++14 support for SRS.
+  --ffmpeg-fit=on|off       Whether enable the FFmpeg fit(source code) for SRS.
 
   --prefix=<path>           The absolute installation path for srs. Default: $SRS_PREFIX
   --gcov=on|off             Whether enable the GCOV compiler options.
@@ -282,7 +286,7 @@ function parse_user_option() {
 
         --with-ffmpeg)                  SRS_FFMPEG_TOOL=YES         ;;
         --without-ffmpeg)               SRS_FFMPEG_TOOL=NO          ;;
-        --ffmpeg)                       if [[ $value == off ]]; then SRS_FFMPEG_TOOL=NO; else SRS_FFMPEG_TOOL=YES; fi    ;;
+        --ffmpeg-tool)                  if [[ $value == off ]]; then SRS_FFMPEG_TOOL=NO; else SRS_FFMPEG_TOOL=YES; fi    ;;
 
         --with-transcode)               SRS_TRANSCODE=YES           ;;
         --without-transcode)            echo "ignore option \"$option\"" ;;
@@ -327,6 +331,7 @@ function parse_user_option() {
 
         --cxx11)                        if [[ $value == off ]]; then SRS_CXX11=NO; else SRS_CXX11=YES; fi    ;;
         --cxx14)                        if [[ $value == off ]]; then SRS_CXX14=NO; else SRS_CXX14=YES; fi    ;;
+        --ffmpeg-fit)                   if [[ $value == off ]]; then SRS_FFMPEG_FIT=NO; else SRS_FFMPEG_FIT=YES; fi    ;;
 
         --with-clean)                   SRS_CLEAN=YES               ;;
         --without-clean)                SRS_CLEAN=NO                ;;
@@ -526,6 +531,11 @@ function apply_user_presets() {
     if [[ $SRS_SRT == YES ]]; then
         SRS_CXX11=YES
     fi
+
+    # Enable FFmpeg fit for RTC to trancode audio from AAC to OPUS, if user has't disabled it.
+    if [[ $SRS_RTC == YES && $SRS_FFMPEG_FIT == RESERVED ]]; then
+        SRS_FFMPEG_FIT=YES
+    fi
 }
 apply_user_presets
 
@@ -627,6 +637,7 @@ function regenerate_options() {
     if [ $SRS_GB28181 = YES ]; then         SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --gb28181=on"; else         SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --gb28181=off"; fi
     if [ $SRS_CXX11 = YES ]; then           SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --cxx11=on"; else           SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --cxx11=off"; fi
     if [ $SRS_CXX14 = YES ]; then           SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --cxx14=on"; else           SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --cxx14=off"; fi
+    if [ $SRS_FFMPEG_FIT = YES ]; then      SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --ffmpeg-fit=on"; else      SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --ffmpeg-fit=off"; fi
     if [ $SRS_NASM = YES ]; then            SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --nasm=on"; else            SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --nasm=off"; fi
     if [ $SRS_SRTP_ASM = YES ]; then        SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --srtp-nasm=on"; else       SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --srtp-nasm=off"; fi
     if [ $SRS_SENDMMSG = YES ]; then        SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --sendmmsg=on"; else        SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --sendmmsg=off"; fi

trunk/conf/full.conf

@@ -440,8 +440,8 @@ rtc_server {
     # We listen multiple times at the same port, by REUSEPORT, to increase the UDP queue.
     # Note that you can set to 1 and increase the system UDP buffer size by net.core.rmem_max
     # and net.core.rmem_default or just increase this to get larger UDP recv and send buffer.
-    # default: 4
+    # default: 1
-    reuseport       4;
+    reuseport       1;
     # Whether merge multiple NALUs into one.
     # @see https://github.com/ossrs/srs/issues/307#issuecomment-612806318
     # default: on

trunk/configure

@@ -136,20 +136,24 @@ END
 # st(state-threads) the basic network library for SRS.
 LibSTRoot="${SRS_OBJS_DIR}/st"; LibSTfile="${LibSTRoot}/libst.a"
 if [[ $SRS_SHARED_ST == YES ]]; then LibSTfile="-lst"; fi
+
 # srtp
 if [[ $SRS_RTC == YES ]]; then
   LibSrtpRoot="${SRS_OBJS_DIR}/srtp2/include"; LibSrtpFile="${SRS_OBJS_DIR}/srtp2/lib/libsrtp2.a"
 fi
+
 # FFMPEG for WebRTC transcoding, such as aac to opus.
-if [[ $SRS_RTC == YES ]]; then
+if [[ $SRS_FFMPEG_FIT == YES ]]; then
   LibFfmpegRoot="${SRS_OBJS_DIR}/ffmpeg/include"; LibFfmpegFile="${SRS_OBJS_DIR}/ffmpeg/lib/libavcodec.a ${SRS_OBJS_DIR}/ffmpeg/lib/libswresample.a ${SRS_OBJS_DIR}/ffmpeg/lib/libavutil.a"
   LibFfmpegRoot="${LibFfmpegRoot} ${SRS_OBJS_DIR}/opus/include"; LibFfmpegFile="${LibFfmpegFile} ${SRS_OBJS_DIR}/opus/lib/libopus.a"
 fi
+
 # openssl-1.1.0e, for the RTMP complex handshake.
 LibSSLRoot="";LibSSLfile=""
 if [[ $SRS_SSL == YES && $SRS_USE_SYS_SSL == NO ]]; then
     LibSSLRoot="${SRS_OBJS_DIR}/openssl/include"; LibSSLfile="${SRS_OBJS_DIR}/openssl/lib/libssl.a ${SRS_OBJS_DIR}/openssl/lib/libcrypto.a";
 fi
+
 # gperftools-2.1, for mem check and mem/cpu profile
 LibGperfRoot=""; LibGperfFile=""
 if [ $SRS_GPERF = YES ]; then
@@ -158,28 +162,35 @@ fi
 if [ $SRS_GPERF_MD = YES ]; then
     LibGperfFile="${SRS_OBJS_DIR}/gperf/lib/libtcmalloc_debug.a";
 fi
+
 # srt code path
 if [[ $SRS_SRT == YES ]]; then
     LibSRTRoot="${SRS_WORKDIR}/src/srt"; LibSRTfile="${SRS_OBJS_DIR}/srt/lib/libsrt.a"
     if [[ $SRS_SHARED_SRT == YES ]]; then LibSRTfile="-lsrt"; fi
 fi
+
 # the link options, always use static link
 SrsLinkOptions="-ldl";
 if [[ $SRS_SRT == YES || $SRS_RTC == YES ]]; then
     SrsLinkOptions="${SrsLinkOptions} -lpthread";
 fi
+
 if [[ $SRS_SSL == YES && $SRS_USE_SYS_SSL == YES ]]; then
     SrsLinkOptions="${SrsLinkOptions} -lssl -lcrypto";
 fi
-# if static specified, add static
+
-# TODO: FIXME: remove static.
+# Static link the c++ libraries, for user who build SRS by a new version of gcc,
+# so we need to link the c++ libraries staticly but not all.
+# @see https://stackoverflow.com/a/26107550
 if [ $SRS_STATIC = YES ]; then
-    SrsLinkOptions="${SrsLinkOptions} -static";
+    SrsLinkOptions="${SrsLinkOptions} -static-libstdc++";
 fi
+
 # For coverage.
 if [[ $SRS_GCOV == YES ]]; then
     SrsLinkOptions="${SrsLinkOptions} ${SrsGcov}";
 fi
+
 # For FFMPEG/RTC.
 if [[ $SRS_RTC == YES && $SRS_NASM == NO && $SRS_OSX == NO ]]; then
     SrsLinkOptions="${SrsLinkOptions} -lrt";
@@ -222,7 +233,10 @@ MODULE_FILES=("srs_protocol_amf0" "srs_protocol_io" "srs_rtmp_stack"
         "srs_service_rtmp_conn" "srs_service_utility" "srs_service_conn")
 if [[ $SRS_RTC == YES ]]; then
     MODULE_FILES+=("srs_rtc_stun_stack")
-    ModuleLibIncs+=("${LibFfmpegRoot[*]}" ${LibSrtpRoot})
+    ModuleLibIncs+=(${LibSrtpRoot})
+fi
+if [[ $SRS_FFMPEG_FIT == YES ]]; then
+    ModuleLibIncs+=("${LibFfmpegRoot[*]}")
 fi
 PROTOCOL_INCS="src/protocol"; MODULE_DIR=${PROTOCOL_INCS} . auto/modules.sh
 PROTOCOL_OBJS="${MODULE_OBJS[@]}"
@@ -246,7 +260,10 @@ if [ $SRS_GPERF = YES ]; then
     ModuleLibIncs+=(${LibGperfRoot})
 fi
 if [[ $SRS_RTC == YES ]]; then
-    ModuleLibIncs+=("${LibFfmpegRoot[*]}" ${LibSrtpRoot})
+    ModuleLibIncs+=(${LibSrtpRoot})
+fi
+if [[ $SRS_FFMPEG_FIT == YES ]]; then
+    ModuleLibIncs+=("${LibFfmpegRoot[*]}")
 fi
 MODULE_FILES=("srs_app_server" "srs_app_conn" "srs_app_rtmp_conn" "srs_app_source"
         "srs_app_refer" "srs_app_hls" "srs_app_forward" "srs_app_encoder" "srs_app_http_stream"
@@ -260,9 +277,12 @@ MODULE_FILES=("srs_app_server" "srs_app_conn" "srs_app_rtmp_conn" "srs_app_sourc
         "srs_app_hourglass" "srs_app_dash" "srs_app_fragment" "srs_app_dvr"
         "srs_app_coworkers" "srs_app_hybrid")
 if [[ $SRS_RTC == YES ]]; then
-    MODULE_FILES+=("srs_app_rtc_conn" "srs_app_rtc_dtls" "srs_app_rtc_codec" "srs_app_rtc_sdp"
+    MODULE_FILES+=("srs_app_rtc_conn" "srs_app_rtc_dtls" "srs_app_rtc_sdp"
         "srs_app_rtc_queue" "srs_app_rtc_server" "srs_app_rtc_source" "srs_app_rtc_api")
 fi
+if [[ $SRS_FFMPEG_FIT == YES ]]; then
+    MODULE_FILES+=("srs_app_rtc_codec")
+fi
 if [[ $SRS_GB28181 == YES ]]; then
     MODULE_FILES+=("srs_app_gb28181" "srs_app_gb28181_sip" "srs_app_gb28181_jitbuffer")
 fi
@@ -284,7 +304,10 @@ if [[ $SRS_SRT == YES ]]; then
 fi
 ModuleLibIncs=(${SRS_OBJS_DIR} ${LibSTRoot} ${LibGperfRoot} ${LibSSLRoot})
 if [[ $SRS_RTC == YES ]]; then
-    ModuleLibIncs+=("${LibFfmpegRoot[*]}" ${LibSrtpRoot})
+    ModuleLibIncs+=(${LibSrtpRoot})
+fi
+if [[ $SRS_FFMPEG_FIT == YES ]]; then
+    ModuleLibIncs+=("${LibFfmpegRoot[*]}")
 fi
 if [[ $SRS_SRT == YES ]]; then
     ModuleLibIncs+=("${LibSRTRoot[*]}")
@@ -298,7 +321,10 @@ MODULE_ID="MAIN"
 MODULE_DEPENDS=("CORE" "KERNEL" "PROTOCOL")
 ModuleLibIncs=(${SRS_OBJS_DIR} ${LibSTRoot} ${LibGperfRoot} ${LibSSLRoot})
 if [[ $SRS_RTC == YES ]]; then
-    ModuleLibIncs+=("${LibFfmpegRoot[*]}" ${LibSrtpRoot})
+    ModuleLibIncs+=(${LibSrtpRoot})
+fi
+if [[ $SRS_FFMPEG_FIT == YES ]]; then
+    ModuleLibIncs+=("${LibFfmpegRoot[*]}")
 fi
 MODULE_FILES=()
 DEFINES=""
@@ -325,7 +351,10 @@ done
 # all depends libraries
 ModuleLibFiles=(${LibSTfile} ${LibSSLfile} ${LibGperfFile})
 if [[ $SRS_RTC == YES ]]; then
-    ModuleLibFiles+=("${LibFfmpegFile[*]}" ${LibSrtpFile})
+    ModuleLibFiles+=(${LibSrtpFile})
+fi
+if [[ $SRS_FFMPEG_FIT == YES ]]; then
+    ModuleLibFiles+=("${LibFfmpegFile[*]}")
 fi
 if [[ $SRS_SRT == YES ]]; then
     ModuleLibFiles+=("${LibSRTfile[*]}")
@@ -334,7 +363,10 @@ fi
 MODULE_OBJS="${CORE_OBJS[@]} ${KERNEL_OBJS[@]} ${PROTOCOL_OBJS[@]} ${APP_OBJS[@]} ${SERVER_OBJS[@]}"
 ModuleLibIncs=(${SRS_OBJS_DIR} ${LibSTRoot} ${LibGperfRoot} ${LibSSLRoot})
 if [[ $SRS_RTC == YES ]]; then
-    ModuleLibIncs+=("${LibFfmpegRoot[*]}" ${LibSrtpRoot})
+    ModuleLibIncs+=(${LibSrtpRoot})
+fi
+if [[ $SRS_FFMPEG_FIT == YES ]]; then
+    ModuleLibIncs+=("${LibFfmpegRoot[*]}")
 fi
 if [[ $SRS_SRT == YES ]]; then
     MODULE_OBJS="${MODULE_OBJS} ${SRT_OBJS[@]}"
@@ -348,7 +380,10 @@ BUILD_KEY="srs" APP_MAIN="srs_main_server" APP_NAME="srs" . auto/apps.sh
 MODULE_OBJS="${CORE_OBJS[@]} ${KERNEL_OBJS[@]} ${PROTOCOL_OBJS[@]} ${MAIN_OBJS[@]}"
 ModuleLibFiles=(${LibSTfile} ${LibSSLfile} ${LibGperfFile})
 if [[ $SRS_RTC == YES ]]; then
-    ModuleLibFiles+=("${LibFfmpegFile[*]}" ${LibSrtpFile})
+    ModuleLibFiles+=(${LibSrtpFile})
+fi
+if [[ $SRS_FFMPEG_FIT == YES ]]; then
+    ModuleLibFiles+=("${LibFfmpegFile[*]}")
 fi
 #
 for SRS_MODULE in ${SRS_MODULES[*]}; do
@@ -370,14 +405,20 @@ if [ $SRS_UTEST = YES ]; then
         "srs_utest_mp4" "srs_utest_service" "srs_utest_app" "srs_utest_rtc")
     ModuleLibIncs=(${SRS_OBJS_DIR} ${LibSTRoot} ${LibSSLRoot})
     if [[ $SRS_RTC == YES ]]; then
-        ModuleLibIncs+=("${LibFfmpegRoot[*]}" ${LibSrtpRoot})
+        ModuleLibIncs+=(${LibSrtpRoot})
+    fi
+    if [[ $SRS_FFMPEG_FIT == YES ]]; then
+        ModuleLibIncs+=("${LibFfmpegRoot[*]}")
     fi
     if [[ $SRS_SRT == YES ]]; then
         ModuleLibIncs+=("${LibSRTRoot[*]}")
     fi
     ModuleLibFiles=(${LibSTfile} ${LibSSLfile})
     if [[ $SRS_RTC == YES ]]; then
-        ModuleLibFiles+=("${LibFfmpegFile[*]}" ${LibSrtpFile})
+        ModuleLibFiles+=(${LibSrtpFile})
+    fi
+    if [[ $SRS_FFMPEG_FIT == YES ]]; then
+        ModuleLibFiles+=("${LibFfmpegFile[*]}")
     fi
     if [[ $SRS_SRT == YES ]]; then
         ModuleLibFiles+=("${LibSRTfile[*]}")

trunk/src/app/srs_app_config.cpp

@@ -4820,8 +4820,10 @@ int SrsConfig::get_rtc_server_reuseport()
     int v = get_rtc_server_reuseport2();
 
 #if !defined(SO_REUSEPORT)
+    if (v > 1) {
         srs_warn("REUSEPORT not supported, reset %d to %d", reuseport, DEFAULT);
         v = 1
+    }
 #endif
 
     return v;
@@ -4829,7 +4831,7 @@ int SrsConfig::get_rtc_server_reuseport()
 
 int SrsConfig::get_rtc_server_reuseport2()
 {
-    static int DEFAULT = 4;
+    static int DEFAULT = 1;
 
     SrsConfDirective* conf = root->get("rtc_server");
     if (!conf) {

trunk/src/app/srs_app_rtc_source.cpp

@@ -247,7 +247,11 @@ SrsRtcSource::SrsRtcSource()
     rtc_publisher_ = NULL;
 
     req = NULL;
+#ifdef SRS_FFMPEG_FIT
     bridger_ = new SrsRtcFromRtmpBridger(this);
+#else
+    bridger_ = new SrsRtcDummyBridger();
+#endif
 }
 
 SrsRtcSource::~SrsRtcSource()
@@ -266,9 +270,12 @@ srs_error_t SrsRtcSource::initialize(SrsRequest* r)
 
     req = r->copy();
 
-    if ((err = bridger_->initialize(req)) != srs_success) {
+#ifdef SRS_FFMPEG_FIT
+    SrsRtcFromRtmpBridger* bridger = dynamic_cast<SrsRtcFromRtmpBridger*>(bridger_);
+    if ((err = bridger->initialize(req)) != srs_success) {
         return srs_error_wrap(err, "bridge initialize");
     }
+#endif
 
     return err;
 }
@@ -414,6 +421,7 @@ srs_error_t SrsRtcSource::on_rtp(SrsRtpPacket2* pkt)
     return err;
 }
 
+#ifdef SRS_FFMPEG_FIT
 SrsRtcFromRtmpBridger::SrsRtcFromRtmpBridger(SrsRtcSource* source)
 {
     req = NULL;
@@ -936,4 +944,32 @@ srs_error_t SrsRtcFromRtmpBridger::consume_packets(vector<SrsRtpPacket2*>& pkts)
 
     return err;
 }
+#endif
+
+SrsRtcDummyBridger::SrsRtcDummyBridger()
+{
+}
+
+SrsRtcDummyBridger::~SrsRtcDummyBridger()
+{
+}
+
+srs_error_t SrsRtcDummyBridger::on_publish()
+{
+    return srs_error_new(ERROR_RTC_DUMMY_BRIDGER, "no FFmpeg fit");
+}
+
+srs_error_t SrsRtcDummyBridger::on_audio(SrsSharedPtrMessage* /*audio*/)
+{
+    return srs_error_new(ERROR_RTC_DUMMY_BRIDGER, "no FFmpeg fit");
+}
+
+srs_error_t SrsRtcDummyBridger::on_video(SrsSharedPtrMessage* /*video*/)
+{
+    return srs_error_new(ERROR_RTC_DUMMY_BRIDGER, "no FFmpeg fit");
+}
+
+void SrsRtcDummyBridger::on_unpublish()
+{
+}
 

trunk/src/app/srs_app_rtc_source.hpp

@@ -115,7 +115,7 @@ private:
     SrsRequest* req;
     ISrsRtcPublisher* rtc_publisher_;
     // Transmux RTMP to RTC.
-    SrsRtcFromRtmpBridger* bridger_;
+    ISrsSourceBridger* bridger_;
 private:
     // To delivery stream to clients.
     std::vector<SrsRtcConsumer*> consumers;
@@ -159,6 +159,7 @@ public:
     srs_error_t on_rtp(SrsRtpPacket2* pkt);
 };
 
+#ifdef SRS_FFMPEG_FIT
 class SrsRtcFromRtmpBridger : public ISrsSourceBridger
 {
 private:
@@ -197,6 +198,19 @@ private:
     srs_error_t package_fu_a(SrsSharedPtrMessage* msg, SrsSample* sample, int fu_payload_size, std::vector<SrsRtpPacket2*>& pkts);
     srs_error_t consume_packets(std::vector<SrsRtpPacket2*>& pkts);
 };
+#endif
+
+class SrsRtcDummyBridger : public ISrsSourceBridger
+{
+public:
+    SrsRtcDummyBridger();
+    virtual ~SrsRtcDummyBridger();
+public:
+    virtual srs_error_t on_publish();
+    virtual srs_error_t on_audio(SrsSharedPtrMessage* audio);
+    virtual srs_error_t on_video(SrsSharedPtrMessage* video);
+    virtual void on_unpublish();
+};
 
 #endif
 

trunk/src/core/srs_core_version4.hpp

@@ -24,6 +24,6 @@
 #ifndef SRS_CORE_VERSION4_HPP
 #define SRS_CORE_VERSION4_HPP
 
-#define SRS_VERSION4_REVISION 30
+#define SRS_VERSION4_REVISION 32
 
 #endif

trunk/src/kernel/srs_kernel_error.hpp

@@ -352,6 +352,7 @@
 #define ERROR_RTC_DISABLED                  5021
 #define ERROR_RTC_NO_SESSION                5022
 #define ERROR_RTC_INVALID_PARAMS            5023
+#define ERROR_RTC_DUMMY_BRIDGER             5024
 
 ///////////////////////////////////////////////////////
 // GB28181 API error.

trunk/src/main/srs_main_server.cpp

@@ -212,6 +212,9 @@ srs_error_t do_main(int argc, char** argv)
 }
 
 int main(int argc, char** argv) {
+    // For background context id.
+    _srs_context->generate_id();
+
     srs_error_t err = do_main(argc, argv);
 
     if (err != srs_success) {

trunk/src/protocol/srs_service_st.cpp

@@ -70,10 +70,19 @@ srs_error_t srs_st_init()
         return srs_error_new(ERROR_ST_SET_EPOLL, "st enable st failed, current is %s", st_get_eventsys_name());
     }
 
+    // Before ST init, we might have already inited the background cid.
+    string cid = _srs_context->get_id();
+    if (cid.empty()) {
+        cid = _srs_context->generate_id();
+    }
+    
     int r0 = 0;
     if((r0 = st_init()) != 0){
         return srs_error_new(ERROR_ST_INITIALIZE, "st initialize failed, r0=%d", r0);
     }
+
+    // Switch to the background cid.
+    _srs_context->set_id(cid);
     srs_trace("st_init success, use %s", st_get_eventsys_name());
     
     return srs_success;