GB28181: Fix bug for parsing GB to RTC.

trunk/3rdparty/srs-bench/Makefile

@@ -5,13 +5,13 @@ default: bench test
 clean:
 	rm -f ./objs/srs_bench ./objs/srs_test
 
-.format.txt: *.go srs/*.go vnet/*.go janus/*.go
+.format.txt: *.go srs/*.go vnet/*.go janus/*.go gb28181/*.go
 	gofmt -w .
 	echo "done" > .format.txt
 
 bench: ./objs/srs_bench
 
-./objs/srs_bench: .format.txt *.go srs/*.go vnet/*.go janus/*.go Makefile
+./objs/srs_bench: .format.txt *.go srs/*.go vnet/*.go janus/*.go gb28181/*.go Makefile
 	go build -mod=vendor -o objs/srs_bench .
 
 test: ./objs/srs_test

trunk/auto/options.sh

@@ -276,6 +276,7 @@ function parse_user_option() {
 
         --with-stream-caster)           SRS_STREAM_CASTER=YES       ;;
         --stream-caster)                SRS_STREAM_CASTER=$(switch2value $value) ;;
+        --stream-converter)             SRS_STREAM_CASTER=$(switch2value $value) ;;
 
         --with-utest)                   SRS_UTEST=YES               ;;
         --without-utest)                SRS_UTEST=NO                ;;
@@ -474,7 +475,7 @@ fi
 function apply_detail_options() {
     # Always enable HTTP utilies.
     if [ $SRS_HTTP_CORE = NO ]; then SRS_HTTP_CORE=YES; echo -e "${YELLOW}[WARN] Always enable HTTP utilies.${BLACK}"; fi
-    if [ $SRS_STREAM_CASTER = NO ]; then SRS_STREAM_CASTER=YES; echo -e "${YELLOW}[WARN] Always enable StreamCaster.${BLACK}"; fi
+    if [ $SRS_STREAM_CASTER = NO ]; then SRS_STREAM_CASTER=YES; echo -e "${YELLOW}[WARN] Always enable StreamConverter.${BLACK}"; fi
     if [ $SRS_INGEST = NO ]; then SRS_INGEST=YES; echo -e "${YELLOW}[WARN] Always enable Ingest.${BLACK}"; fi
     if [ $SRS_SSL = NO ]; then SRS_SSL=YES; echo -e "${YELLOW}[WARN] Always enable SSL.${BLACK}"; fi
     if [ $SRS_STAT = NO ]; then SRS_STAT=YES; echo -e "${YELLOW}[WARN] Always enable Statistic.${BLACK}"; fi
@@ -514,7 +515,7 @@ function regenerate_options() {
     SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --stat=$(value2switch $SRS_STAT)"
     SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --http-callback=$(value2switch $SRS_HTTP_CALLBACK)"
     SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --http-server=$(value2switch $SRS_HTTP_SERVER)"
-    SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --stream-caster=$(value2switch $SRS_STREAM_CASTER)"
+    SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --stream-converter=$(value2switch $SRS_STREAM_CASTER)"
     SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --http-api=$(value2switch $SRS_HTTP_API)"
     SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --utest=$(value2switch $SRS_UTEST)"
     SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --cherrypy=$(value2switch $SRS_CHERRYPY)"
@@ -607,7 +608,7 @@ function check_option_conflicts() {
     # check variable neccessary
     if [ $SRS_HDS = RESERVED ]; then echo "you must specifies the hds, see: ./configure --help"; __check_ok=NO; fi
     if [ $SRS_SSL = RESERVED ]; then echo "you must specifies the ssl, see: ./configure --help"; __check_ok=NO; fi
-    if [ $SRS_STREAM_CASTER = RESERVED ]; then echo "you must specifies the stream-caster, see: ./configure --help"; __check_ok=NO; fi
+    if [ $SRS_STREAM_CASTER = RESERVED ]; then echo "you must specifies the stream-converter, see: ./configure --help"; __check_ok=NO; fi
     if [ $SRS_UTEST = RESERVED ]; then echo "you must specifies the utest, see: ./configure --help"; __check_ok=NO; fi
     if [ $SRS_GPERF = RESERVED ]; then echo "you must specifies the gperf, see: ./configure --help"; __check_ok=NO; fi
     if [ $SRS_GPERF_MC = RESERVED ]; then echo "you must specifies the gperf-mc, see: ./configure --help"; __check_ok=NO; fi

trunk/auto/summary.sh

@@ -11,7 +11,7 @@ SrsIngestSummaryColor="\${YELLOW}(Disabled) "; if [ $SRS_INGEST = YES ]; then Sr
 SrsHttpCallbackSummaryColor="\${YELLOW}(Disabled) "; if [ $SRS_HTTP_CALLBACK = YES ]; then SrsHttpCallbackSummaryColor="\${GREEN}"; fi
 SrsHttpServerSummaryColor="\${YELLOW}(Disabled) "; if [ $SRS_HTTP_SERVER = YES ]; then SrsHttpServerSummaryColor="\${GREEN}"; fi
 SrsHttpApiSummaryColor="\${YELLOW}(Disabled) "; if [ $SRS_HTTP_API = YES ]; then SrsHttpApiSummaryColor="\${GREEN}"; fi
-SrsStreamCasterSummaryColor="\${YELLOW}(Disabled) "; if [ $SRS_STREAM_CASTER = YES ]; then SrsStreamCasterSummaryColor="\${GREEN}"; fi
+SrsStreamConverterSummaryColor="\${YELLOW}(Disabled) "; if [ $SRS_STREAM_CASTER = YES ]; then SrsStreamConverterSummaryColor="\${GREEN}"; fi
 SrsUtestSummaryColor="\${YELLOW}(Disabled) "; if [ $SRS_UTEST = YES ]; then SrsUtestSummaryColor="\${GREEN}"; fi
 SrsGperfSummaryColor="\${GREEN}(Disabled) "; if [ $SRS_GPERF = YES ]; then SrsGperfSummaryColor="\${GREEN}"; fi
 SrsGperfMCSummaryColor="\${GREEN}(Disabled) "; if [ $SRS_GPERF_MC = YES ]; then SrsGperfMCSummaryColor="\${YELLOW}"; fi
@@ -45,7 +45,7 @@ echo -e "     |     ${SrsIngestSummaryColor}About ingester, please read https://
 echo -e "     |     ${SrsHttpCallbackSummaryColor}About http-callback, please read https://ossrs.net/lts/zh-cn/docs/v4/doc/http-callback\${BLACK}"
 echo -e "     |     ${SrsHttpServerSummaryColor}Aoubt http-server, please read https://ossrs.net/lts/zh-cn/docs/v4/doc/http-server\${BLACK}"
 echo -e "     |     ${SrsHttpApiSummaryColor}About http-api, please read https://ossrs.net/lts/zh-cn/docs/v4/doc/http-api\${BLACK}"
-echo -e "     |     ${SrsStreamCasterSummaryColor}About stream-caster, please read https://ossrs.net/lts/zh-cn/docs/v4/doc/streamer\${BLACK}"
+echo -e "     |     ${SrsStreamConverterSummaryColor}About stream-caster, please read https://ossrs.net/lts/zh-cn/docs/v4/doc/streamer\${BLACK}"
 echo -e "     |     ${SrsValgrindSummaryColor}About VALGRIND, please read https://github.com/ossrs/state-threads/issues/2\${BLACK}"
 echo -e "     \${BLACK}+------------------------------------------------------------------------------------\${BLACK}"
 echo -e "\${GREEN}binaries, please read https://ossrs.net/lts/zh-cn/docs/v4/doc/install\${BLACK}"

trunk/conf/full.conf

@@ -540,67 +540,67 @@ vhost rtc.vhost.srs.com {
 }
 
 #############################################################################################
-# Streamer sections
+# Stream converter sections
 #############################################################################################
-# the streamer cast stream from other protocol to SRS over RTMP.
+# The stream converter coverts stream from other protocol to SRS over RTMP.
 # @see https://github.com/ossrs/srs/tree/develop#stream-architecture
 
-# Push MPEGTS over UDP, see https://github.com/ossrs/srs/issues/250
+# Push MPEGTS over UDP to SRS.
 stream_caster {
-    # whether stream caster is enabled.
+    # Whether stream converter is enabled.
-    # default: off
+    # Default: off
-    enabled         on;
+    enabled on;
-    # the caster type of stream, the casters:
+    # The type of stream converter, could be:
-    #       mpegts_over_udp, MPEG-TS over UDP caster.
+    #       mpegts_over_udp, push MPEG-TS over UDP and convert to RTMP.
-    caster          mpegts_over_udp;
+    caster mpegts_over_udp;
-    # the output rtmp url.
+    # The output rtmp url.
-    # for mpegts_over_udp caster, the typically output url:
+    # For mpegts_over_udp converter, the typically output url:
     #           rtmp://127.0.0.1/live/livestream
-    output          rtmp://127.0.0.1/live/livestream;
+    output rtmp://127.0.0.1/live/livestream;
-    # the listen port for stream caster.
+    # The listen port for stream converter.
-    #       for mpegts_over_udp caster, listen at udp port. for example, 8935.
+    # For mpegts_over_udp converter, listen at udp port. for example, 8935.
-    listen          8935;
+    listen 8935;
 }
 
-# Push HTTP-FLV stream, see https://github.com/ossrs/srs/issues/2611
+# Push FLV by HTTP POST to SRS.
 stream_caster {
-    # whether stream caster is enabled.
+    # Whether stream converter is enabled.
-    # default: off
+    # Default: off
-    enabled         on;
+    enabled on;
-    # the caster type of stream, the casters:
+    # The type of stream converter, could be:
-    #       flv, FLV over HTTP by POST.
+    #       flv, push FLV by HTTP POST and convert to RTMP.
-    caster          flv;
+    caster flv;
-    # the output rtmp url.
+    # The output rtmp url.
-    # for flv caster, the typically output url:
+    # For flv converter, the typically output url:
     #           rtmp://127.0.0.1/[app]/[stream]
-    #       for example, POST to url:
+    # For example, POST to url:
     #           http://127.0.0.1:8936/live/livestream.flv
-    #       where the [app] is "live" and [stream] is "livestream", output is:
+    # Where the [app] is "live" and [stream] is "livestream", output is:
     #           rtmp://127.0.0.1/live/livestream
-    output          rtmp://127.0.0.1/[app]/[stream];
+    output rtmp://127.0.0.1/[app]/[stream];
-    # the listen port for stream caster.
+    # The listen port for stream converter.
-    #       for flv caster, listen at tcp port. for example, 8936.
+    # For flv converter, listen at tcp port. for example, 8936.
-    listen          8936;
+    listen 8936;
 }
 
 # For GB28181 server, see https://github.com/ossrs/srs/issues/3176
 # For SIP specification, see https://www.ietf.org/rfc/rfc3261.html
 # For GB28181 2016 spec, see https://openstd.samr.gov.cn/bzgk/gb/newGbInfo?hcno=469659DC56B9B8187671FF08748CEC89
 stream_caster {
-    # Whether GB28181 server is enabled.
+    # Whether stream converter is enabled.
     # Default: off
     enabled off;
-    # The caster type of stream, the casters:
+    # The type of stream converter, could be:
-    #       gb28181, Push GB28181 stream to SRS.
+    #       gb28181, Push GB28181 stream and convert to RTMP.
     caster gb28181;
     # The output rtmp url.
-    # For gb28181 caster, the typically output url:
+    # For gb28181 converter, the typically output url:
     #           rtmp://127.0.0.1/live/[stream]
     # The available variables:
     #           [stream] The video channel codec ID.
     output rtmp://127.0.0.1/live/[stream];
-    # The listen TCP/UDP port for stream caster.
+    # The listen TCP/UDP port for stream converter.
-    #       For gb28181 caster, listen at TCP/UDP port. for example, 9000.
+    #       For gb28181 converter, listen at TCP/UDP port. for example, 9000.
     # @remark We always enable bundle for media streams at this port.
     listen 9000;
     # SIP server for GB28181. Please note that this is only a demonstrated SIP server, please never use it in your

trunk/conf/gb28181.conf

@@ -12,6 +12,8 @@ stream_caster {
     sip {
         enabled on;
         listen 5060;
+        # @see https://ossrs.net/lts/zh-cn/docs/v5/doc/gb28181#config-candidate
+        candidate $CANDIDATE;
     }
 }
 

trunk/conf/push.flv.conf

@@ -6,20 +6,40 @@ listen              1935;
 max_connections     1000;
 daemon              off;
 srs_log_tank        console;
-http_server {
-    enabled         on;
-    listen          8080;
-    dir             ./objs/nginx/html;
-}
 stream_caster {
     enabled         on;
     caster          flv;
     output          rtmp://127.0.0.1/[app]/[stream];
     listen          8936;
 }
+http_server {
+    enabled         on;
+    listen          8080;
+    dir             ./objs/nginx/html;
+}
+http_api {
+    enabled on;
+    listen 1985;
+}
+rtc_server {
+    enabled on;
+    listen 8000; # UDP port
+    # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
+    candidate $CANDIDATE;
+}
 vhost __defaultVhost__ {
+    rtc {
+        enabled on;
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
+        rtmp_to_rtc on;
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
+        rtc_to_rtmp on;
+    }
     http_remux {
-        enabled     on;
+        enabled on;
-        mount       [vhost]/[app]/[stream].flv;
+        mount [vhost]/[app]/[stream].flv;
+    }
+    hls {
+        enabled on;
     }
 }

trunk/conf/push.mpegts.over.udp.conf

@@ -18,9 +18,29 @@ http_server {
     listen          8080;
     dir             ./objs/nginx/html;
 }
+http_api {
+    enabled on;
+    listen 1985;
+}
+rtc_server {
+    enabled on;
+    listen 8000; # UDP port
+    # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
+    candidate $CANDIDATE;
+}
 vhost __defaultVhost__ {
+    rtc {
+        enabled on;
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
+        rtmp_to_rtc on;
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
+        rtc_to_rtmp on;
+    }
     http_remux {
-        enabled     on;
+        enabled on;
-        mount       [vhost]/[app]/[stream].flv;
+        mount [vhost]/[app]/[stream].flv;
+    }
+    hls {
+        enabled on;
     }
 }

trunk/configure

@@ -642,9 +642,9 @@ else
     echo -e "${YELLOW}Warning: HLS is disabled.${BLACK}"
 fi
 if [ $SRS_STREAM_CASTER = YES ]; then
-    echo -e "${YELLOW}Experiment: StreamCaster is enabled.${BLACK}"
+    echo -e "${YELLOW}Experiment: StreamConverter is enabled.${BLACK}"
 else
-    echo -e "${GREEN}Note: StreamCaster is disabled.${BLACK}"
+    echo -e "${GREEN}Note: StreamConverter is disabled.${BLACK}"
 fi
 if [ $SRS_HDS = YES ]; then
     echo -e "${YELLOW}Experiment: HDS is enabled.${BLACK}"

trunk/doc/Architecture.md

@@ -35,8 +35,8 @@ The stream architecture of SRS.
 +----------------------+                            |                |
 |  MediaSource(2)      |                            |                |
 |  (MPEGTSoverUDP      |                            |                |
-|   HTTP-FLV,   --push-+->- StreamCaster(4) -(rtmp)-+-> SRS          |
+|   HTTP-FLV,   --push-+->--   Stream(4) ----(rtmp)-+-> SRS          |
-|   SRT,               |                            |                |
+|   SRT,               |      Converter             |                |
 |   ......)            |                            |                |
 +----------------------+                            |                |
 |  FFMPEG --push(srt)--+->- SRTModule(5)  ---(rtmp)-+-> SRS          |
@@ -48,7 +48,7 @@ Remark:
 1. Encoder: Encoder pushs RTMP stream to SRS.
 1. MediaSource: Supports any media source, ingesting by ffmpeg.
 1. Ingester: Forks a ffmpeg(or other tools) to ingest as rtmp to SRS, please read [Ingest][v4_CN_Ingest].
-1. Streamer: Remuxs other protocols to RTMP, please read [Streamer][v4_CN_Streamer].
+1. Stream Converter: Remuxs other protocols to RTMP, please read [Stream Converter][v4_CN_Streamer].
 1. EXEC: Like NGINX-RTMP, EXEC forks external tools for events, please read [ng-exec][v4_CN_NgExec].
 1. SRTModule: A isolate module which run in [hybrid](https://github.com/ossrs/srs/issues/1147#issuecomment-577574883) model.
 

trunk/src/app/srs_app_gb28181.cpp

@@ -1745,7 +1745,7 @@ srs_error_t SrsGbMuxer::write_h264_ipb_frame(char* frame, int frame_size, uint32
 
     // when sps or pps not sent, ignore the packet.
     if (!h264_sps_pps_sent_) {
-        return srs_error_new(ERROR_H264_DROP_BEFORE_SPS_PPS, "drop sps/pps");
+        return srs_error_new(ERROR_H264_DROP_BEFORE_SPS_PPS, "drop for no sps/pps");
     }
 
     // 5bits, 7.3.1 NAL unit syntax,
@@ -1892,6 +1892,9 @@ srs_error_t SrsGbMuxer::connect()
         return err;
     }
 
+    // Cleanup the data before connect again.
+    close();
+
     string url = srs_string_replace(output_, "[stream]", session_->sip_transport()->resource()->device_id());
     srs_trace("Muxer: Convert GB to RTMP %s", url.c_str());
 
@@ -1915,6 +1918,14 @@ srs_error_t SrsGbMuxer::connect()
 void SrsGbMuxer::close()
 {
     srs_freep(sdk_);
+
+    // Regenerate the AAC sequence header.
+    aac_specific_config_ = "";
+
+    // Wait for the next AVC sequence header.
+    h264_sps_pps_sent_ = false;
+    h264_sps_ = "";
+    h264_pps_ = "";
 }
 
 SrsSipResponseWriter::SrsSipResponseWriter(ISrsProtocolReadWriter* io) : SrsHttpResponseWriter(io)

trunk/src/app/srs_app_rtc_source.cpp

@@ -82,8 +82,14 @@ srs_error_t aac_raw_append_adts_header(SrsSharedPtrMessage* shared_audio, SrsFor
         return err;
     }
 
+    // If no audio RAW frame, or not parsed for no sequence header, drop the packet.
+    if (format->audio->nb_samples == 0) {
+        srs_warn("RTC: Drop AAC %d bytes for no sample", shared_audio->size);
+        return err;
+    }
+
     if (format->audio->nb_samples != 1) {
-        return srs_error_new(ERROR_RTC_RTP_MUXER, "adts");
+        return srs_error_new(ERROR_RTC_RTP_MUXER, "adts samples=%d", format->audio->nb_samples);
     }
 
     int nb_buf = format->audio->samples[0].size + 7;

trunk/src/core/srs_core_version5.hpp

@@ -9,6 +9,6 @@
 
 #define VERSION_MAJOR       5
 #define VERSION_MINOR       0
-#define VERSION_REVISION    74
+#define VERSION_REVISION    75
 
 #endif

trunk/src/kernel/srs_kernel_error.hpp

@@ -268,7 +268,7 @@
     XX(ERROR_HTTP_WITH_BODY                , 3097, "HttpWithBody", "Failed for HTTP body")
 
 /**************************************************/
-/* HTTP/StreamCaster protocol error. */
+/* HTTP/StreamConverter protocol error. */
 #define SRS_ERRNO_MAP_HTTP(XX) \
     XX(ERROR_HTTP_PATTERN_EMPTY            , 4000, "HttpPatternEmpty", "Failed to handle HTTP request for no pattern") \
     XX(ERROR_HTTP_PATTERN_DUPLICATED       , 4001, "HttpPatternDuplicated", "Failed to handle HTTP request for pattern duplicated") \

trunk/src/utest/srs_utest_config.cpp

@@ -2766,7 +2766,7 @@ VOID TEST(ConfigMainTest, CheckGlobalConfig)
     }
 }
 
-VOID TEST(ConfigMainTest, CheckStreamCaster)
+VOID TEST(ConfigMainTest, CheckStreamConverter)
 {
     srs_error_t err;