Change GB28181 to feature/gb28181. 4.0.127

winlinvip · Jun 16, 2021 · 68c48e2 · 68c48e2
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Showing 35 changed files with 14 additions and 9,868 deletions.
diff --git a/.circleci/config.yml b/.circleci/config.yml
@@ -24,14 +24,6 @@ jobs:
       - run: |
           echo "Build SRS without NASM or SRTP-NASM" &&
           cd trunk && ./configure --nasm=off --srtp-nasm=off && make
-  build-c7-gb28181:
-    docker:
-      - image: ossrs/srs:dev
-    steps:
-      - checkout
-      - run: |
-          echo "Build SRS with GB28181" &&
-          cd trunk && ./configure --gb28181=on && make
   build-c7-srt:
     docker:
       - image: ossrs/srs:dev
@@ -111,7 +103,7 @@ jobs:
       - checkout
       - run: |
           echo "Build and run utest for SRS" &&
-          cd trunk && ./configure --gb28181=on --utest=on --gcov=on && make &&
+          cd trunk && ./configure --utest=on --gcov=on && make &&
           ./objs/srs_utest && bash auto/codecov.sh
   run-regression-test:
     docker:
@@ -140,7 +132,6 @@ workflows:
       - run-regression-test
       - build-c7-nortc
       - build-c7-noasm
-      - build-c7-gb28181
       - build-c7-srt
       - build-c8-baseline
       - build-c8-srt

diff --git a/CHANGELOG.md b/CHANGELOG.md
@@ -6,16 +6,14 @@ The changelog for SRS.
 
 ## SRS 4.0 Changelog
 
+* v4.0, 2021-06-16, Change [GB28181](https://github.com/ossrs/srs/issues/1500) to [feature/gb28181](https://github.com/ossrs/srs/tree/feature/gb28181). 4.0.127
 * v4.0, 2021-06-01, Support --shared-ffmpeg to link with *.so for LGPL license. 4.0.126
 * v4.0, 2021-06-01, Support --shared-srt to link with *.so for MPL license. 4.0.125
 * v4.0, 2021-05-31, Use [SPDX-License-Identifier: MIT](https://spdx.dev/ids/). 4.0.124
-* v4.0, 2021-05-28, Fix bugs for GB28181 and RTC. 4.0.123
 * v4.0, 2021-05-21, Fix [#2370][bug #2370] bug for Firefox play stream(published by Chrome). 4.0.121
 * v4.0, 2021-05-21, RTC: Refine sdk, migrate from onaddstream to ontrack. 4.0.120
 * v4.0, 2021-05-21, Tools: Refine configure options. 4.0.119
 * v4.0, 2021-05-20, Fix build fail when disable RTC by --rtc=off. 4.0.118
-* v4.0, 2021-05-19, Fix [#2362][bug #2362]: Allow WebRTC to play before publishing, for GB28181 as such. 4.0.117
-* v4.0, 2021-05-18, Fix [#2355][bug #2355]: GB28181: Fix play by RTC bug. 4.0.116
 * v4.0, 2021-05-15, SRT: Build SRT from source by SRS. 4.0.115
 * v4.0, 2021-05-15, Rename SrsConsumer* to SrsLiveConsumer*. 4.0.114
 * v4.0, 2021-05-15, Rename SrsRtcStream* to SrsRtcSource*. 4.0.113
@@ -63,7 +61,6 @@ The changelog for SRS.
 * v4.0, 2021-01-08, HTML5 video tag resolution adaptive. 4.0.59
 * v4.0, 2021-01-08, Fix memory leak and bugs for RTC. 4.0.58
 * v4.0, 2021-01-06, Merge #2109, Refine srs_string_split.
-* v4.0, 2021-01-06, Merge #2109, Fix bugs for GB28181.
 * v4.0, 2020-12-24, Support disable CherryPy. 4.0.57
 * v4.0, 2020-11-12, For [#1998][bug #1998], Support Firefox, use PT in offer. 4.0.55
 * v4.0, 2020-11-11, For [#1508][bug #1508], Transform http header name to upper camel case. 4.0.54
@@ -81,10 +78,8 @@ The changelog for SRS.
 * v4.0, 2020-07-25, RTC: Support multiple address for client. 4.0.36
 * v4.0, 2020-07-11, Refine log context with random string. 4.0.35
 * v4.0, 2020-07-04, Fix some bugs for RTC. 4.0.34
-* v4.0, 2020-07-03, Merge [#1830][bug #1830] to fix bugs in GB28181. 4.0.33
 * 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
 * v4.0, 2020-05-21, For [#307][bug #307], disable GSO and sendmmsg. 4.0.27
@@ -93,10 +88,8 @@ The changelog for SRS.
 * v4.0, 2020-04-30, For [#307][bug #307], support publish RTC with passing opus. 4.0.24
 * v4.0, 2020-04-14, For [#307][bug #307], support sendmmsg, GSO and reuseport. 4.0.23
 * v4.0, 2020-04-05, For [#307][bug #307], SRTP ASM only works with openssl-1.0, auto detect it. 4.0.22
-* v4.0, 2020-04-04, Merge RTC and GB28181, with bugs fixed. 4.0.21
 * v4.0, 2020-04-04, For [#307][bug #307], refine RTC latency from 600ms to 200ms. 4.0.20
 * v4.0, 2020-04-03, For [#307][bug #307], build SRTP with openssl to improve performance. 4.0.19
-* v4.0, 2020-03-31, For [#1500][bug #1500], support push stream by GB28181. 4.0.18
 * v4.0, 2020-03-31, Play stream by WebRTC on iOS/Android/PC browser. 4.0.17
 * v4.0, 2020-03-28, Support multiple OS/Platform build cache. 4.0.16
 * v4.0, 2020-03-28, For [#1250][bug #1250], support macOS, OSX, MacbookPro, Apple Darwin.

diff --git a/README.md b/README.md
@@ -5,9 +5,9 @@
 [![](https://codecov.io/gh/ossrs/srs/branch/4.0release/graph/badge.svg)](https://codecov.io/gh/ossrs/srs/branch/4.0release)
 [![](https://cloud.githubusercontent.com/assets/2777660/22814959/c51cbe72-ef92-11e6-81cc-32b657b285d5.png)](https://github.com/ossrs/srs/wiki/v4_CN_Contact#wechat)
 
-SRS/4.0，[Leo][release4]，是一个简单高效的实时视频服务器，支持RTMP/WebRTC/HLS/HTTP-FLV/SRT/GB28181。
+SRS/4.0，[Leo][release4]，是一个简单高效的实时视频服务器，支持RTMP/WebRTC/HLS/HTTP-FLV/SRT。
 
-SRS is a simple, high efficiency and realtime video server, supports RTMP/WebRTC/HLS/HTTP-FLV/SRT/GB28181.
+SRS is a simple, high efficiency and realtime video server, supports RTMP/WebRTC/HLS/HTTP-FLV/SRT.
 
 SRS is licenced under [MIT][LICENSE], but some depended libraries are distributed using their [own licenses][LicenseMixing].
 
@@ -69,7 +69,6 @@ Fast index for Wikis:
 
 Other important wiki:
 
-* Usage: How to publish GB28181 to SRS? [#1500](https://github.com/ossrs/srs/issues/1500#issuecomment-606695679)
 * Usage: How to delivery DASH(Experimental)?([CN][v4_CN_SampleDASH], [EN][v4_EN_SampleDASH])
 * Usage: How to transode RTMP stream by FFMPEG?([CN][v4_CN_SampleFFMPEG], [EN][v4_EN_SampleFFMPEG])
 * Usage: How to delivery HTTP FLV Live Streaming Cluster?([CN][v4_CN_SampleHttpFlvCluster], [EN][v4_EN_SampleHttpFlvCluster])
@@ -121,9 +120,6 @@ For optional stream caster services, to push streams to SRS:
 * udp://8935, Stream Caster: [Push MPEGTS over UDP](https://github.com/ossrs/srs/wiki/v4_CN_Streamer#push-mpeg-ts-over-udp) server.
 * tcp://554, Stream Caster: [Push RTSP](https://github.com/ossrs/srs/wiki/v4_CN_Streamer#push-rtsp-to-srs) server.
 * tcp://8936, Stream Caster: [Push HTTP-FLV](https://github.com/ossrs/srs/wiki/v4_CN_Streamer#push-http-flv-to-srs) server.
-* tcp://5060, Stream Caster: [Push GB28181 SIP](https://github.com/ossrs/srs/issues/1500#issuecomment-606695679) server.
-* udp://9000, Stream Caster: [Push GB28181 Media(bundle)](https://github.com/ossrs/srs/issues/1500#issuecomment-606695679) server.
-* udp://58200-58300, Stream Caster: [Push GB28181 Media(no-bundle)](https://github.com/ossrs/srs/issues/1500#issuecomment-606695679) server.
 * udp://10080, Stream Caster: [Push SRT Media](https://github.com/ossrs/srs/issues/1147#issuecomment-577469119) server.
 
 For external services to work with SRS:
@@ -176,7 +172,7 @@ For external services to work with SRS:
 - [x] [Experimental] Support transcode RTMP/AAC to WebRTC/Opus, [#307][bug #307].
 - [x] [Experimental] Support AV1 codec for WebRTC, [#2324][bug #2324].
 - [x] [Experimental] Enhance HTTP Stream Server for HTTP-FLV, HTTPS, HLS etc. [#1657][bug #1657].
-- [x] [Experimental] Support push stream by GB28181, [#1500][bug #1500].
+- [ ] Support push stream by GB28181, [#1500][bug #1500].
 - [x] [Experimental] Support DVR in MP4 format, read [#738][bug #738].
 - [x] [Experimental] Support MPEG-DASH, the future live streaming protocol, read [#299][bug #299].
 - [x] [Experimental] Support pushing MPEG-TS over UDP, please read [bug #250][bug #250].
@@ -306,7 +302,7 @@ The stream architecture of SRS.
 |  MediaSource(2)      |                            |                |
 |  (MPEGTSoverUDP      |                            |                |
 |   HTTP-FLV,   --push-+->- StreamCaster(4) -(rtmp)-+-> SRS          |
-|   GB28181,SRT,       |                            |                |
+|   SRT,               |                            |                |
 |   ......)            |                            |                |
 +----------------------+                            |                |
 |  FFMPEG --push(srt)--+->- SRTModule(5)  ---(rtmp)-+-> SRS          |

diff --git a/trunk/auto/auto_headers.sh b/trunk/auto/auto_headers.sh
@@ -85,12 +85,6 @@ else
     srs_undefine_macro "SRS_SIMULATOR" $SRS_AUTO_HEADERS_H
 fi
 
-if [ $SRS_GB28181 = YES ]; then
-    srs_define_macro "SRS_GB28181" $SRS_AUTO_HEADERS_H
-else
-    srs_undefine_macro "SRS_GB28181" $SRS_AUTO_HEADERS_H
-fi
-
 if [ $SRS_HTTPS = YES ]; then
     srs_define_macro "SRS_HTTPS" $SRS_AUTO_HEADERS_H
 else

diff --git a/trunk/auto/depends.sh b/trunk/auto/depends.sh
@@ -299,15 +299,6 @@ function OSX_prepare()
         echo "Please install pkg-config"; exit -1;
     fi
 
-    if [[ $SRS_GB28181 == YES ]]; then
-        if [[ ! -f /usr/local/opt/libiconv/lib/libiconv.a ]]; then
-            echo "install libiconv"
-            echo "brew install libiconv"
-            brew install libiconv; ret=$?; if [[ 0 -ne $ret ]]; then return $ret; fi
-            echo "install libiconv success"
-        fi
-    fi
-
     if [[ $SRS_SRT == YES ]]; then
         echo "SRT enable, install depend tools"
         tclsh <<< "exit" >/dev/null 2>&1; ret=$?; if [[ 0 -ne $ret ]]; then

diff --git a/trunk/auto/options.sh b/trunk/auto/options.sh
@@ -6,7 +6,6 @@ help=no
 SRS_HDS=NO
 SRS_SRT=NO
 SRS_RTC=RESERVED
-SRS_GB28181=NO
 SRS_CXX11=NO
 SRS_CXX14=NO
 SRS_NGINX=NO
@@ -115,7 +114,6 @@ Features:
   --utest=on|off            Whether build the utest. Default: $(value2switch $SRS_UTEST)
   --srt=on|off              Whether build the SRT. Default: $(value2switch $SRS_SRT)
   --rtc=on|off              Whether build the WebRTC. Default: $(value2switch $SRS_RTC)
-  --gb28181=on|off          Whether build the GB28181. Default: $(value2switch $SRS_GB28181)
   --cxx11=on|off            Whether enable the C++11. Default: $(value2switch $SRS_CXX11)
   --cxx14=on|off            Whether enable the C++14. Default: $(value2switch $SRS_CXX14)
   --ffmpeg-fit=on|off       Whether enable the FFmpeg fit(source code). Default: $(value2switch $SRS_FFMPEG_FIT)
@@ -265,10 +263,6 @@ function parse_user_option() {
         --simulator)                    SRS_SIMULATOR=$(switch2value $value) ;;
         --ffmpeg-fit)                   SRS_FFMPEG_FIT=$(switch2value $value) ;;
 
-        --with-gb28181)                 SRS_GB28181=YES             ;;
-        --without-gb28181)              SRS_GB28181=NO              ;;
-        --gb28181)                      SRS_GB28181=$(switch2value $value) ;;
-
         --cxx11)                        SRS_CXX11=$(switch2value $value) ;;
         --cxx14)                        SRS_CXX14=$(switch2value $value) ;;
 
@@ -473,7 +467,6 @@ function regenerate_options() {
     SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --srt=$(value2switch $SRS_SRT)"
     SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --rtc=$(value2switch $SRS_RTC)"
     SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --simulator=$(value2switch $SRS_SIMULATOR)"
-    SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --gb28181=$(value2switch $SRS_GB28181)"
     SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --cxx11=$(value2switch $SRS_CXX11)"
     SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --cxx14=$(value2switch $SRS_CXX14)"
     SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --ffmpeg-fit=$(value2switch $SRS_FFMPEG_FIT)"

diff --git a/trunk/conf/full.conf b/trunk/conf/full.conf
@@ -366,89 +366,6 @@ stream_caster {
     listen          8936;
 }
 
-# GB28181
-stream_caster {
-    # whether stream caster is enabled.
-    # default: off
-    enabled         on;
-    # the caster type of stream, the casters:
-    #       gb28181, Push GB28181 to SRS.
-    caster          gb28181;
-    # the output rtmp url.
-    # for gb28181 caster, the typically output url:
-    #           rtmp://127.0.0.1/live/[stream]
-    #       where the [stream] is the VideoChannelCodecID.
-    output          rtmp://127.0.0.1/live/[stream];
-    # the listen port for stream caster.
-    #       for gb28181 caster, listen at udp port. for example, 9000.
-    # @remark We can bundle all gb28181 to this port, to reuse this port.
-    #       User can choose to bundle port in API port_mode or SIP invite_port_fixed.
-    listen          9000;
-    # Listen as TCP if on; otherwise, listen as UDP.
-    # default: off
-    tcp_enable      off;
-    # If not bundle ports, use specified ports for each stream.
-    rtp_port_min        58200;
-    rtp_port_max        58300;
-    # Whether wait for keyframe then forward to RTMP.
-    # default: on
-    wait_keyframe       on;
-    # Max timeout in seconds for RTP stream, if timeout, RTCP bye and close stream.
-    # default: 30
-    rtp_idle_timeout    30;
-    # Whether has audio.
-    # @remark Flash/RTMP only supports 11025 22050 44100 sample rate, if not the audio may corrupt.
-    # default: off
-    audio_enable        off;
-    # The exposed IP to receive media stream.
-    #       *           Retrieve server IP automatically, from all network interfaces.
-    #       eth0        Retrieve server IP by specified network interface name. # TODO: Implements it.
-    #       $CANDIDATE  Read the IP from ENV variable $EIP, use * if not set, see https://github.com/ossrs/srs/issues/307#issuecomment-599028124
-    #       x.x.x.x     A specified IP address or DNS name, which can be access by client such as Chrome.
-    # You can specific more than one interface name:
-    #       eth0 eth1   Use network interface eth0 and eth1. # TODO: Implements it.
-    # Also by IP or DNS names:
-    #       192.168.1.3 10.1.2.3 rtc.me # TODO: Implements it.
-    # And by multiple ENV variables:
-    #       $CANDIDATE $EIP # TODO: Implements it.
-    # default: *
-    host                *;
-    #The media channel is automatically created according to the received RTP packet,
-    # and the channel ID is generated according to the RTP SSRC
-    # channelid format: 'chid[ssrc]' [ssrc] is rtp's ssrc
-    auto_create_channel   off;
-
-    sip {
-        # Whether enable embeded SIP server.
-        # default: on
-        enabled         on;
-        # The SIP listen port.
-        # default: 5060
-        listen          5060;
-        # The SIP server ID.
-        # default: 34020000002000000001
-        serial          34020000002000000001;
-        # The SIP server domain.
-        # default: 3402000000
-        realm           3402000000;
-        # The SIP ACK response timeout in seconds.
-        # default: 30
-        ack_timeout         30;
-        # The keepalive timeout in seconds.
-        # default: 120
-        keepalive_timeout   120;
-        # Whether play immediately after registered.
-        # default: on
-        auto_play           on;
-        # Whether bundle media stream port.
-        # default: on
-        invite_port_fixed   on;
-        # interval to query equipment list from equipment or subordinate domain, unit(s)
-        # default: 60
-        query_catalog_interval 60;
-    }
-}
-
 #############################################################################################
 # SRT server section
 #############################################################################################