From 463105c012055b4167f56ccd239595bf309ab0f9 Mon Sep 17 00:00:00 2001 From: winlin Date: Tue, 25 May 2021 14:39:47 +0800 Subject: [PATCH 1/8] Refine README.md --- README.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/README.md b/README.md index 7a9dffeb7..6afbf63b1 100755 --- a/README.md +++ b/README.md @@ -9,7 +9,7 @@ SRS/4.0，[Leo][release4]，是一个简单高效的实时视频服务器，支 SRS is a simple, high efficiency and realtime video server, supports RTMP/WebRTC/HLS/HTTP-FLV/SRT/GB28181. -> Note: SRS is licenced under [MIT][LICENSE], but some depended libraries are distributed using their [own licenses][LicenseMixing]. +SRS is licenced under [MIT][LICENSE], but some depended libraries are distributed using their [own licenses][LicenseMixing]. ## Usage From bd1ec99d5b5462b55370dc4df47e5580ccaa1c03 Mon Sep 17 00:00:00 2001 From: winlin Date: Tue, 25 May 2021 16:04:44 +0800 Subject: [PATCH 2/8] Demo: Update srs.sdk.js --- trunk/3rdparty/signaling/www/demos/index.html | 11 +++- .../signaling/www/demos/js/srs.sdk.js | 51 ++++++++++++++----- .../3rdparty/signaling/www/demos/one2one.html | 14 ++--- trunk/3rdparty/signaling/www/demos/room.html | 14 ++--- 4 files changed, 56 insertions(+), 34 deletions(-) diff --git a/trunk/3rdparty/signaling/www/demos/index.html b/trunk/3rdparty/signaling/www/demos/index.html index c2b617daa..20f2f9a00 100644 --- a/trunk/3rdparty/signaling/www/demos/index.html +++ b/trunk/3rdparty/signaling/www/demos/index.html @@ -19,7 +19,16 @@ let elems = document.getElementsByClassName('srs_demo'); for (var i = 0; i < elems.length; i++) { let elem = elems.item(i); - elem.setAttribute('href', elem.getAttribute('href') + '&room=' + roomName); + + // Use random room. + let href = elem.getAttribute('href') + '&room=' + roomName; + + // For run demos on SRS http server. + if (window.location.port === '8080') { + href += '&wsp=1989'; + } + + elem.setAttribute('href', href); } diff --git a/trunk/3rdparty/signaling/www/demos/js/srs.sdk.js b/trunk/3rdparty/signaling/www/demos/js/srs.sdk.js index 4b030542d..2e3cd87d1 100644 --- a/trunk/3rdparty/signaling/www/demos/js/srs.sdk.js +++ b/trunk/3rdparty/signaling/www/demos/js/srs.sdk.js @@ -29,6 +29,14 @@ function SrsRtcPublisherAsync() { var self = {}; + // https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia + self.constraints = { + audio: true, + video: { + width: {ideal: 320, max: 576} + } + }; + // @see https://github.com/rtcdn/rtcdn-draft // @url The WebRTC url to play with, for example: // webrtc://r.ossrs.net/live/livestream @@ -56,12 +64,14 @@ function SrsRtcPublisherAsync() { self.pc.addTransceiver("audio", {direction: "sendonly"}); self.pc.addTransceiver("video", {direction: "sendonly"}); - var stream = await navigator.mediaDevices.getUserMedia( - {audio: true, video: {width: {max: 320}}} - ); + var stream = await navigator.mediaDevices.getUserMedia(self.constraints); + // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack stream.getTracks().forEach(function (track) { self.pc.addTrack(track); + + // Notify about local track when stream is ok. + self.ontrack && self.ontrack({track: track}); }); var offer = await self.pc.createOffer(); @@ -94,9 +104,6 @@ function SrsRtcPublisherAsync() { ); session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/'; - // Notify about local stream when success. - self.onaddstream && self.onaddstream({stream: stream}); - return session; }; @@ -107,7 +114,10 @@ function SrsRtcPublisherAsync() { }; // The callback when got local stream. - self.onaddstream = function (event) { + // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack + self.ontrack = function (event) { + // Add track to stream of SDK. + self.stream.addTrack(event.track); }; // Internal APIs. @@ -253,6 +263,11 @@ function SrsRtcPublisherAsync() { self.pc = new RTCPeerConnection(null); + // To keep api consistent between player and publisher. + // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack + // @see https://webrtc.org/getting-started/media-devices + self.stream = new MediaStream(); + return self; } @@ -324,8 +339,12 @@ function SrsRtcPlayerAsync() { self.pc = null; }; - // The callback when got remote stream. - self.onaddstream = function (event) {}; + // The callback when got remote track. + // Note that the onaddstream is deprecated, @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onaddstream + self.ontrack = function (event) { + // https://webrtc.org/getting-started/remote-streams + self.stream.addTrack(event.track); + }; // Internal APIs. self.__internal = { @@ -469,9 +488,14 @@ function SrsRtcPlayerAsync() { }; self.pc = new RTCPeerConnection(null); - self.pc.onaddstream = function (event) { - if (self.onaddstream) { - self.onaddstream(event); + + // Create a stream to add track to the stream, @see https://webrtc.org/getting-started/remote-streams + self.stream = new MediaStream(); + + // https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack + self.pc.ontrack = function(event) { + if (self.ontrack) { + self.ontrack(event); } }; @@ -483,7 +507,8 @@ function SrsRtcPlayerAsync() { function SrsRtcFormatSenders(senders, kind) { var codecs = []; senders.forEach(function (sender) { - sender.getParameters().codecs.forEach(function(c) { + var params = sender.getParameters(); + params && params.codecs && params.codecs.forEach(function(c) { if (kind && sender.track.kind !== kind) { return; } diff --git a/trunk/3rdparty/signaling/www/demos/one2one.html b/trunk/3rdparty/signaling/www/demos/one2one.html index 6271e0511..5bef9a4ba 100644 --- a/trunk/3rdparty/signaling/www/demos/one2one.html +++ b/trunk/3rdparty/signaling/www/demos/one2one.html @@ -22,7 +22,7 @@ SRS
    -
  • 一对一通话
    • +
  • 一对一通话
  • 多人通话
  • @@ -226,10 +226,7 @@ publisher.close(); } publisher = new SrsRtcPublisherAsync(); - publisher.onaddstream = function (event) { - console.log('Start publish, event: ', event); - $('#rtc_media_publisher').prop('srcObject', event.stream); - }; + $('#rtc_media_publisher').prop('srcObject', publisher.stream); return publisher.publish(url).then(function(session){ $('#self').text('Self: ' + url); @@ -254,10 +251,7 @@ } player = new SrsRtcPlayerAsync(); - player.onaddstream = function (event) { - console.log('Start play, event: ', event); - $('#rtc_media_player').prop('srcObject', event.stream); - }; + $('#rtc_media_player').prop('srcObject', player.stream); player.play(url).then(function(session){ $('#peer').text('Peer: ' + display); @@ -284,7 +278,7 @@ $('#ff_preview').attr('href', 'http://ossrs.net/players/srs_player.html?app=' + $('#txt_room').val() + '&stream=merge.flv&server=' + conf.host + '&vhost=' + conf.host + '&autostart=true'); // Update href for all navs. - $('ul.srs_nav').children('li').not('.srs_ignore').children('a').not("[href='#']").each(function (i, e) { + $('ul.srs_nav').children('li').not('.srs_ignore').children('a').each(function (i, e) { $(e).attr('href', $(e).attr('href') + conf.rawQuery); }); diff --git a/trunk/3rdparty/signaling/www/demos/room.html b/trunk/3rdparty/signaling/www/demos/room.html index 6f70c98e0..5d75f7fbf 100644 --- a/trunk/3rdparty/signaling/www/demos/room.html +++ b/trunk/3rdparty/signaling/www/demos/room.html @@ -23,7 +23,7 @@
    • 一对一通话
      • -
    • 多人通话
      • +
    • 多人通话
    • GitHub Repo stars @@ -139,10 +139,7 @@ publisher.close(); } publisher = new SrsRtcPublisherAsync(); - publisher.onaddstream = function (event) { - console.log('Start publish, event: ', event); - $('#rtc_media_publisher').prop('srcObject', event.stream); - }; + $('#rtc_media_publisher').prop('srcObject', publisher.stream); return publisher.publish(url).then(function(session){ $('#self').text('Self: ' + url); @@ -178,10 +175,7 @@ video.show(); ui.show(); - player.onaddstream = function (event) { - console.log('Start play, event: ', event); - video.prop('srcObject', event.stream); - }; + video.prop('srcObject', player.stream); player.play(url).then(function(session){ ui.children('#peer').text('Peer: ' + url); @@ -200,7 +194,7 @@ $('#txt_display').val(conf.display); // Update href for all navs. - $('ul.srs_nav').children('li').not('.srs_ignore').children('a').not("[href='#']").each(function (i, e) { + $('ul.srs_nav').children('li').not('.srs_ignore').children('a').each(function (i, e) { $(e).attr('href', $(e).attr('href') + conf.rawQuery); }); From c353f1fe572b6d2e0d93859f8c6eff2d2c8abb49 Mon Sep 17 00:00:00 2001 From: winlin Date: Wed, 26 May 2021 14:21:23 +0800 Subject: [PATCH 3/8] Update Usage --- README.md | 2 ++ trunk/3rdparty/signaling/www/demos/js/srs.sdk.js | 1 + trunk/research/players/js/srs.sdk.js | 2 ++ 3 files changed, 5 insertions(+) diff --git a/README.md b/README.md index 6afbf63b1..1c0f9e26d 100755 --- a/README.md +++ b/README.md @@ -12,6 +12,7 @@ SRS is a simple, high efficiency and realtime video server, supports RTMP/WebRTC SRS is licenced under [MIT][LICENSE], but some depended libraries are distributed using their [own licenses][LicenseMixing]. + ## Usage Run SRS by [docker][docker-srs4], images is [here](https://hub.docker.com/r/ossrs/srs/tags) or [there](https://cr.console.aliyun.com/repository/cn-hangzhou/ossrs/srs/images), @@ -23,6 +24,7 @@ docker run --rm -it -p 1935:1935 -p 1985:1985 -p 8080:8080 \ ossrs/srs:v4.0.117 ./objs/srs -c conf/srs.conf ``` + Or build SRS from source(or [mirrors](#mirrors)), by CentOS7(or Linux([CN][v4_CN_Build],[EN][v4_EN_Build])): ``` diff --git a/trunk/3rdparty/signaling/www/demos/js/srs.sdk.js b/trunk/3rdparty/signaling/www/demos/js/srs.sdk.js index 2e3cd87d1..6d4c6a8b4 100644 --- a/trunk/3rdparty/signaling/www/demos/js/srs.sdk.js +++ b/trunk/3rdparty/signaling/www/demos/js/srs.sdk.js @@ -330,6 +330,7 @@ function SrsRtcPlayerAsync() { await self.pc.setRemoteDescription( new RTCSessionDescription({type: 'answer', sdp: session.sdp}) ); + session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/'; return session; }; diff --git a/trunk/research/players/js/srs.sdk.js b/trunk/research/players/js/srs.sdk.js index 2e3cd87d1..8983eb919 100644 --- a/trunk/research/players/js/srs.sdk.js +++ b/trunk/research/players/js/srs.sdk.js @@ -330,6 +330,8 @@ function SrsRtcPlayerAsync() { await self.pc.setRemoteDescription( new RTCSessionDescription({type: 'answer', sdp: session.sdp}) ); + session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/'; + return session; }; From 295ef90815408dbb837d220ec3672f030c64f305 Mon Sep 17 00:00:00 2001 From: winlin Date: Thu, 27 May 2021 11:13:31 +0800 Subject: [PATCH 4/8] SRS: Update README.md --- README.md | 23 ++++++++++++++++------- 1 file changed, 16 insertions(+), 7 deletions(-) diff --git a/README.md b/README.md index 1c0f9e26d..1a83f9a55 100755 --- a/README.md +++ b/README.md @@ -79,15 +79,20 @@ Other important wiki: ## Ports -The ports used by SRS: +The ports used by SRS, kernel services: + +* tcp://1935, for RTMP live streaming server([CN][v4_CN_DeliveryRTMP],[EN][v4_EN_DeliveryRTMP]). +* tcp://1985, HTTP API server, for HTTP-API([CN][v4_CN_HTTPApi], [EN][v4_EN_HTTPApi]), WebRTC([CN][v4_CN_WebRTC], [EN][v4_EN_WebRTC]), etc. +* tcp://8080, HTTP live streaming server, HTTP-FLV([CN][v4_CN_SampleHttpFlv], [EN][v4_EN_SampleHttpFlv]), HLS([CN][v4_CN_SampleHLS], [EN][v4_EN_SampleHLS]) as such. +* udp://8000, WebRTC Media([CN][v4_CN_WebRTC], [EN][v4_EN_WebRTC]) server. + +For optional HTTPS services, which might be provided by other web servers: -* tcp://1935, for RTMP live streaming server. -* tcp://1985, HTTP API server. -* tcp://1990, HTTPS API server. -* tcp://8080, HTTP live streaming server. * tcp://8088, HTTPS live streaming server. -* udp://8000, [WebRTC Media](https://github.com/ossrs/srs/wiki/v4_CN_WebRTC) server. -* udp://1980, [WebRTC Signaling](https://github.com/ossrs/signaling#usage) server. +* tcp://1990, HTTPS API server. + +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. @@ -95,6 +100,10 @@ The ports used by SRS: * 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: + +* udp://1989, [WebRTC Signaling](https://github.com/ossrs/signaling#usage) server. ## Features From f270d428494d749f68c644f2b5dd3a66b1c344b8 Mon Sep 17 00:00:00 2001 From: winlin Date: Thu, 27 May 2021 13:44:46 +0800 Subject: [PATCH 5/8] Update README.md --- README.md | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/README.md b/README.md index 1a83f9a55..bc58fb037 100755 --- a/README.md +++ b/README.md @@ -1302,14 +1302,13 @@ Remark: | ......) | | | +----------------------+ | | | MediaSource(2) | | | -| (RTSP,FILE, | | | -| HTTP,HLS, --push-+->- StreamCaster(4) -(rtmp)-+-> SRS | -| Device, | | | +| (MPEGTSoverUDP | | | +| HTTP-FLV, --push-+->- StreamCaster(4) -(rtmp)-+-> SRS | +| GB28181,SRT, | | | | ......) | | | +----------------------+ | | | FFMPEG --push(srt)--+->- SRTModule(5) ---(rtmp)-+-> SRS | +----------------------+----------------------------+----------------+ - ``` Remark: From 3e7536c49377b22894de4dcb6067803ec1a283fc Mon Sep 17 00:00:00 2001 From: cfw11 <34058899+cfw11@users.noreply.github.com> Date: Fri, 28 May 2021 21:19:05 +0800 Subject: [PATCH 6/8] GB28181: fix parse rtp-tcp failed (#2378) * fix parse rtp-tcp failed * fix parse rtp-tcp failed Co-authored-by: cfw --- .gitignore | 2 + trunk/conf/push.gb28181.conf | 7 +- trunk/src/app/srs_app_gb28181.cpp | 470 ++++-------------------------- trunk/src/app/srs_app_gb28181.hpp | 34 +-- 4 files changed, 71 insertions(+), 442 deletions(-) diff --git a/.gitignore b/.gitignore index 50715e164..f7e6541ff 100644 --- a/.gitignore +++ b/.gitignore @@ -34,3 +34,5 @@ .idea .DS_Store +/cmake-build-debug/ +/CMakeLists.txt diff --git a/trunk/conf/push.gb28181.conf b/trunk/conf/push.gb28181.conf index c64a6dde0..0890dd94f 100644 --- a/trunk/conf/push.gb28181.conf +++ b/trunk/conf/push.gb28181.conf @@ -33,8 +33,8 @@ stream_caster { # 接收设备端rtp流的多路复用端口 listen 9000; # 多路复用端口类型，on为tcp，off为udp - # 默认：off - tcp_enable off; + # 默认：on + tcp_enable on; # rtp接收监听端口范围，最小值 rtp_port_min 58200; @@ -64,7 +64,8 @@ stream_caster { # 是否开启rtp缓冲 # 开启之后能有效解决rtp乱序等问题 - jitterbuffer_enable on; + # tcp模式建议关闭 + jitterbuffer_enable off; # 服务器主机号，可以域名或ip地址 # 也就是设备端将媒体发送的地址，如果是服务器是内外网 diff --git a/trunk/src/app/srs_app_gb28181.cpp b/trunk/src/app/srs_app_gb28181.cpp index 3fd610428..0fe9ff637 100644 --- a/trunk/src/app/srs_app_gb28181.cpp +++ b/trunk/src/app/srs_app_gb28181.cpp @@ -202,6 +202,11 @@ srs_error_t SrsGb28181PsRtpProcessor::on_udp_packet(const sockaddr* from, const return on_rtp_packet(from, fromlen, buf, nb_buf); } } +srs_error_t SrsGb28181PsRtpProcessor::on_tcp_packet(const sockaddr* from, const int fromlen, char* buf, int nb_buf) +{ + on_udp_packet(from, fromlen, buf, nb_buf); +} + srs_error_t SrsGb28181PsRtpProcessor::on_rtp_packet(const sockaddr* from, const int fromlen, char* buf, int nb_buf) { @@ -460,334 +465,6 @@ srs_error_t SrsGb28181PsRtpProcessor::on_rtp_packet_jitter(const sockaddr* from, return err; } -//SrsGb28181TcpPsRtpProcessor -SrsGb28181TcpPsRtpProcessor::SrsGb28181TcpPsRtpProcessor(SrsGb28181Config* c, std::string id) -{ - config = c; - pprint = SrsPithyPrint::create_caster(); - channel_id = id; -} - -SrsGb28181TcpPsRtpProcessor::~SrsGb28181TcpPsRtpProcessor() -{ - dispose(); - srs_freep(pprint); -} - -void SrsGb28181TcpPsRtpProcessor::dispose() -{ - map::iterator it2; - for (it2 = cache_ps_rtp_packet.begin(); it2 != cache_ps_rtp_packet.end(); ++it2) { - srs_freep(it2->second); - } - cache_ps_rtp_packet.clear(); - - clear_pre_packet(); - - return; -} - -void SrsGb28181TcpPsRtpProcessor::clear_pre_packet() -{ - map::iterator it; - for (it = pre_packet.begin(); it != pre_packet.end(); ++it) { - srs_freep(it->second); - } - pre_packet.clear(); -} - -srs_error_t SrsGb28181TcpPsRtpProcessor::on_rtp(char* buf, int nb_buf, std::string ip, int port) -{ - srs_error_t err = srs_success; - - if (config->jitterbuffer_enable) { - err = on_rtp_packet_jitter(buf, nb_buf, ip, port); - if (err != srs_success) { - srs_warn("SrsGb28181TcpPsRtpProcessor::on_rtp on_rtp_packet_jitter err"); - } - } - else { - return on_rtp_packet(buf, nb_buf, ip, port); - } - return err; -} - -srs_error_t SrsGb28181TcpPsRtpProcessor::on_rtp_packet(char* buf, int nb_buf, std::string ip, int port) -{ - srs_error_t err = srs_success; - bool completed = false; - - pprint->elapse(); - - char address_string[64] = {0}; - char port_string[16] = {0}; - /*if (getnameinfo(from, fromlen, - (char*)&address_string, sizeof(address_string), - (char*)&port_string, sizeof(port_string), - NI_NUMERICHOST | NI_NUMERICSERV)) { - return srs_error_new(ERROR_SYSTEM_IP_INVALID, "bad address"); - }*/ - - //itoa(port, port_string, 10); - int peer_port = port;// atoi(port_string); - - if (true) { - SrsBuffer stream(buf, nb_buf); - SrsPsRtpPacket pkt; - - if ((err = pkt.decode(&stream)) != srs_success) { - return srs_error_wrap(err, "ps rtp decode error"); - } - - //TODO: fixme: the same device uses the same SSRC to send with different local ports - std::stringstream ss; - ss << pkt.ssrc << ":" << pkt.timestamp << ":" << port;// port_string; - std::string pkt_key = ss.str(); - - std::stringstream ss2; - ss2 << pkt.ssrc << ":" << port_string; - std::string pre_pkt_key = ss2.str(); - - if (pre_packet.find(pre_pkt_key) == pre_packet.end()) { - pre_packet[pre_pkt_key] = new SrsPsRtpPacket(); - pre_packet[pre_pkt_key]->copy(&pkt); - } - //cache pkt by ssrc and timestamp - if (cache_ps_rtp_packet.find(pkt_key) == cache_ps_rtp_packet.end()) { - cache_ps_rtp_packet[pkt_key] = new SrsPsRtpPacket(); - } - - //get previous timestamp by ssrc - uint32_t pre_timestamp = pre_packet[pre_pkt_key]->timestamp; - uint32_t pre_sequence_number = pre_packet[pre_pkt_key]->sequence_number; - - //TODO: check sequence number out of order - //it may be out of order, or multiple streaming ssrc are the same - if (((pre_sequence_number + 1) % 65536) != pkt.sequence_number && - pre_sequence_number != pkt.sequence_number) { - srs_warn("gb28181: ps sequence_number out of order, ssrc=%#x, pre=%u, cur=%u, peer(%s, %s)", - pkt.ssrc, pre_sequence_number, pkt.sequence_number, ip.c_str(), port_string); - //return err; - } - - //copy header to cache - cache_ps_rtp_packet[pkt_key]->copy(&pkt); - //accumulate one frame of data, to payload cache - cache_ps_rtp_packet[pkt_key]->payload->append(pkt.payload); - - //detect whether it is a completed frame - if (pkt.marker) {// rtp maker is true, is a completed frame - completed = true; - } - else if (pre_timestamp != pkt.timestamp) { - //current timestamp is different from previous timestamp - //previous timestamp, is a completed frame - std::stringstream ss; - ss << pkt.ssrc << ":" << pre_timestamp << ":" << port_string; - pkt_key = ss.str(); - if (cache_ps_rtp_packet.find(pkt_key) != cache_ps_rtp_packet.end()) { - completed = true; - } - } - - if (pprint->can_print()) { - srs_trace("<- " SRS_CONSTS_LOG_GB28181_CASTER " gb28181: client_id %s, peer(%s, %d) ps rtp packet %dB, age=%d, vt=%d/%u, sts=%u/%u/%#x, paylod=%dB", - channel_id.c_str(), ip.c_str(), peer_port, nb_buf, pprint->age(), pkt.version, - pkt.payload_type, pkt.sequence_number, pkt.timestamp, pkt.ssrc, - pkt.payload->length() - ); - } - - //current packet becomes previous packet - srs_freep(pre_packet[pre_pkt_key]); - pre_packet[pre_pkt_key] = new SrsPsRtpPacket(); - pre_packet[pre_pkt_key]->copy(&pkt);; - - if (!completed) { - return err; - } - //process completed frame data - //clear processed one ps frame - //on completed frame data rtp packet in muxer enqueue - map::iterator key = cache_ps_rtp_packet.find(pkt_key); - if (key != cache_ps_rtp_packet.end()) - { - SrsGb28181RtmpMuxer* muxer = NULL; - //First, search according to the channel_id. Otherwise, search according to the SSRC. - //Some channel_id are created by RTP pool, which are different ports. - //No channel_id are created by multiplexing ports, which are the same port - if (!channel_id.empty()) { - muxer = _srs_gb28181->fetch_rtmpmuxer(channel_id); - } - else { - muxer = _srs_gb28181->fetch_rtmpmuxer_by_ssrc(pkt.ssrc); - } - - //auto crate channel - if (!muxer && config->auto_create_channel) { - //auto create channel generated id - std::stringstream ss, ss1; - ss << "chid" << pkt.ssrc; - std::string tmp_id = ss.str(); - - SrsGb28181StreamChannel channel; - channel.set_channel_id(tmp_id); - channel.set_port_mode(RTP_PORT_MODE_FIXED); - channel.set_ssrc(pkt.ssrc); - - srs_error_t err2 = srs_success; - if ((err2 = _srs_gb28181->create_stream_channel(&channel)) != srs_success) { - srs_warn("gb28181: RtpProcessor create stream channel error %s", srs_error_desc(err2).c_str()); - srs_error_reset(err2); - }; - - muxer = _srs_gb28181->fetch_rtmpmuxer(tmp_id); - } - - if (muxer) { - //TODO: fixme: the same device uses the same SSRC to send with different local ports - //record the first peer port - muxer->set_channel_peer_port(peer_port); - muxer->set_channel_peer_ip(address_string); - //not the first peer port's non processing - if (muxer->channel_peer_port() != peer_port) { - srs_warn("<- " SRS_CONSTS_LOG_GB28181_CASTER " gb28181: client_id %s, ssrc=%#x, first peer_port=%d cur peer_port=%d", - muxer->get_channel_id().c_str(), pkt.ssrc, muxer->channel_peer_port(), peer_port); - srs_freep(key->second); - } - else { - //put it in queue, wait for consumer to process, and then free - muxer->ps_packet_enqueue(key->second); - } - } - else { - //no consumer process it, discarded - srs_freep(key->second); - } - cache_ps_rtp_packet.erase(pkt_key); - } - } - return err; -} - -SrsGb28181RtmpMuxer* SrsGb28181TcpPsRtpProcessor::create_rtmpmuxer(std::string channel_id, uint32_t ssrc) -{ - if (true) { - SrsGb28181RtmpMuxer* muxer = NULL; - //First, search according to the channel_id. Otherwise, search according to the SSRC. - //Some channel_id are created by RTP pool, which are different ports. - //No channel_id are created by multiplexing ports, which are the same port - if (!channel_id.empty()) { - muxer = _srs_gb28181->fetch_rtmpmuxer(channel_id); - } - else { - muxer = _srs_gb28181->fetch_rtmpmuxer_by_ssrc(ssrc); - } - - //auto crate channel - if (!muxer && config->auto_create_channel) { - //auto create channel generated id - std::stringstream ss, ss1; - ss << "chid" << ssrc; - std::string tmp_id = ss.str(); - - SrsGb28181StreamChannel channel; - channel.set_channel_id(tmp_id); - channel.set_port_mode(RTP_PORT_MODE_FIXED); - channel.set_ssrc(ssrc); - - srs_error_t err2 = srs_success; - if ((err2 = _srs_gb28181->create_stream_channel(&channel)) != srs_success) { - srs_warn("gb28181: RtpProcessor create stream channel error %s", srs_error_desc(err2).c_str()); - srs_error_reset(err2); - }; - - muxer = _srs_gb28181->fetch_rtmpmuxer(tmp_id); - } - - return muxer; - }//end if FoundFrame -} - -srs_error_t SrsGb28181TcpPsRtpProcessor::rtmpmuxer_enqueue_data(SrsGb28181RtmpMuxer *muxer, uint32_t ssrc, - int peer_port, std::string address_string, SrsPsRtpPacket *pkt) -{ - srs_error_t err = srs_success; - - if (!muxer) - return err; - - if (muxer) { - //TODO: fixme: the same device uses the same SSRC to send with different local ports - //record the first peer port - muxer->set_channel_peer_port(peer_port); - muxer->set_channel_peer_ip(address_string); - //not the first peer port's non processing - if (muxer->channel_peer_port() != peer_port) { - srs_warn("<- " SRS_CONSTS_LOG_GB28181_CASTER " gb28181: client_id %s, ssrc=%#x, first peer_port=%d cur peer_port=%d", - muxer->get_channel_id().c_str(), ssrc, muxer->channel_peer_port(), peer_port); - } - else { - //muxer->ps_packet_enqueue(pkt); - muxer->insert_jitterbuffer(pkt); - }//end if (muxer->channel_peer_port() != peer_port) - }//end if (muxer) - - return err; -} - -srs_error_t SrsGb28181TcpPsRtpProcessor::on_rtp_packet_jitter(char* buf, int nb_buf, std::string ip, int port) -{ - srs_error_t err = srs_success; - - pprint->elapse(); - - char address_string[64] = {0}; - /*char port_string[16] = {0}; - if (getnameinfo(from, fromlen, - (char*)&address_string, sizeof(address_string), - (char*)&port_string, sizeof(port_string), - NI_NUMERICHOST | NI_NUMERICSERV)) { - return srs_error_new(ERROR_SYSTEM_IP_INVALID, "bad address"); - }*/ - - //itoa(port, port_string, 10); - int peer_port = port;// atoi(port_string); - - if (true) { - SrsBuffer stream(buf, nb_buf); - SrsPsRtpPacket *pkt = new SrsPsRtpPacket();; - - if ((err = pkt->decode(&stream)) != srs_success) { - srs_freep(pkt); - return srs_error_wrap(err, "ps rtp decode error"); - } - - std::stringstream ss3; - ss3 << pkt->ssrc << ":" << port;// port_string; - std::string jitter_key = ss3.str(); - - pkt->completed = pkt->marker; - - - if (pprint->can_print()) { - srs_trace("<- " SRS_CONSTS_LOG_GB28181_CASTER " SrsGb28181TcpPsRtpProcessor::on_rtp_packet_jitter gb28181: client_id %s, peer(%s, %d) ps rtp packet %dB, age=%d, vt=%d/%u, sts=%u/%u/%#x, paylod=%dB", - channel_id.c_str(), address_string, peer_port, nb_buf, pprint->age(), pkt->version, - pkt->payload_type, pkt->sequence_number, pkt->timestamp, pkt->ssrc, - pkt->payload->length() - ); - } - - SrsGb28181RtmpMuxer *muxer = create_rtmpmuxer(channel_id, pkt->ssrc); - if (muxer) { - rtmpmuxer_enqueue_data(muxer, pkt->ssrc, peer_port, ip, pkt); - } - - SrsAutoFree(SrsPsRtpPacket, pkt); - } - - return err; -} //ISrsPsStreamHander ps stream raw video/audio hander interface ISrsPsStreamHander::ISrsPsStreamHander() @@ -2830,9 +2507,10 @@ srs_error_t SrsGb28181Manger::query_device_list(std::string id, SrsJsonArray* ar return sip_service->query_device_list(id, arr); } - -#define SRS_RTSP_BUFFER 262144 -SrsGb28181Conn::SrsGb28181Conn(SrsGb28181Caster* c, srs_netfd_t fd, SrsGb28181TcpPsRtpProcessor *rtp_processor) +#define SRS_RTSP_BUFFER 8192 +#define RTP_TCP_HEADER 2 +#define MAX_PACKAGE_SIZE 1024 * 10 +SrsGb28181Conn::SrsGb28181Conn(SrsGb28181Caster* c, srs_netfd_t fd, SrsGb28181PsRtpProcessor *rtp_processor) { caster = c; stfd = fd; @@ -2877,90 +2555,64 @@ srs_error_t SrsGb28181Conn::do_cycle() { srs_error_t err = srs_success; - // retrieve ip of client. - int fd = srs_netfd_fileno(stfd); - std::string ip = srs_get_peer_ip(fd); - int port = srs_get_peer_port(fd); + // retrieve ip of client. + int fd = srs_netfd_fileno(stfd); + std::string ip = srs_get_peer_ip(fd); + int port = srs_get_peer_port(fd); + int addr_len = sizeof(sockaddr_in); + sockaddr_in *peer_sockaddr = (sockaddr_in*)malloc(addr_len); + peer_sockaddr->sin_family = AF_INET; //设置地址家族 + peer_sockaddr->sin_port = htons(port); //设置端口 + peer_sockaddr->sin_addr.s_addr = inet_addr(ip.c_str()); - if (ip.empty() && !_srs_config->empty_ip_ok()) { - srs_warn("empty ip for fd=%d", srs_netfd_fileno(stfd)); - } - srs_trace("rtsp: serve %s:%d", ip.c_str(), port); - - char* leftData = (char*)malloc(SRS_RTSP_BUFFER);; - uint32_t leftDataLength = 0; - int16_t length = 0; - char* pp = (char*)&length; - char* p = &(mbuffer[0]); - ssize_t nb_read = 0; - int16_t length2; - - // consume all rtp data. - while (true) { - if ((err = trd->pull()) != srs_success) { - free(leftData); - return srs_error_wrap(err, "rtsp cycle"); - } - - //memset(buffer, 0, SRS_RTSP_BUFFER); - nb_read = 0; - if ((err = skt->read(mbuffer + leftDataLength, SRS_RTSP_BUFFER - leftDataLength, &nb_read)) != srs_success) { - free(leftData); - return srs_error_wrap(err, "recv data"); - } - nb_read = nb_read + leftDataLength; - - pp = (char*)&length; - p = &(mbuffer[0]); - pp[1] = *p++; - pp[0] = *p++; + if (ip.empty() && !_srs_config->empty_ip_ok()) { + srs_warn("empty ip for fd=%d", srs_netfd_fileno(stfd)); + } + srs_trace("gb28181 new connect by rtp-tcp from: %s:%d", ip.c_str(), port); - if (nb_read < (length + 2)) {//Not enough one packet. - leftDataLength = leftDataLength + nb_read; - continue; - } + uint32_t left_data_len = 0; //缓存剩余数据 + ssize_t nb_read = 0; + uint16_t packet_len = 0; //rtp包长度 - memset(leftData, 0, SRS_RTSP_BUFFER); + // consume all rtp data. + while (true) { + if ((err = trd->pull()) != srs_success) { + return srs_error_wrap(err, "rtsp cycle"); + } + nb_read = 0; + if ((err = skt->read(mbuffer + left_data_len, SRS_RTSP_BUFFER - left_data_len, &nb_read)) != srs_success) { + return srs_error_wrap(err, "recv data"); + } - while (length > 0) { - if ((length + 2) == nb_read) {//Only one packet. - nb_read = nb_read - 2; - processor->on_rtp(mbuffer + 2, nb_read, ip, port); - leftDataLength = 0; - break; - } - else { //multi packets. - pp = (char*)&length2; - p = &(mbuffer[length + 2]); - pp[1] = *p++; - pp[0] = *p++; - - processor->on_rtp(mbuffer + 2, length, ip, port); - - leftDataLength = nb_read - (length + 2); - nb_read = leftDataLength; - memcpy(leftData, mbuffer + length + 2, leftDataLength); - - pp = (char*)&length; - p = &(mbuffer[length + 2]); - pp[1] = *p++; - pp[0] = *p++; - - if (leftDataLength < (length + 2)) {//Not enough one packet. - memcpy(mbuffer, leftData, leftDataLength); - break; - } - else { - memcpy(mbuffer, leftData, leftDataLength); - } - } - } - } + left_data_len = nb_read + left_data_len; + char * buf = mbuffer; - free(leftData); + uint32_t index = 0; + for( ; index < left_data_len; ){ + if (index + RTP_TCP_HEADER >= left_data_len){ //less rtp package + break; + } + packet_len = (((uint8_t *) buf)[index] << 8) | ((uint8_t *) buf)[index + 1]; + if (packet_len > MAX_PACKAGE_SIZE){ + //FIXME 自动重新invite? + srs_error("abnormal RTP packet length:%d, close the tcp conn:%s", packet_len, remote_ip().c_str()); + return err; + } + if (index + RTP_TCP_HEADER + packet_len >= left_data_len){ + break; + } + processor->on_tcp_packet((sockaddr*)peer_sockaddr, addr_len, buf + index + RTP_TCP_HEADER, packet_len); + index = index + RTP_TCP_HEADER + packet_len; + } + if (index != 0) { //update left data + left_data_len = left_data_len - index; + memmove(mbuffer, buf + index, left_data_len); + } - return err; + } + free(peer_sockaddr); + return err; } srs_error_t SrsGb28181Conn::cycle() @@ -2996,7 +2648,7 @@ SrsGb28181Caster::SrsGb28181Caster(SrsConfDirective* c) // TODO: FIXME: support reload. output = _srs_config->get_stream_caster_output(c); config = new SrsGb28181Config(c); - rtp_processor = new SrsGb28181TcpPsRtpProcessor(config, ""); + rtp_processor = new SrsGb28181PsRtpProcessor(config, ""); manager = new SrsResourceManager("GB28181TCP", true); } diff --git a/trunk/src/app/srs_app_gb28181.hpp b/trunk/src/app/srs_app_gb28181.hpp index 07bd0cfa2..04283bc27 100644 --- a/trunk/src/app/srs_app_gb28181.hpp +++ b/trunk/src/app/srs_app_gb28181.hpp @@ -94,7 +94,6 @@ class SrsSipRequest; class SrsGb28181RtmpMuxer; class SrsGb28181Config; class SrsGb28181PsRtpProcessor; -class SrsGb28181TcpPsRtpProcessor; class SrsGb28181SipService; class SrsGb28181StreamChannel; class SrsGb28181SipSession; @@ -176,37 +175,12 @@ private: // Interface ISrsUdpHandler public: virtual srs_error_t on_udp_packet(const sockaddr* from, const int fromlen, char* buf, int nb_buf); + virtual srs_error_t on_tcp_packet(const sockaddr* from, const int fromlen, char* buf, int nb_buf); public: virtual srs_error_t on_rtp_packet_jitter(const sockaddr* from, const int fromlen, char* buf, int nb_buf); virtual srs_error_t on_rtp_packet(const sockaddr* from, const int fromlen, char* buf, int nb_buf); }; -class SrsGb28181TcpPsRtpProcessor -{ -private: - SrsPithyPrint* pprint; - SrsGb28181Config* config; - std::map cache_ps_rtp_packet; - std::map pre_packet; - std::string channel_id; - bool auto_create_channel; -public: - SrsGb28181TcpPsRtpProcessor(SrsGb28181Config* c, std::string sid); - virtual ~SrsGb28181TcpPsRtpProcessor(); -private: - bool can_send_ps_av_packet(); - void dispose(); - void clear_pre_packet(); - SrsGb28181RtmpMuxer* create_rtmpmuxer(std::string channel_id, uint32_t ssrc); - srs_error_t rtmpmuxer_enqueue_data(SrsGb28181RtmpMuxer *muxer, uint32_t ssrc, - int peer_port, std::string address_string, SrsPsRtpPacket *pkt); - // Interface ISrsTcpHandler -public: - virtual srs_error_t on_rtp(char* buf, int nb_buf, std::string ip, int port); -public: - virtual srs_error_t on_rtp_packet_jitter(char* buf, int nb_buf, std::string ip, int port); - virtual srs_error_t on_rtp_packet(char* buf, int nb_buf, std::string ip, int port); -}; //ps stream processing parsing interface class ISrsPsStreamHander @@ -581,9 +555,9 @@ private: SrsRtspStack* rtsp; SrsGb28181Caster* caster; SrsCoroutine* trd; - SrsGb28181TcpPsRtpProcessor *processor; + SrsGb28181PsRtpProcessor *processor; public: - SrsGb28181Conn(SrsGb28181Caster* c, srs_netfd_t fd, SrsGb28181TcpPsRtpProcessor *rtp_processor); + SrsGb28181Conn(SrsGb28181Caster* c, srs_netfd_t fd, SrsGb28181PsRtpProcessor *rtp_processor); virtual ~SrsGb28181Conn(); public: virtual srs_error_t serve(); @@ -603,7 +577,7 @@ class SrsGb28181Caster : public ISrsTcpHandler private: std::string output; SrsGb28181Config *config; - SrsGb28181TcpPsRtpProcessor *rtp_processor; + SrsGb28181PsRtpProcessor *rtp_processor; private: std::vector clients; SrsResourceManager* manager; From 1c7c74b01a5e02adefa6542f84c6168a9662d2e7 Mon Sep 17 00:00:00 2001 From: "louis.xia" <68469352@qq.com> Date: Fri, 28 May 2021 21:38:12 +0800 Subject: [PATCH 7/8] GB28181: Fix parse rtp tcp failed (#2382) * fix parse rtp-tcp failed * fix parse rtp-tcp failed * fix gb28181 support tcp stack is setup:passive * Update push.gb28181.tcp.conf Co-authored-by: cfw Co-authored-by: Winlin --- trunk/conf/push.gb28181.tcp.conf | 143 ++++++++++++++++++++++++ trunk/src/app/srs_app_gb28181_stack.cpp | 2 + 2 files changed, 145 insertions(+) create mode 100644 trunk/conf/push.gb28181.tcp.conf diff --git a/trunk/conf/push.gb28181.tcp.conf b/trunk/conf/push.gb28181.tcp.conf new file mode 100644 index 000000000..257195afd --- /dev/null +++ b/trunk/conf/push.gb28181.tcp.conf @@ -0,0 +1,143 @@ +# push gb28181 stream to SRS. + +listen 1935; +max_connections 1000; +daemon off; +srs_log_tank console; + +http_api { + enabled on; + listen 1985; +} + +http_server { + enabled on; + listen 8080; +} + +stats { + network 0; +} + +stream_caster { + enabled on; + caster gb28181; + + # 转发流到rtmp服务器地址与端口 + # TODO: https://github.com/ossrs/srs/pull/1679/files#r400875104 + # [stream] is VideoChannelCodecID(视频通道编码ID) for sip + # 自动创建的道通[stream] 是‘chid[ssrc]’ [ssrc]是rtp的ssrc + # [ssrc] rtp中的ssrc + output rtmp://127.0.0.1:1935/live/[stream]; + + # 接收设备端rtp流的多路复用端口 + listen 9000; + # 多路复用端口类型，on为tcp，off为udp + # 默认：off + tcp_enable on; + + # rtp接收监听端口范围，最小值 + rtp_port_min 58200; + # rtp接收监听端口范围，最大值 + rtp_port_max 58300; + + # 是否等待关键帧之后，再转发， + # off:不需等待，直接转发 + # on:等第一个关键帧后，再转发 + wait_keyframe on; + + # rtp包空闲等待时间，如果指定时间没有收到任何包 + # rtp监听连接自动停止，发送BYE命令 + rtp_idle_timeout 30; + + # 是否转发音频流 + # 目前只支持aac格式，所以需要设备支持aac格式 + # on:转发音频 + # off:不转发音频，只有视频 + # *注意*!!!:flv 只支持11025 22050 44100 三种 + # 如果设备端没有三种中任何一个，转发时为自动选择一种格式 + # 同时也会将adts的头封装在flv aac raw数据中 + # 这样的话播放器为自动通过adts头自动选择采样频率 + # 像ffplay, vlc都可以，但是flash是没有声音， + # 因为flash,只支持11025 22050 44100 + audio_enable off; + + # 是否开启rtp缓冲 + # 开启之后能有效解决rtp乱序等问题 + # tcp模式建议关闭 + jitterbuffer_enable off; + + # 服务器主机号，可以域名或ip地址 + # 也就是设备端将媒体发送的地址，如果是服务器是内外网 + # 需要写外网地址， + # 调用api创建stream session时返回ip地址也是host + # $CANDIDATE 是系统环境变量，从环境变量获取地址，如果没有配置，用* + # *代表指定stats network 的网卡号地址，如果没有配置network，默认则是第0号网卡地址 + # TODO: https://github.com/ossrs/srs/pull/1679/files#r400917594 + host $CANDIDATE; + + #根据收到ps rtp包自带创建rtmp媒体通道，不需要api接口创建 + #rtmp地址参数[stream] 就是通道id 格式chid[ssrc] + auto_create_channel off; + + sip { + # 是否启用srs内部sip信令 + # 为on信令走srs, off 只转发ps流 + enabled on; + + # sip监听udp端口 + listen 5060; + + # SIP server ID(SIP服务器ID). + # 设备端配置编号需要与该值一致，否则无法注册 + serial 34020000002000000001; + + # SIP server domain(SIP服务器域) + realm 3402000000; + + # 服务端发送ack后，接收回应的超时时间，单位为秒 + # 如果指定时间没有回应，认为失败 + ack_timeout 30; + + # 设备心跳维持时间，如果指定时间内(秒）没有接收一个心跳 + # 认为设备离线 + keepalive_timeout 120; + + # 注册之后是否自动给设备端发送invite + # on: 是 off 不是，需要通过api控制 + auto_play on; + # 设备将流发送的端口，是否固定 + # on 发送流到多路复用端口 如9000 + # off 自动从rtp_mix_port - rtp_max_port 之间的值中 + # 选一个可以用的端口 + invite_port_fixed on; + + # 向设备或下级域查询设备列表的间隔，单位(秒) + # 默认60秒 + query_catalog_interval 60; + } +} + +rtc_server { + enabled on; + # Listen at udp://8000 + listen 8000; + # + # The $CANDIDATE means fetch from env, if not configed, use * as default. + # + # The * means retrieving server IP automatically, from all network interfaces, + # @see https://github.com/ossrs/srs/issues/307#issuecomment-599028124 + candidate $CANDIDATE; +} + +vhost __defaultVhost__ { + rtc { + enabled on; + bframe discard; + } + + http_remux { + enabled on; + mount [vhost]/[app]/[stream].flv; + } +} diff --git a/trunk/src/app/srs_app_gb28181_stack.cpp b/trunk/src/app/srs_app_gb28181_stack.cpp index c5bcfda0c..8d0c81b9d 100644 --- a/trunk/src/app/srs_app_gb28181_stack.cpp +++ b/trunk/src/app/srs_app_gb28181_stack.cpp @@ -1062,6 +1062,8 @@ void SrsSipStack::req_invite(stringstream& ss, SrsSipRequest *req, string ip, in //<< "m=video " << port << " TCP/RTP/AVP 98" << SRS_RTSP_CRLF << "a=recvonly" << SRS_RTSP_CRLF << "a=rtpmap:96 PS/90000" << SRS_RTSP_CRLF + << "a=setup:passive" << SRS_RTSP_CRLF + << "a=connection:new" << SRS_RTSP_CRLF //TODO: current no support //<< "a=rtpmap:97 MPEG4/90000" << SRS_RTSP_CRLF //<< "a=rtpmap:98 H264/90000" << SRS_RTSP_CRLF From a8d40f4cb1ed4973392eedc8f2d8f35563d2e215 Mon Sep 17 00:00:00 2001 From: winlin Date: Fri, 28 May 2021 21:40:43 +0800 Subject: [PATCH 8/8] Update AUTHORS --- AUTHORS.txt | 2 ++ 1 file changed, 2 insertions(+) diff --git a/AUTHORS.txt b/AUTHORS.txt index 099d3d620..81290103e 100644 --- a/AUTHORS.txt +++ b/AUTHORS.txt @@ -84,3 +84,5 @@ CONTRIBUTORS ordered by first contribution. * xbpeng121<53243357+xbpeng121@users.noreply.github.com> * johzzy * stone +* cfw11<34058899+cfw11@users.noreply.github.com> +* louis.xia<68469352@qq.com>