Merge branch '4.0release' into merge/develop

.gitignore

@@ -34,3 +34,5 @@
 .idea
 .DS_Store
 
+/cmake-build-debug/
+/CMakeLists.txt

AUTHORS.txt

@@ -84,3 +84,5 @@ CONTRIBUTORS ordered by first contribution.
 * xbpeng121<53243357+xbpeng121@users.noreply.github.com>
 * johzzy<hellojinqiang@gmail.com>
 * stone<bluestn@163.com>
+* cfw11<34058899+cfw11@users.noreply.github.com>
+* louis.xia<68469352@qq.com>

README.md

@@ -9,9 +9,10 @@ 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].
 
 <a name="product"></a>
+<a name="usage-docker"></a>
 ## 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
 ```
 
+<a name="usage-source"></a>
 Or build SRS from source(or [mirrors](#mirrors)), by CentOS7(or Linux([CN][v4_CN_Build],[EN][v4_EN_Build])):
 
 ```
@@ -77,15 +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.
+* tcp://1990, HTTPS API server.
-* udp://1980, [WebRTC Signaling](https://github.com/ossrs/signaling#usage) 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.
@@ -94,6 +101,10 @@ The ports used by SRS:
 * 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
 
 - [x] Using coroutine by ST, it's really simple and stupid enough.
@@ -1298,14 +1309,13 @@ Remark:
 |   ......)            |                            |                |
 +----------------------+                            |                |
 |  MediaSource(2)      |                            |                |
-|  (RTSP,FILE,         |                            |                |
+|  (MPEGTSoverUDP      |                            |                |
-|   HTTP,HLS,   --push-+->- StreamCaster(4) -(rtmp)-+-> SRS          |
+|   HTTP-FLV,   --push-+->- StreamCaster(4) -(rtmp)-+-> SRS          |
-|   Device,            |                            |                |
+|   GB28181,SRT,       |                            |                |
 |   ......)            |                            |                |
 +----------------------+                            |                |
 |  FFMPEG --push(srt)--+->- SRTModule(5)  ---(rtmp)-+-> SRS          |
 +----------------------+----------------------------+----------------+
-
 ```
 
 Remark:

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);
     }
 </script>
 </body>

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(
+        var stream = await navigator.mediaDevices.getUserMedia(self.constraints);
-            {audio: true, video: {width: {max: 320}}}
+
-        );
         // @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;
 }
 
@@ -315,6 +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;
     };
 
@@ -324,8 +340,12 @@ function SrsRtcPlayerAsync() {
         self.pc = null;
     };
 
-    // The callback when got remote stream.
+    // The callback when got remote track.
-    self.onaddstream = function (event) {};
+    // 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 +489,14 @@ function SrsRtcPlayerAsync() {
     };
 
     self.pc = new RTCPeerConnection(null);
-    self.pc.onaddstream = function (event) {
+
-        if (self.onaddstream) {
+    // Create a stream to add track to the stream, @see https://webrtc.org/getting-started/remote-streams
-            self.onaddstream(event);
+    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 +508,8 @@ 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;
             }

trunk/3rdparty/signaling/www/demos/one2one.html

@@ -22,7 +22,7 @@
             <a class="brand" href="https://github.com/ossrs/srs">SRS</a>
             <div class="nav-collapse collapse">
                 <ul class="nav srs_nav">
-                    <li class="active"><a href="#">一对一通话</a></li>
+                    <li class="active"><a href="one2one.html">一对一通话</a></li>
                     <li><a href="room.html">多人通话</a></li>
                     <li class="srs_ignore">
                         <a href="https://github.com/ossrs/signaling">
@@ -226,10 +226,7 @@
                 publisher.close();
             }
             publisher = new SrsRtcPublisherAsync();
-            publisher.onaddstream = function (event) {
+            $('#rtc_media_publisher').prop('srcObject', publisher.stream);
-                console.log('Start publish, event: ', event);
-                $('#rtc_media_publisher').prop('srcObject', event.stream);
-            };
 
             return publisher.publish(url).then(function(session){
                 $('#self').text('Self: ' + url);
@@ -254,10 +251,7 @@
             }
 
             player = new SrsRtcPlayerAsync();
-            player.onaddstream = function (event) {
+            $('#rtc_media_player').prop('srcObject', player.stream);
-                console.log('Start play, event: ', event);
-                $('#rtc_media_player').prop('srcObject', event.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);
         });
 

trunk/3rdparty/signaling/www/demos/room.html

@@ -23,7 +23,7 @@
             <div class="nav-collapse collapse">
                 <ul class="nav srs_nav">
                     <li><a href="one2one.html">一对一通话</a></li>
-                    <li class="active"><a href="#">多人通话</a></li>
+                    <li class="active"><a href="room.html">多人通话</a></li>
                     <li class="srs_ignore">
                         <a href="https://github.com/ossrs/signaling">
                             <img alt="GitHub Repo stars" src="https://img.shields.io/github/stars/ossrs/signaling?style=social">
@@ -139,10 +139,7 @@
                 publisher.close();
             }
             publisher = new SrsRtcPublisherAsync();
-            publisher.onaddstream = function (event) {
+            $('#rtc_media_publisher').prop('srcObject', publisher.stream);
-                console.log('Start publish, event: ', event);
-                $('#rtc_media_publisher').prop('srcObject', event.stream);
-            };
 
             return publisher.publish(url).then(function(session){
                 $('#self').text('Self: ' + url);
@@ -178,10 +175,7 @@
             video.show();
             ui.show();
 
-            player.onaddstream = function (event) {
+            video.prop('srcObject', player.stream);
-                console.log('Start play, event: ', event);
-                video.prop('srcObject', event.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);
         });
 

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
+    # 默认：on
-    tcp_enable            off;
+    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

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;
+    }
+}

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;
     };
 

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<std::string, SrsPsRtpPacket*>::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<std::string, SrsPsRtpPacket*>::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<std::string, SrsPsRtpPacket*>::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 8192
-#define SRS_RTSP_BUFFER 262144
+#define RTP_TCP_HEADER 2
-SrsGb28181Conn::SrsGb28181Conn(SrsGb28181Caster* c, srs_netfd_t fd, SrsGb28181TcpPsRtpProcessor *rtp_processor)
+#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.
+    // retrieve ip of client.
-	int fd = srs_netfd_fileno(stfd);
+    int fd = srs_netfd_fileno(stfd);
-	std::string ip = srs_get_peer_ip(fd);
+    std::string ip = srs_get_peer_ip(fd);
-	int port = srs_get_peer_port(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);;
+    if (ip.empty() && !_srs_config->empty_ip_ok()) {
-	uint32_t leftDataLength = 0;
+        srs_warn("empty ip for fd=%d", srs_netfd_fileno(stfd));
-	int16_t length = 0;
+    }
-	char* pp = (char*)&length;
+    srs_trace("gb28181 new connect by rtp-tcp from: %s:%d", ip.c_str(), port);
-	char* p = &(mbuffer[0]);
-	ssize_t nb_read = 0;
-	int16_t length2;
 
-	// consume all rtp data.
+    uint32_t left_data_len = 0; //缓存剩余数据
-	while (true) {
+    ssize_t nb_read = 0;
-		if ((err = trd->pull()) != srs_success) {
+    uint16_t  packet_len = 0; //rtp包长度
-			free(leftData);
-			return srs_error_wrap(err, "rtsp cycle");
-		}
 
-		//memset(buffer, 0, SRS_RTSP_BUFFER);
+    // consume all rtp data.
-		nb_read = 0;
+    while (true) {
-		if ((err = skt->read(mbuffer + leftDataLength, SRS_RTSP_BUFFER - leftDataLength, &nb_read)) != srs_success) {
+        if ((err = trd->pull()) != srs_success) {
-			free(leftData);
+            return srs_error_wrap(err, "rtsp cycle");
-			return srs_error_wrap(err, "recv data");
+        }
-		}
+        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");
+        }
 
-		nb_read = nb_read + leftDataLength;
+        left_data_len = nb_read + left_data_len;
+        char * buf = mbuffer;
 
-		pp = (char*)&length;
+        uint32_t index = 0;
-		p = &(mbuffer[0]);
+        for( ; index < left_data_len; ){
-		pp[1] = *p++;
+            if (index + RTP_TCP_HEADER >= left_data_len){ //less rtp package
-		pp[0] = *p++;
+                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);
+        }
 
-		if (nb_read < (length + 2)) {//Not enough one packet.
+    }
-			leftDataLength = leftDataLength + nb_read;
+    free(peer_sockaddr);
-			continue;
+    return err;
-		}
-
-		memset(leftData, 0, SRS_RTSP_BUFFER);
-
-		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);
-				}
-			}
-		}
-	}
-
-	free(leftData);
-
-	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);
 }
 

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<std::string, SrsPsRtpPacket*> cache_ps_rtp_packet;
-	std::map<std::string, SrsPsRtpPacket*> 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<SrsGb28181Conn*> clients;
 	SrsResourceManager* manager;

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