新增openRtpServerMultiplex接口，支持创建多路复用RTP服务器端口 (#2954)

#2953

postman/ZLMediaKit.postman_collection.json

@@ -1,9 +1,10 @@
 {
 	"info": {
-		"_postman_id": "39e8a1df-cc8e-4e3f-bf5e-197c86e7bf0f",
+		"_postman_id": "509e5f6b-728c-4d5f-b3e8-521d76b2cc7a",
 		"name": "ZLMediaKit",
 		"description": "媒体服务器",
-		"schema": "https://schema.getpostman.com/json/collection/v2.1.0/collection.json"
+		"schema": "https://schema.getpostman.com/json/collection/v2.1.0/collection.json",
+		"_exporter_id": "29185956"
 	},
 	"item": [
 		{
@@ -918,7 +919,7 @@
 				"method": "GET",
 				"header": [],
 				"url": {
-					"raw": "{{ZLMediaKit_URL}}/index/api/broadcastMessage?secret={{ZLMediaKit_secret}}&schema=rtsp&vhost={{defaultVhost}}&app=live&stream=test&msg=Hello zlmediakit123",
+					"raw": "{{ZLMediaKit_URL}}/index/api/broadcastMessage?secret={{ZLMediaKit_secret}}&schema=rtsp&vhost={{defaultVhost}}&app=live&stream=test&msg=Hello ZLMediakit",
 					"host": [
 						"{{ZLMediaKit_URL}}"
 					],
@@ -1247,7 +1248,7 @@
 						},
 						{
 							"key": "stamp",
-							"value": 1000,
+							"value": "1000",
 							"description": "要设置的录像播放位置"
 						}
 					]
@@ -1478,6 +1479,53 @@
 			},
 			"response": []
 		},
+		{
+			"name": "创建多路复用RTP服务器(openRtpServerMultiplex)",
+			"request": {
+				"method": "GET",
+				"header": [],
+				"url": {
+					"raw": "{{ZLMediaKit_URL}}/index/api/openRtpServer?secret={{ZLMediaKit_secret}}&port=0&tcp_mode=1&stream_id=test",
+					"host": [
+						"{{ZLMediaKit_URL}}"
+					],
+					"path": [
+						"index",
+						"api",
+						"openRtpServer"
+					],
+					"query": [
+						{
+							"key": "secret",
+							"value": "{{ZLMediaKit_secret}}",
+							"description": "api操作密钥(配置文件配置)"
+						},
+						{
+							"key": "port",
+							"value": "0",
+							"description": "绑定的端口，0时为随机端口"
+						},
+						{
+							"key": "tcp_mode",
+							"value": "1",
+							"description": "tcp模式，0时为不启用tcp监听，1时为启用tcp监听"
+						},
+						{
+							"key": "stream_id",
+							"value": "test",
+							"description": "该端口绑定的流id\n"
+						},
+						{
+							"key": "only_audio",
+							"value": "0",
+							"description": "是否为单音频track，用于语音对讲",
+							"disabled": true
+						}
+					]
+				}
+			},
+			"response": []
+		},
 		{
 			"name": "连接RTP服务器(connectRtpServer)",
 			"request": {

server/WebApi.cpp

@@ -404,7 +404,7 @@ Value makeMediaSourceJson(MediaSource &media){
 }
 
 #if defined(ENABLE_RTPPROXY)
-uint16_t openRtpServer(uint16_t local_port, const string &stream_id, int tcp_mode, const string &local_ip, bool re_use_port, uint32_t ssrc, bool only_audio) {
+uint16_t openRtpServer(uint16_t local_port, const string &stream_id, int tcp_mode, const string &local_ip, bool re_use_port, uint32_t ssrc, bool only_audio, bool multiplex) {
     lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
     if (s_rtpServerMap.find(stream_id) != s_rtpServerMap.end()) {
         //为了防止RtpProcess所有权限混乱的问题，不允许重复添加相同的stream_id
@@ -412,7 +412,7 @@ uint16_t openRtpServer(uint16_t local_port, const string &stream_id, int tcp_mod
     }
 
     RtpServer::Ptr server = std::make_shared<RtpServer>();
-    server->start(local_port, stream_id, (RtpServer::TcpMode)tcp_mode, local_ip.c_str(), re_use_port, ssrc, only_audio);
+    server->start(local_port, stream_id, (RtpServer::TcpMode)tcp_mode, local_ip.c_str(), re_use_port, ssrc, only_audio, multiplex);
     server->setOnDetach([stream_id]() {
         //设置rtp超时移除事件
         lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
@@ -1182,6 +1182,24 @@ void installWebApi() {
         //回复json
         val["port"] = port;
     });
+      api_regist("/media/api/openRtpServerMultiplex", [](API_ARGS_MAP) {
+      CHECK_SECRET();
+      CHECK_ARGS("port", "stream_id");
+      auto stream_id = allArgs["stream_id"];
+      auto tcp_mode = allArgs["tcp_mode"].as<int>();
+      if (allArgs["enable_tcp"].as<int>() && !tcp_mode) {
+          // 兼容老版本请求，新版本去除enable_tcp参数并新增tcp_mode参数
+          tcp_mode = 1;
+      }
+
+      auto port = openRtpServer(
+          allArgs["port"], stream_id, tcp_mode, "::", true, 0, allArgs["only_audio"].as<bool>(),true);
+      if (port == 0) {
+          throw InvalidArgsException("该stream_id已存在");
+      }
+      // 回复json
+      val["port"] = port;
+  });
 
     api_regist("/index/api/connectRtpServer", [](API_ARGS_MAP_ASYNC) {
         CHECK_SECRET();

server/WebApi.h

@@ -239,7 +239,7 @@ void installWebApi();
 void unInstallWebApi();
 
 #if defined(ENABLE_RTPPROXY)
-uint16_t openRtpServer(uint16_t local_port, const std::string &stream_id, int tcp_mode, const std::string &local_ip, bool re_use_port, uint32_t ssrc, bool only_audio);
+uint16_t openRtpServer(uint16_t local_port, const std::string &stream_id, int tcp_mode, const std::string &local_ip, bool re_use_port, uint32_t ssrc, bool only_audio, bool multiplex=false);
 void connectRtpServer(const std::string &stream_id, const std::string &dst_url, uint16_t dst_port, const std::function<void(const toolkit::SockException &ex)> &cb);
 bool closeRtpServer(const std::string &stream_id);
 #endif

src/Rtp/RtpServer.cpp

@@ -156,7 +156,7 @@ private:
     EventPoller::DelayTask::Ptr _delay_task;
 };
 
-void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_mode, const char *local_ip, bool re_use_port, uint32_t ssrc, bool only_audio) {
+void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_mode, const char *local_ip, bool re_use_port, uint32_t ssrc, bool only_audio, bool multiplex) {
     //创建udp服务器
     Socket::Ptr rtp_socket = Socket::createSocket(nullptr, true);
     Socket::Ptr rtcp_socket = Socket::createSocket(nullptr, true);
@@ -195,7 +195,8 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_
     //创建udp服务器
     UdpServer::Ptr udp_server;
     RtcpHelper::Ptr helper;
-    if (!stream_id.empty()) {
+    //增加了多路复用判断，如果多路复用为true，就走else逻辑，同时保留了原来stream_id为空走else逻辑
+    if (!stream_id.empty() && !multiplex) {
         //指定了流id，那么一个端口一个流(不管是否包含多个ssrc的多个流，绑定rtp源后，会筛选掉ip端口不匹配的流)
         helper = std::make_shared<RtcpHelper>(std::move(rtcp_socket), stream_id);
         helper->startRtcp();

src/Rtp/RtpServer.h

@@ -42,9 +42,10 @@ public:
      * @param local_ip 绑定的本地网卡ip
      * @param re_use_port 是否设置socket为re_use属性
      * @param ssrc 指定的ssrc
+     * @param multiplex 多路复用
      */
     void start(uint16_t local_port, const std::string &stream_id = "", TcpMode tcp_mode = PASSIVE,
-               const char *local_ip = "::", bool re_use_port = true, uint32_t ssrc = 0, bool only_audio = false);
+               const char *local_ip = "::", bool re_use_port = true, uint32_t ssrc = 0, bool only_audio = false, bool multiplex = false);
 
     /**
      * 连接到tcp服务(tcp主动模式)