c sdk 新增webrtc相关函数 (#4473)

另外调整函数位置，whip、whep请求设置Content-Type为application/sdp

api/include/mk_common.h

@@ -259,31 +259,24 @@ API_EXPORT uint16_t API_CALL mk_rtp_server_start(uint16_t port);
  */
 API_EXPORT uint16_t API_CALL mk_rtc_server_start(uint16_t port);
 
-// 获取webrtc answer sdp回调函数  [AUTO-TRANSLATED:10c93fa9]
-// Get webrtc answer sdp callback function
-typedef void(API_CALL *on_mk_webrtc_get_answer_sdp)(void *user_data, const char *answer, const char *err);
 
 /**
- * webrtc交换sdp，根据offer sdp生成answer sdp
- * @param user_data 回调用户指针
- * @param cb 回调函数
- * @param type webrtc插件类型，支持echo,play,push
- * @param offer webrtc offer sdp
- * @param url rtc url, 例如 rtc://__defaultVhost/app/stream?key1=val1&key2=val2
- * webrtc exchange sdp, generate answer sdp based on offer sdp
- * @param user_data Callback user pointer
- * @param cb Callback function
- * @param type webrtc plugin type, supports echo, play, push
- * @param offer webrtc offer sdp
- * @param url rtc url, for example rtc://__defaultVhost/app/stream?key1=val1&key2=val2
- 
- * [AUTO-TRANSLATED:ea79659b]
+ * 创建websocket[s]信令服务器
+ * @param port websocket监听端口
+ * @param ssl 是否为ssl类型服务器
+ * @return 0:失败,非0:端口号
+ * 
  */
-API_EXPORT void API_CALL mk_webrtc_get_answer_sdp(void *user_data, on_mk_webrtc_get_answer_sdp cb, const char *type,
-                                                  const char *offer, const char *url);
+API_EXPORT uint16_t API_CALL mk_signaling_server_start(uint16_t port, int ssl);
+
+/**
+ * 创建webrtc-ice[s]服务器
+ * @param port websocket监听端口
+ * @return 0:失败,非0:端口号
+ *
+ */
+API_EXPORT uint16_t API_CALL mk_ice_server_start(uint16_t port);
 
-API_EXPORT void API_CALL mk_webrtc_get_answer_sdp2(void *user_data, on_user_data_free user_data_free, on_mk_webrtc_get_answer_sdp cb, const char *type,
-                                                  const char *offer, const char *url);
 
 /**
  * 创建srt服务器

api/include/mk_mediakit.h

@@ -27,5 +27,6 @@
 #include "mk_frame.h"
 #include "mk_track.h"
 #include "mk_transcode.h"
+#include "mk_webrtc.h"
 
 #endif /* MK_API_H_ */

api/include/mk_webrtc.h

@@ -0,0 +1,111 @@
+/*
+ * Copyright (c) 2016-present The ZLMediaKit project authors. All Rights Reserved.
+ *
+ * This file is part of ZLMediaKit(https://github.com/ZLMediaKit/ZLMediaKit).
+ *
+ * Use of this source code is governed by MIT-like license that can be found in the
+ * LICENSE file in the root of the source tree. All contributing project authors
+ * may be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MK_WEBRTC_H
+#define MK_WEBRTC_H
+#include "mk_common.h"
+#include "mk_proxyplayer.h"
+#include <stdint.h>
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+
+// 获取webrtc answer sdp回调函数  [AUTO-TRANSLATED:10c93fa9]
+// Get webrtc answer sdp callback function
+typedef void(API_CALL *on_mk_webrtc_get_answer_sdp)(void *user_data, const char *answer, const char *err);
+
+// 获取webrtc proxy player信息回调函数
+typedef void(API_CALL *on_mk_webrtc_get_proxy_player_info_cb)(const char *info_json, const char *err);
+
+//WebRTC-注册到信令服务器、WebRTC-从信令服务器注销回调函数
+typedef void(API_CALL *on_mk_webrtc_room_keeper_info_cb)(void *user_data, const char *room_key, const char *err);
+
+//获取WebRTC-Peer查看注册信息、WebRTC-信令服务器查看注册信息回调函数
+typedef void(API_CALL *on_mk_webrtc_room_keeper_data_cb)(const char *data);
+
+
+/**
+ * webrtc交换sdp，根据offer sdp生成answer sdp
+ * @param user_data 回调用户指针
+ * @param cb 回调函数
+ * @param type webrtc插件类型，支持echo,play,push
+ * @param offer webrtc offer sdp
+ * @param url rtc url, 例如 rtc://__defaultVhost/app/stream?key1=val1&key2=val2
+ * webrtc exchange sdp, generate answer sdp based on offer sdp
+ * @param user_data Callback user pointer
+ * @param cb Callback function
+ * @param type webrtc plugin type, supports echo, play, push
+ * @param offer webrtc offer sdp
+ * @param url rtc url, for example rtc://__defaultVhost/app/stream?key1=val1&key2=val2
+
+ * [AUTO-TRANSLATED:ea79659b]
+ */
+API_EXPORT void API_CALL mk_webrtc_get_answer_sdp(void *user_data, on_mk_webrtc_get_answer_sdp cb, const char *type, const char *offer, const char *url);
+
+API_EXPORT void API_CALL mk_webrtc_get_answer_sdp2(
+    void *user_data, on_user_data_free user_data_free, on_mk_webrtc_get_answer_sdp cb, const char *type, const char *offer, const char *url);
+
+/**
+ * 获取webrtc proxy player信息
+ * @param mk_proxy_player 代理
+ * @param cb 回调函数
+ */
+API_EXPORT void API_CALL mk_webrtc_get_proxy_player_info(mk_proxy_player ctx, on_mk_webrtc_get_proxy_player_info_cb cb);
+
+
+/**
+ * WebRTC-注册到信令服务器
+ * @param server_host 信令服务器host
+ * @param server_port 信令服务器port
+ * @param room_id 房间id
+ * @param ssl 是否启用ssl
+ * @param cb 回调函数
+ * @param user_data 用户数据
+ */
+API_EXPORT void API_CALL
+mk_webrtc_add_room_keeper(const char *room_id, const char *server_host, uint16_t server_port, int ssl, on_mk_webrtc_room_keeper_info_cb cb, void *user_data);
+
+
+API_EXPORT void API_CALL mk_webrtc_add_room_keeper2(
+    const char *room_id, const char *server_host, uint16_t server_port, int ssl, on_mk_webrtc_room_keeper_info_cb cb, void *user_data,
+    on_user_data_free user_data_free);
+
+
+/**
+ * WebRTC-从信令服务器注销
+ * @param room_key 房间key
+ * @param cb 回调函数
+ * @param user_data 用户数据
+ */
+API_EXPORT void API_CALL mk_webrtc_del_room_keeper(const char *room_key, on_mk_webrtc_room_keeper_info_cb cb, void *user_data);
+
+API_EXPORT void API_CALL
+mk_webrtc_del_room_keeper2(const char *room_key, on_mk_webrtc_room_keeper_info_cb cb, void *user_data, on_user_data_free user_data_free);
+
+
+/**
+ * WebRTC-Peer查看注册信息
+ * @param cb 回调函数
+ */
+API_EXPORT void API_CALL mk_webrtc_list_room_keeper(on_mk_webrtc_room_keeper_data_cb cb);
+
+/**
+ * WebRTC-信令服务器查看注册信息
+ * @param cb 回调函数
+ */
+API_EXPORT void API_CALL mk_webrtc_list_rooms(on_mk_webrtc_room_keeper_data_cb cb);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* MK_WEBRTC_H */

api/source/mk_common.cpp

@@ -29,6 +29,7 @@ using namespace mediakit;
 static TcpServer::Ptr rtsp_server[2];
 static TcpServer::Ptr rtmp_server[2];
 static TcpServer::Ptr http_server[2];
+static TcpServer::Ptr signaling_server[2];
 static TcpServer::Ptr shell_server;
 
 #ifdef ENABLE_RTPPROXY
@@ -37,9 +38,14 @@ static RtpServer::Ptr rtpServer;
 #endif
 
 #ifdef ENABLE_WEBRTC
-#include "../webrtc/WebRtcSession.h"
+#include "webrtc/WebRtcSession.h"
+#include "webrtc/IceSession.hpp"
+#include "webrtc/WebRtcSignalingSession.h"
+#include "webrtc/WebRtcTransport.h"
 static UdpServer::Ptr rtcServer_udp;
 static TcpServer::Ptr rtcServer_tcp;
+static UdpServer::Ptr iceServer_udp;
+static TcpServer::Ptr iceServer_tcp;
 #endif
 
 #if defined(ENABLE_SRT)
@@ -288,46 +294,45 @@ API_EXPORT uint16_t API_CALL mk_rtc_server_start(uint16_t port) {
 #endif
 }
 
-#ifdef ENABLE_WEBRTC
-class WebRtcArgsUrl : public mediakit::WebRtcArgs {
-public:
-    WebRtcArgsUrl(std::string url) { _url = std::move(url); }
 
-    toolkit::variant operator[](const std::string &key) const override {
-        if (key == "url") {
-            return _url;
+API_EXPORT uint16_t API_CALL mk_signaling_server_start(uint16_t port, int ssl) {
+#ifdef ENABLE_WEBRTC
+    ssl = MAX(0, MIN(ssl, 1));
+    try {
+        signaling_server[ssl] = std::make_shared<TcpServer>();
+        if (ssl) {
+            signaling_server[ssl]->start<WebRtcWebcosktSignalSslSession>(port);
+        } else {
+            signaling_server[ssl]->start<WebRtcWebcosktSignalingSession>(port);
         }
-        return "";
+        return signaling_server[ssl]->getPort();
+    } catch (std::exception &ex) {
+        signaling_server[ssl] = nullptr;
+        WarnL << ex.what();
+        return 0;
     }
-
-private:
-    std::string _url;
-};
-#endif
-
-API_EXPORT void API_CALL mk_webrtc_get_answer_sdp(void *user_data,  on_mk_webrtc_get_answer_sdp cb, const char *type,
-                                                   const char *offer, const char *url) {
-    mk_webrtc_get_answer_sdp2(user_data, nullptr, cb, type, offer, url);
-}
-API_EXPORT void API_CALL mk_webrtc_get_answer_sdp2(void *user_data, on_user_data_free user_data_free, on_mk_webrtc_get_answer_sdp cb, const char *type,
-                                                  const char *offer, const char *url) {
-#ifdef ENABLE_WEBRTC
-    assert(type && offer && url && cb);
-    auto session = std::make_shared<HttpSession>(Socket::createSocket());
-    std::string offer_str = offer;
-    std::shared_ptr<void> ptr(user_data, user_data_free ? user_data_free : [](void *) {});
-    auto args = std::make_shared<WebRtcArgsUrl>(url);
-    WebRtcPluginManager::Instance().negotiateSdp(*session, type, *args, [offer_str, session, ptr, cb](const WebRtcInterface &exchanger) mutable {
-        auto &handler = const_cast<WebRtcInterface &>(exchanger);
-        try {
-            auto sdp_answer = handler.getAnswerSdp(offer_str);
-            cb(ptr.get(), sdp_answer.data(), nullptr);
-        } catch (std::exception &ex) {
-            cb(ptr.get(), nullptr, ex.what());
-        }
-    });
 #else
     WarnL << "未启用webrtc功能, 编译时请开启ENABLE_WEBRTC";
+    return 0;
+#endif
+}
+
+API_EXPORT uint16_t API_CALL mk_ice_server_start(uint16_t port){
+#ifdef ENABLE_WEBRTC
+    try {
+        iceServer_tcp = std::make_shared<TcpServer>();
+        iceServer_udp = std::make_shared<UdpServer>();
+        iceServer_udp->start<IceSession>(port);
+        iceServer_tcp->start<IceSession>(port);
+    } catch (std::exception &ex) {
+        iceServer_udp = nullptr;
+        iceServer_tcp = nullptr;
+        WarnL << ex.what();
+        return 0;
+    }
+#else
+    WarnL << "未启用webrtc功能, 编译时请开启ENABLE_WEBRTC";
+    return 0;
 #endif
 }
 

api/source/mk_webrtc.cpp

@@ -0,0 +1,187 @@
+/*
+ * Copyright (c) 2016-present The ZLMediaKit project authors. All Rights Reserved.
+ *
+ * This file is part of ZLMediaKit(https://github.com/ZLMediaKit/ZLMediaKit).
+ *
+ * Use of this source code is governed by MIT-like license that can be found in the
+ * LICENSE file in the root of the source tree. All contributing project authors
+ * may be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "mk_webrtc.h"
+#include "mk_util.h"
+
+#include <stdarg.h>
+#include <unordered_map>
+#include "Util/logger.h"
+#include "Util/SSLBox.h"
+#include "Util/File.h"
+#include "Network/TcpServer.h"
+#include "Network/UdpServer.h"
+#include "Thread/WorkThreadPool.h"
+
+#include "Rtsp/RtspSession.h"
+#include "Rtmp/RtmpSession.h"
+#include "Http/HttpSession.h"
+#include "Shell/ShellSession.h"
+#include "Player/PlayerProxy.h"
+#include "webrtc/WebRtcProxyPlayer.h"
+#include "webrtc/WebRtcProxyPlayerImp.h"
+#include "../webrtc/WebRtcSignalingPeer.h"
+#include "../webrtc/WebRtcSignalingSession.h"
+using namespace std;
+using namespace toolkit;
+using namespace mediakit;
+
+#ifdef ENABLE_WEBRTC
+#include "../webrtc/WebRtcSession.h"
+static UdpServer::Ptr rtcServer_udp;
+static TcpServer::Ptr rtcServer_tcp;
+class WebRtcArgsUrl : public mediakit::WebRtcArgs {
+public:
+    WebRtcArgsUrl(std::string url) { _url = std::move(url); }
+
+    toolkit::variant operator[](const std::string &key) const override {
+        if (key == "url") {
+            return _url;
+        }
+        return "";
+    }
+
+private:
+    std::string _url;
+};
+#endif
+
+API_EXPORT void API_CALL mk_webrtc_get_answer_sdp(void *user_data, on_mk_webrtc_get_answer_sdp cb, const char *type, const char *offer, const char *url) {
+    mk_webrtc_get_answer_sdp2(user_data, nullptr, cb, type, offer, url);
+}
+API_EXPORT void API_CALL mk_webrtc_get_answer_sdp2(
+    void *user_data, on_user_data_free user_data_free, on_mk_webrtc_get_answer_sdp cb, const char *type, const char *offer, const char *url) {
+#ifdef ENABLE_WEBRTC
+    assert(type && offer && url && cb);
+    auto session = std::make_shared<HttpSession>(Socket::createSocket());
+    std::string offer_str = offer;
+    std::shared_ptr<void> ptr(user_data, user_data_free ? user_data_free : [](void *) {});
+    auto args = std::make_shared<WebRtcArgsUrl>(url);
+    WebRtcPluginManager::Instance().negotiateSdp(*session, type, *args, [offer_str, session, ptr, cb](const WebRtcInterface &exchanger) mutable {
+        auto &handler = const_cast<WebRtcInterface &>(exchanger);
+        try {
+            auto sdp_answer = handler.getAnswerSdp(offer_str);
+            cb(ptr.get(), sdp_answer.data(), nullptr);
+        } catch (std::exception &ex) {
+            cb(ptr.get(), nullptr, ex.what());
+        }
+    });
+#else
+    WarnL << "未启用webrtc功能, 编译时请开启ENABLE_WEBRTC";
+#endif
+}
+
+API_EXPORT void API_CALL mk_webrtc_get_proxy_player_info(mk_proxy_player ctx, on_mk_webrtc_get_proxy_player_info_cb cb) {
+#ifdef ENABLE_WEBRTC
+    assert(ctx && cb);
+    PlayerProxy::Ptr *obj = (PlayerProxy::Ptr *)ctx;
+    auto media_player = obj->get()->getDelegate();
+    if (!media_player) {
+        cb(nullptr, "Media player not found");
+        return;
+    }
+
+    auto webrtc_player_imp = std::dynamic_pointer_cast<WebRtcProxyPlayerImp>(media_player);
+    if (!webrtc_player_imp) {
+        cb(nullptr, "Stream proxy is not WebRTC type");
+        return;
+    }
+
+    auto webrtc_transport = webrtc_player_imp->getWebRtcTransport();
+    if (!webrtc_transport) {
+        cb(nullptr, "WebRTC transport not available");
+        return;
+    }
+
+    webrtc_transport->getTransportInfo([cb](Json::Value transport_info) mutable {
+        if (transport_info.isMember("error")) {
+            cb(nullptr, strdup(transport_info["error"].asCString()));
+            return;
+        }
+        cb(strdup(transport_info.toStyledString().c_str()), "");
+    });
+#else
+    WarnL << "未启用webrtc功能, 编译时请开启ENABLE_WEBRTC";
+#endif
+}
+
+API_EXPORT void API_CALL mk_webrtc_add_room_keeper(
+    const char *room_id, const char *server_host, uint16_t server_port, int ssl, on_mk_webrtc_room_keeper_info_cb cb, void *user_data) {
+    mk_webrtc_add_room_keeper2(room_id, server_host, server_port, ssl, cb, user_data, nullptr);
+}
+
+API_EXPORT void API_CALL mk_webrtc_add_room_keeper2(
+    const char *room_id, const char *server_host, uint16_t server_port, int ssl, on_mk_webrtc_room_keeper_info_cb cb, void *user_data,
+    on_user_data_free user_data_free) {
+#ifdef ENABLE_WEBRTC
+    assert(server_host && server_port && room_id && cb);
+    // server_host: 信令服务器host
+    // server_post: 信令服务器host
+    // room_id: 注册的id,信令服务器会对该id进行唯一性检查
+    std::string server_host_str(server_host), room_id_str(room_id);
+    std::shared_ptr<void> ptr(user_data, user_data_free ? user_data_free : [](void *) {});
+    addWebrtcRoomKeeper(server_host_str, server_port, room_id_str, ssl, [ptr,cb](const SockException &ex, const string &key) mutable {
+        if (ex) {
+            cb(ptr.get(), nullptr, ex.what());
+        } else {
+            cb(ptr.get(), key.c_str(), nullptr);
+        }
+    });
+#else
+    WarnL << "未启用webrtc功能, 编译时请开启ENABLE_WEBRTC";
+#endif
+}
+
+API_EXPORT void API_CALL mk_webrtc_del_room_keeper(const char *room_key, on_mk_webrtc_room_keeper_info_cb cb, void *user_data) {
+    mk_webrtc_del_room_keeper2(room_key,cb,user_data,nullptr);
+}
+
+API_EXPORT void API_CALL
+mk_webrtc_del_room_keeper2(const char *room_key, on_mk_webrtc_room_keeper_info_cb cb, void *user_data, on_user_data_free user_data_free) {
+#ifdef ENABLE_WEBRTC
+    assert(room_key && cb);
+    std::string room_key_str(room_key);
+    std::shared_ptr<void> ptr(user_data, user_data_free ? user_data_free : [](void *) {});
+    delWebrtcRoomKeeper(room_key_str, [room_key_str, ptr, cb](const SockException &ex) mutable {
+        if (ex) {
+            cb(ptr.get(), room_key_str.c_str(), ex.what());
+        }
+        cb(ptr.get(), room_key_str.c_str(), nullptr);
+    });
+#else
+    WarnL << "未启用webrtc功能, 编译时请开启ENABLE_WEBRTC";
+#endif
+}
+
+API_EXPORT void API_CALL mk_webrtc_list_room_keeper(on_mk_webrtc_room_keeper_data_cb cb) {
+#ifdef ENABLE_WEBRTC
+    assert(cb);
+    listWebrtcRoomKeepers([cb](const std::string &key, const WebRtcSignalingPeer::Ptr &p) {
+        Json::Value item = ToJson(p);
+        item["room_key"] = key;
+        cb(strdup(item.toStyledString().c_str()));
+    });
+#else
+    WarnL << "未启用webrtc功能, 编译时请开启ENABLE_WEBRTC";
+#endif
+}
+
+API_EXPORT void API_CALL mk_webrtc_list_rooms(on_mk_webrtc_room_keeper_data_cb cb){
+#ifdef ENABLE_WEBRTC
+    assert(cb);
+    listWebrtcRooms([cb](const std::string &key, const WebRtcSignalingSession::Ptr &p) {
+        Json::Value item = ToJson(p);
+        item["room_id"] = key;
+        cb(strdup(item.toStyledString().c_str()));
+    });
+#else
+    WarnL << "未启用webrtc功能, 编译时请开启ENABLE_WEBRTC";
+#endif
+}

webrtc/WebRtcClient.cpp

@@ -154,6 +154,7 @@ void WebRtcClient::doNegotiateWhepOrWhip() {
 
     _negotiate = make_shared<HttpRequester>();
     _negotiate->setMethod("POST");
+    _negotiate->addHeader("Content-Type", "application/sdp");
     _negotiate->setBody(std::move(offer_sdp));
     _negotiate->startRequester(_url._negotiate_url, [weak_self](const toolkit::SockException &ex, const Parser &response) {
         auto strong_self = weak_self.lock();