feat: 增加webrtc代理拉流 (#4389)

- 增加客户端模式，支持主动拉流、推流： - addStreamProxy接口新增支持whep主动拉流，拉流地址目前只兼容zlm的whep url。 - addStreamPusherProxy接口新增支持whip主动推流，推流地址目前只兼容zlm的whip url。 - 以上推流url格式为webrtc[s]://server_host:server_port/app/stream_id?key=value, 内部会自动转换为http[s]://server_host:server_port/index/api/[whip/whep]?app=app&stream=stream_id&key=value。 - 增加WebRtc p2p 模式： - 增加 ICE FULL模式。 - 增加STUN/TURN 服务器。 - 增加websocket 信令。 - 增加P2P代理拉流。 --------- Co-authored-by: xia-chu <771730766@qq.com> Co-authored-by: mtdxc <mtdxc@126.com> Co-authored-by: cqm <cqm@97kid.com>
2026-06-30 06:42:22 +08:00 · 2025-09-20 16:23:30 +08:00
parent 97d2a1fb08
commit 3fb43c5fef
72 changed files with 16912 additions and 10319 deletions
--- a/webrtc/WebRtcClient.h
+++ b/webrtc/WebRtcClient.h
@@ -0,0 +1,105 @@
+/*
+ * 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 ZLMEDIAKIT_WEBRTC_CLIENT_H
+#define ZLMEDIAKIT_WEBRTC_CLIENT_H
+
+#include "Network/Socket.h"
+#include "Poller/Timer.h"
+#include "Util/TimeTicker.h"
+#include "Http/HttpRequester.h"
+#include "Sdp.h"
+#include "WebRtcTransport.h"
+#include "WebRtcSignalingPeer.h"
+#include <memory>
+#include <string>
+
+namespace mediakit {
+
+// 解析webrtc 信令url的工具类
+class WebRTCUrl {
+public:
+    bool _is_ssl;
+    std::string _full_url;
+    std::string _negotiate_url; // for whep or whip
+    std::string _delete_url; // for whep or whip
+    std::string _target_secret;
+    std::string _params;
+    std::string _host;
+    uint16_t _port;
+    std::string _vhost;
+    std::string _app;
+    std::string _stream;
+    WebRtcTransport::SignalingProtocols _signaling_protocols = WebRtcTransport::SignalingProtocols::WHEP_WHIP;
+    std::string _peer_room_id; // peer room_id
+
+public:
+    void parse(const std::string &url, bool isPlayer);
+
+private:
+};
+
+namespace Rtc {
+typedef enum {
+    Signaling_Invalid   = -1,
+    Signaling_WHEP_WHIP = 0,
+    Signaling_WEBSOCKET = 1,
+} eSignalingProtocols;
+} // namespace Rtc
+
+// 实现了webrtc代理功能
+class WebRtcClient : public std::enable_shared_from_this<WebRtcClient> {
+public:
+    using Ptr = std::shared_ptr<WebRtcClient>;
+
+    WebRtcClient(toolkit::EventPoller::Ptr poller);
+    virtual ~WebRtcClient();
+
+    const toolkit::EventPoller::Ptr &getPoller() const { return _poller; }
+    void setPoller(toolkit::EventPoller::Ptr poller) { _poller = std::move(poller); }
+
+    // 获取WebRTC transport，用于API查询
+    const WebRtcTransport::Ptr &getWebRtcTransport() const { return _transport; }
+
+protected:
+    virtual bool isPlayer() = 0;
+    virtual void startConnect();
+    virtual void onResult(const toolkit::SockException &ex) = 0;
+    virtual void onNegotiateFinish();
+    virtual float getTimeOutSec();
+
+    void doNegotiate();
+    void doNegotiateWebsocket();
+    void doNegotiateWhepOrWhip();
+    void checkIn();
+    void doBye();
+    void doByeWhepOrWhip();
+    void checkOut();
+
+    void gatheringCandidate(IceServerInfo::Ptr ice_server);
+    void connectivityCheck();
+    void candidate(const std::string &candidate, const std::string &ufrag, const std::string &pwd);
+
+protected:
+    toolkit::EventPoller::Ptr _poller;
+
+    // for _negotiate_sdp
+    WebRTCUrl _url;
+    HttpRequester::Ptr _negotiate = nullptr;
+    WebRtcSignalingPeer::Ptr _peer = nullptr;
+    WebRtcTransport::Ptr _transport = nullptr;
+    bool _is_negotiate_finished = false;
+
+private:
+    std::map<std::string /*candidate key*/, toolkit::SocketHelper::Ptr> _socket_map;
+};
+
+} /*namespace mediakit */
+#endif /* ZLMEDIAKIT_WEBRTC_CLIENT_H */