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- 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>
106 lines
3.1 KiB
C++
Executable File
106 lines
3.1 KiB
C++
Executable File
/*
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* Copyright (c) 2016-present The ZLMediaKit project authors. All Rights Reserved.
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*
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* This file is part of ZLMediaKit(https://github.com/ZLMediaKit/ZLMediaKit).
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*
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* Use of this source code is governed by MIT-like license that can be found in the
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* LICENSE file in the root of the source tree. All contributing project authors
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* may be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef ZLMEDIAKIT_WEBRTC_CLIENT_H
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#define ZLMEDIAKIT_WEBRTC_CLIENT_H
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#include "Network/Socket.h"
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#include "Poller/Timer.h"
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#include "Util/TimeTicker.h"
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#include "Http/HttpRequester.h"
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#include "Sdp.h"
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#include "WebRtcTransport.h"
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#include "WebRtcSignalingPeer.h"
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#include <memory>
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#include <string>
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namespace mediakit {
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// 解析webrtc 信令url的工具类
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class WebRTCUrl {
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public:
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bool _is_ssl;
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std::string _full_url;
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std::string _negotiate_url; // for whep or whip
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std::string _delete_url; // for whep or whip
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std::string _target_secret;
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std::string _params;
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std::string _host;
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uint16_t _port;
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std::string _vhost;
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std::string _app;
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std::string _stream;
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WebRtcTransport::SignalingProtocols _signaling_protocols = WebRtcTransport::SignalingProtocols::WHEP_WHIP;
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std::string _peer_room_id; // peer room_id
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public:
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void parse(const std::string &url, bool isPlayer);
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private:
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};
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namespace Rtc {
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typedef enum {
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Signaling_Invalid = -1,
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Signaling_WHEP_WHIP = 0,
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Signaling_WEBSOCKET = 1,
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} eSignalingProtocols;
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} // namespace Rtc
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// 实现了webrtc代理功能
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class WebRtcClient : public std::enable_shared_from_this<WebRtcClient> {
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public:
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using Ptr = std::shared_ptr<WebRtcClient>;
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WebRtcClient(toolkit::EventPoller::Ptr poller);
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virtual ~WebRtcClient();
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const toolkit::EventPoller::Ptr &getPoller() const { return _poller; }
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void setPoller(toolkit::EventPoller::Ptr poller) { _poller = std::move(poller); }
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// 获取WebRTC transport，用于API查询
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const WebRtcTransport::Ptr &getWebRtcTransport() const { return _transport; }
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protected:
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virtual bool isPlayer() = 0;
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virtual void startConnect();
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virtual void onResult(const toolkit::SockException &ex) = 0;
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virtual void onNegotiateFinish();
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virtual float getTimeOutSec();
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void doNegotiate();
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void doNegotiateWebsocket();
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void doNegotiateWhepOrWhip();
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void checkIn();
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void doBye();
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void doByeWhepOrWhip();
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void checkOut();
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void gatheringCandidate(IceServerInfo::Ptr ice_server);
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void connectivityCheck();
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void candidate(const std::string &candidate, const std::string &ufrag, const std::string &pwd);
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protected:
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toolkit::EventPoller::Ptr _poller;
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// for _negotiate_sdp
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WebRTCUrl _url;
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HttpRequester::Ptr _negotiate = nullptr;
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WebRtcSignalingPeer::Ptr _peer = nullptr;
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WebRtcTransport::Ptr _transport = nullptr;
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bool _is_negotiate_finished = false;
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private:
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std::map<std::string /*candidate key*/, toolkit::SocketHelper::Ptr> _socket_map;
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};
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} /*namespace mediakit */
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#endif /* ZLMEDIAKIT_WEBRTC_CLIENT_H */
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