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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> |
||
|---|---|---|
| .. | ||
| answer.sdp | ||
| CMakeLists.txt | ||
| DtlsTransport.cpp | ||
| DtlsTransport.hpp | ||
| IceSession.cpp | ||
| IceSession.hpp | ||
| IceTransport.cpp | ||
| IceTransport.hpp | ||
| janus_answer.sdp | ||
| janus_offer.sdp | ||
| logger.h | ||
| Nack.cpp | ||
| Nack.h | ||
| offer-simulcast.sdp | ||
| offer.sdp | ||
| readme.md | ||
| RtpExt.cpp | ||
| RtpExt.h | ||
| RtpMap.h | ||
| SctpAssociation.cpp | ||
| SctpAssociation.hpp | ||
| Sdp.cpp | ||
| Sdp.h | ||
| SrtpSession.cpp | ||
| SrtpSession.hpp | ||
| StunPacket.cpp | ||
| StunPacket.hpp | ||
| TwccContext.cpp | ||
| TwccContext.h | ||
| USAGE.md | ||
| WebRtcClient.cpp | ||
| WebRtcClient.h | ||
| WebRtcEchoTest.cpp | ||
| WebRtcEchoTest.h | ||
| WebRtcPlayer.cpp | ||
| WebRtcPlayer.h | ||
| WebRtcProxyPlayer.cpp | ||
| WebRtcProxyPlayer.h | ||
| WebRtcProxyPlayerImp.h | ||
| WebRtcProxyPusher.cpp | ||
| WebRtcProxyPusher.h | ||
| WebRtcPusher.cpp | ||
| WebRtcPusher.h | ||
| WebRtcSession.cpp | ||
| WebRtcSession.h | ||
| webrtcSignal.txt | ||
| WebRtcSignalingMsg.cpp | ||
| WebRtcSignalingMsg.h | ||
| WebRtcSignalingPeer.cpp | ||
| WebRtcSignalingPeer.h | ||
| WebRtcSignalingSession.cpp | ||
| WebRtcSignalingSession.h | ||
| WebRtcTransport.cpp | ||
| WebRtcTransport.h | ||
致谢与声明
本文件夹下部分文件提取自MediaSoup ，分别为：
-
dtls相关功能：
- DtlsTransport.cpp
- DtlsTransport.hpp
-
srtp相关功能：
- SrtpSession.cpp
- SrtpSession.hpp
-
datachannel相关功能：
- SctpAssociation.cpp
- SctpAssociation.hpp
以上源码有一定的修改和裁剪，感谢MediaSoup开源项目及作者， 用户在使用本项目的同时，应该同时遵循MediaSoup的开源协议。
同时,在此也感谢开源项目easy_webrtc_server 及作者， 在集成MediaSoup相关代码前期，主要参考这个项目。
另外，感谢big panda 开发并贡献的webrtc js测试客户端(www/webrtc目录下文件)， 其开源项目地址为：https://gitee.com/xiongguangjie/zlmrtcclient.js