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## Summary This PR fixes GStreamer interoperability issues during WebRTC/WHEP negotiation with ZLMediaServer. GStreamer could fail to establish the connection for two separate reasons: 1. ZLMediaServer generated a non-compliant ICE `ufrag`. The generated value contained `_`, which is not a valid ICE character, so GStreamer rejected the SDP. 2. ZLMediaServer did not correctly handle `bundle-only` offers and could answer an accepted bundled m-line with `port=0`, which caused the later WHEP negotiation to fail. ## Changes - Generate ICE `ufrag` values using ICE-compliant characters only. - Preserve and handle `a=bundle-only` correctly during SDP parsing and answer generation. - Return `port=9` instead of `port=0` for accepted bundled m-lines. - Add regression coverage for `bundle-only` SDP handling. - URL-encode `delete_webrtc` query parameters in the returned `Location` header so ICE-safe identifiers remain round-trippable over HTTP. ## Validation - Built with WebRTC and SCTP enabled. - Added regression test: `test_webrtc_regression` - Verified: - ICE-safe identifier round-trip through `delete_webrtc` - `bundle-only` SDP answer generation |
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|---|---|---|
| .. | ||
| 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 | ||
| WebRtcTalk.cpp | ||
| WebRtcTalk.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