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>
This commit is contained in:
baigao-X
2025-09-20 16:23:30 +08:00
committed by GitHub
parent 97d2a1fb08
commit 3fb43c5fef
72 changed files with 16912 additions and 10319 deletions

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@@ -45,7 +45,7 @@
## Feature List
### Overview of Features
<img width="800" alt="Overview of Features" src="https://github.com/ZLMediaKit/ZLMediaKit/assets/11495632/481ea769-5b27-495e-bf7d-31191e6af9d2">
<img width="749" alt="Overview of Features" src="https://github.com/user-attachments/assets/7072fe1c-e2b3-47e9-bd50-e5266523edf1">
- RTSP[S]
- RTSP[S] server, supports RTMP/MP4/HLS to RTSP[S] conversion, supports devices such as Amazon Echo Show
@@ -124,6 +124,8 @@
- Supports WebRTC over TCP mode
- Excellent NACK and jitter buffer algorithms with outstanding packet loss resistance
- Supports WHIP/WHEP protocols
- [Supports ice-full, works as a WebRTC client for pulling streams, pushing streams, and P2P mode](./webrtc/USAGE.md)
- [SRT support](./srt/srt.md)
- Others
- Supports rich RESTful APIs and webhook events