cqm
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03526f141e
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Merge branch 'master' of https://github.com/ZLMediaKit/ZLMediaKit into feature/transcode2
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# Conflicts:
# conf/config.ini
# src/Codec/Transcode.cpp
# src/Common/MediaSource.h
# src/Common/MultiMediaSourceMuxer.cpp
# src/Common/MultiMediaSourceMuxer.h
# src/Common/macros.h
# webrtc/WebRtcPusher.cpp
# webrtc/WebRtcTransport.cpp
# webrtc/WebRtcTransport.h
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2026-04-03 09:56:05 +08:00 |
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xia-chu
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6c82601fe2
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修复状态紊乱导致访问空指针的bug
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有用户反馈崩溃在此处，尝试修复之
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2026-03-01 12:11:47 +08:00 |
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xia-chu
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fb491f3e79
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新增webrtc配置选项nackAudioRtpSize、preferred_tcp
nackAudioRtpSize: 音频nack包中rtp个数，减小此值可以让nack包响应更灵敏
preferred_tcp: 是否优先采用webrtc over tcp模式
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2026-01-09 11:39:22 +08:00 |
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xia-chu
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ae3d551c8a
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解决webrtc丢包重传误判为重放攻击的问题
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2026-01-09 11:35:37 +08:00 |
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xia-chu
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1e6a8964cc
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修复WebRtcSession对象可能泄露的bug (#4596)
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2025-12-17 21:42:25 +08:00 |
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xia-chu
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5165ac4f74
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初步支持语音双向对讲webrtc插件
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2025-11-25 22:59:58 +08:00 |
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xia-chu
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7b2bd221ea
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Revert "fix: handle BUNDLE filed compatibility (#4488)"
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This reverts commit 0b57a573c6.
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2025-11-25 10:41:39 +08:00 |
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mtdxc
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70a2140f27
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修复WebRtcProxyPusher速率为0 bug (#4558)
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2025-11-22 08:22:42 +08:00 |
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mtdxc
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d8a1c70e2d
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Revert "sr发送限频" (#4533 #4521)
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This reverts commit 21d1ea9676.
之前代码改多了，会导致空指针错误，现进行还原
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2025-11-04 18:39:52 +08:00 |
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xia-chu
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c9d3c5b516
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转发rtp时，关闭srtp防重放攻击逻辑
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2025-10-24 16:45:41 +08:00 |
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baigao
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0b57a573c6
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fix: handle BUNDLE filed compatibility (#4488)
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2025-10-09 11:38:05 +08:00 |
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mtdxc
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b9d513f9aa
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采用定时器发送rr
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2025-09-29 19:06:47 +08:00 |
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mtdxc
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21d1ea9676
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sr发送限频
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2025-09-29 19:06:47 +08:00 |
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mtdxc
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13f603328f
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rtc拉流代理支持显示sockInfo和码率
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2025-09-29 19:06:47 +08:00 |
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mtdxc
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016f6e0e59
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修复拼写错误，并精简WebRtcClient无用代码
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2025-09-29 19:06:47 +08:00 |
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Lidaofu
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493714bc7d
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c sdk 新增webrtc相关函数 (#4473)
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另外调整函数位置，whip、whep请求设置Content-Type为application/sdp
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2025-09-24 17:40:43 +08:00 |
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baigao-X
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3fb43c5fef
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feat: 增加webrtc代理拉流 (#4389)
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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>
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2025-09-20 16:23:30 +08:00 |
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xia-chu
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5686027fc2
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修复MediaSource::close线程安全问题
主动或无人观看关闭流可能会由于线程安全问题导致崩溃
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2025-09-09 22:02:20 +08:00 |
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weishao
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ceb78bd54c
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feat: add H.265 profile parameters support for WebRTC SDP negotiation
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2025-08-22 20:38:17 +08:00 |
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xia-chu
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57fede9508
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修复编译警告
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2025-08-09 15:42:28 +08:00 |
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xia-chu
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c98137e505
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优化线程安全问题
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2025-06-29 17:56:10 +08:00 |
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custompal
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f0204ea3b3
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播放器(reader)setGetInfoCB时统一使用Session对象 (#4195)
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Co-authored-by: custompal <chenrengen@gosuncn.com>
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2025-03-21 15:22:48 +08:00 |
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Dw9
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479a3fb9bb
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Remove B-frames when playing WebRTC streams (#4176)
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1. webrtc 播放移除了 h264 rtp中的b 帧
2. 增加了 rtc.bfilter 配置
ref: ISO_IEC_14496-10-AVC-2012
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2025-03-03 14:37:57 +08:00 |
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cqm
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89e21038c2
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Merge branch 'master' of github.com:ZLMediaKit/ZLMediaKit into feature/transcode2
# Conflicts:
# src/Common/MediaSource.h
# src/Common/config.h
# src/Rtsp/RtspMediaSourceImp.h
# src/Rtsp/RtspMediaSourceMuxer.h
# src/Rtsp/RtspMuxer.cpp
# webrtc/WebRtcTransport.cpp
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2024-09-22 11:40:39 +08:00 |
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alex
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4152dcd409
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AI automatically translates all comments in the code into English (#3917)
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2024-09-19 14:53:50 +08:00 |
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cqm
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d329f76edf
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Merge branch 'master' of https://gitee.com/xia-chu/ZLMediaKit into transcode2
# Conflicts:
# src/Common/config.cpp
# src/Common/config.h
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2024-08-19 18:03:03 +08:00 |
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xia-chu
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a14e40f8d2
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修复webrtc rtp重发列队可能由于rtp时间戳异常导致内存溢出的bug
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2024-07-21 19:16:34 +08:00 |
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cqm
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e6838f24af
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fixed build issue
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2024-07-15 08:47:31 +08:00 |
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cqm
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2ffdfab71e
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Merge branch 'master' of github.com:ZLMediaKit/ZLMediaKit into transcode2
# Conflicts:
# CMakeLists.txt
# conf/config.ini
# src/Common/MediaSink.cpp
# src/Common/MediaSink.h
# src/Common/MediaSource.cpp
# src/Common/MultiMediaSourceMuxer.h
# src/Common/config.cpp
# src/Common/config.h
# src/Extension/AAC.cpp
# src/Extension/AAC.h
# src/Rtsp/RtpCodec.h
# src/Rtsp/RtspMuxer.cpp
# src/Rtsp/RtspMuxer.h
# webrtc/Nack.cpp
# webrtc/WebRtcTransport.cpp
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2024-07-15 08:47:30 +08:00 |
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xiongziliang
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2ead272187
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修复webrtc播放时生成config rtp线程安全的bug
rtsp直接代理时会从config frame生成rtp，在获取VideoTrack配置帧列表时存在线程安全风险；
同时简化getConfigFrames函数代码，去除缓存逻辑。
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2024-06-22 13:04:06 +08:00 |
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xiongziliang
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1ef3642ed4
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优化webrtc jitter buffer相关参数
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2024-06-22 13:04:06 +08:00 |
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Xiaofeng Wang
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84dbe4b076
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Send config frames once before sending DirectProxy RTP packets (#3611)
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2024-06-16 09:37:33 +08:00 |
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Luke
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fbac3f9ba0
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RC4密码套件漏洞 (#3590)
禁止使用RC4
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2024-06-09 10:45:40 +08:00 |
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gongluck
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5df2b80757
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Nack相关参数可配置化 (#3510 #3507)
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2024-05-12 18:15:11 +08:00 |
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xia-chu
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2d66dd7cb3
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Fixing the problem of WebRTC handshake failure caused by unsupported RTP extensions (#3486)
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2024-04-20 22:49:27 +08:00 |
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Dw9
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edca662208
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fix webrtc echo error (#3442)
遵循着pr #3360 的修改方式解决webrtc echo模式失败的问题
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2024-04-05 22:11:50 +08:00 |
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xiongziliang
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7aaafa18e7
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Format code
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2024-03-23 23:08:10 +08:00 |
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xiongziliang
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d8893877b2
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Delete invalid code
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2024-03-23 22:56:12 +08:00 |
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johzzy
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029813402d
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feat: update negotiateSdp and WebRtcArgs (#3371)
- update negotiateSdp
- update HttpAllArgs and alias
- update onRtcConfigure
- define setWebRtcArgs, handle set_webrtc_cands and setLocalIp
---------
Co-authored-by: xiongziliang <771730766@qq.com>
Co-authored-by: KkemChen <kkemchen@qq.com>
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2024-03-23 22:46:30 +08:00 |
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xia-chu
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db4c570d19
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WebRTC audio preferred PCMA
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2024-03-22 20:50:07 +08:00 |
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小强先生
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8e16a698b6
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降低webrtc握手未结束，数据先到的日志等级 (#3368)
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2024-03-13 10:57:43 +08:00 |
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johzzy
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1e39594335
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fix for https://bugs.chromium.org/p/webrtc/issues/detail?id=15845 (#3360)
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2024-03-10 21:34:03 +08:00 |
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gongluck
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5a6364bae2
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Add datachannel c apis and callbacks(#3328)
增加datachannel数据收发的回调通知 #3326，和控制datachannel回显的开关
---------
Co-authored-by: xiongziliang <771730766@qq.com>
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2024-03-02 16:52:51 +08:00 |
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waken
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fd1ebb1a51
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Auto get icecand ip address from http request (#3251)
Get ip address from http `Host` header, and set it to icecand ip for webrtc
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2024-01-26 10:26:30 +08:00 |
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老衲不出家
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62641f426b
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Fix a bug in WebRTC where the DTLS certificate is too large(big than MTU) (#3250)
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2024-01-24 15:14:23 +08:00 |
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sandro-qiang
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aeda47ec8c
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修正SctpAssociation.cpp的访问越界 (#3182)
ZLMediaKit/ZLMediaKit#3181
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2024-01-05 15:33:51 +08:00 |
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Dw9
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7c42ac4d79
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Set the default port for WebRTC (#3175)
Set the default port for WebRTC to 8000 to prevent the generated default configuration file from disabling WebRTC features.
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2023-12-29 10:53:47 +08:00 |
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夏楚
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bbe8f4a018
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release 8.0
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2023-12-09 16:23:51 +08:00 |
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夏楚
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76aebd74c6
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整理MediaSource协议类型
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2023-12-02 16:31:40 +08:00 |
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LeiZhi.Mai
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586bb73912
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兼容webrtc offer sdp中重复行的情况 (#3038)
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2023-12-01 10:44:59 +08:00 |
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