Dw9
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e8a870e215
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rtc port默认8000
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2023-07-14 20:15:11 +08:00 |
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xia-chu
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af0ebf4633
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优化Parser代码命名
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2023-06-10 14:55:40 +08:00 |
|
Johnny
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0232caf068
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add MediaTuple
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2023-05-30 11:01:27 +08:00 |
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Johnny
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f4ee607feb
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add RtspMediaSource::Clone
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2023-05-30 11:01:27 +08:00 |
|
Johnny
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bfec1b1e0e
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fix typo
|
2023-05-02 20:41:13 +08:00 |
|
xia-chu
|
8c049d4e28
|
实现whip/whep delete相关功能
通过whip/whep 回复http头中的Location url进行删除资源
新增delete token随机数实现删除鉴权
|
2023-04-28 22:36:58 +08:00 |
|
xia-chu
|
d2349f01bd
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使用static_pointer_cast优化性能
|
2023-04-28 22:04:38 +08:00 |
|
xia-chu
|
cdf97e7605
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更新zltoolkit,适配最新代码
|
2023-04-28 22:04:09 +08:00 |
|
xiongziliang
|
c8a20f85f6
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直接打印SockException类
|
2023-04-23 00:10:18 +08:00 |
|
Johnny
|
82bc416546
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add exchangeSdp
|
2023-04-21 20:40:37 +08:00 |
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xia-chu
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70ad9cb0b6
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设置webrtc端口缺省配置为0(不开启)
|
2023-03-13 21:00:00 +08:00 |
|
xia-chu
|
4cc2982a2c
|
重写jitter buffer算法,修复seq回环时可能丢包的bug
|
2023-03-11 10:17:31 +08:00 |
|
mtdxc
|
4783ac0808
|
修复webrtc多候选地址无法来回切换的bug (#2266)
最后一个连通的候选地址会被赋值并锁定为_selected_session,如果之前的候选地址再发送数据,将通过_selected_session回复,导致无法切换为旧的候选地址。
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2023-03-03 11:18:21 +08:00 |
|
xiongziliang
|
c2e1083493
|
复用rtp/rtcp判断逻辑过滤非rtp包: #2247
|
2023-02-28 22:23:30 +08:00 |
|
xiongguangjie
|
96f549ab32
|
avoid bug for #2159
|
2022-12-15 19:17:09 +08:00 |
|
monktan89
|
f6a36180b4
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修复webrtc转发可能内存越界的bug: #2159
|
2022-12-14 15:08:59 +08:00 |
|
monktan89
|
a12b7c8021
|
修正代码拼写错误
|
2022-11-30 18:24:35 +08:00 |
|
mtdxc
|
754073918a
|
Header refactor (#2115)
* 优化MultiMediaSourceMuxer头文件包含
* 将MediaSinkDelegate和Demux移到MediaSink中
* MediaSource头文件重构, 独立出PacketCache.h
精简Frame和Track的头文件
* Rtmp头文件重构
* Rtsp头文件重构
* webrtc头文件重构
* 规范.h头文件包含,并将其移到.cpp中:
- 尽量不包含Common\config.h
- Util\File.h
- Rtsp/RtspPlayer.h
- Rtmp/RtmpPlayer.h
* 删除多余的Stamp.h和Base64包含
|
2022-11-29 11:07:13 +08:00 |
|
johzzy
|
bc4fa2bc53
|
fix(rtcp/bye): 收到rtcp bye时不断开webrtc连接 (#2123)
bye(203) 会在 sender audio track mute 时出现, 不能作为 shutdown 的依据
|
2022-11-28 17:52:37 +08:00 |
|
Dw9
|
fef9d31631
|
webrtc支持通过http参数指定是否优先tcp模式 (#2105)
* webrtc push/play支持通过http参数指定tcp
* force_tcp改成perferred_tcp
Co-authored-by: xiongziliang <771730766@qq.com>
|
2022-11-27 12:43:16 +08:00 |
|
ziyue
|
9fd5152aa1
|
确保webrtc关闭udp模式时可用
|
2022-11-19 09:52:10 +08:00 |
|
ziyue
|
d910c80a00
|
优化webrtc iceandidate端口相关代码
|
2022-11-19 09:46:04 +08:00 |
|
xiongguangjie
|
0d6fa1281a
|
add rtc tcp port config and ignore candidate when port is 0
|
2022-11-19 01:51:53 +08:00 |
|
Dw9
|
47530ce830
|
新增支持webrtc over tcp模式 (#2092)
* webrtc server/session/cadidate 改为tcp
* 先屏蔽检查isCurrentThread
* 接受和发送的数据处理tcp 2字节头
* 处理rtc tcp 分片
* 完善webrtc over tcp
* 精简rtp服务器相关代码
* 适配webrtc AV1编码: #2091
* webrtc tcp模式支持Firefox
* webrtc tcp模式支持线程安全
* c sdk支持webrtc tcp
Co-authored-by: ziyue <1213642868@qq.com>
|
2022-11-18 22:52:57 +08:00 |
|
xiongziliang
|
a916760ac3
|
整理webrtc c接口
|
2022-10-06 12:35:14 +08:00 |
|
ziyue
|
0b355759de
|
整理webrtc相关代码命名空间
|
2022-09-18 21:03:05 +08:00 |
|
xiongguangjie
|
d04a6d4355
|
rtc receive sr reset alive ticker
|
2022-09-09 23:04:28 +08:00 |
|
xiongguangjie
|
7f86299c80
|
rtc stop push unregister mediasource
|
2022-09-07 19:10:06 +08:00 |
|
xiongziliang
|
20f1275c58
|
rtsp拉流、rtp单端口推流新增支持获取丢包率: #1877
|
2022-09-03 16:32:09 +08:00 |
|
Dw9
|
b7ef766701
|
支持单独的datachannel通道 (#1894)
* 支持单独的datachannel 通道
* 当仅有datachannel时 ,忽略rtp和rtcp超时
* 单独开启datachannel时,通过dtls包维持心跳
Co-authored-by: xiongziliang <771730766@qq.com>
|
2022-08-20 10:28:19 +08:00 |
|
Xiaofeng Wang
|
0fc38bbadb
|
避免对外传递 srtp 依赖
|
2022-07-26 00:04:03 +08:00 |
|
xiongguangjie
|
cffc0743d7
|
format code use clang-format
|
2022-07-14 22:49:31 +08:00 |
|
xiongguangjie
|
602b475a3c
|
for rtc send can receive rtcp xr, player can calculate rtt , for metaRTC player send rtcp xr
|
2022-07-12 19:26:17 +08:00 |
|
Haibo Chen
|
576056c465
|
get ip from env, for externIP in file 'config.ini' (#1786)
|
2022-07-09 21:44:57 +08:00 |
|
xiongziliang
|
94839fd93c
|
添加bom头
|
2022-06-18 14:34:58 +08:00 |
|
xiongziliang
|
df7a3242bb
|
修复获取webrtc推流丢包率除0的bug
|
2022-06-11 14:23:43 +08:00 |
|
xiongziliang
|
b6d3ec5251
|
完善获取丢包率pull request; 新增MediaSource::getOwnerPoller接口
|
2022-06-11 12:56:03 +08:00 |
|
Leon
|
be995f9cd2
|
!17 【功能请求】 /index/api/getMediaList接口 增加LOSS字段用于统计每个轨道丢包率反馈
* update webrtc/WebRtcPusher.h.
* update webrtc/WebRtcPusher.cpp.
* update webrtc/WebRtcTransport.h.
* update webrtc/WebRtcTransport.cpp.
* update src/Common/MediaSource.h.
* update src/Common/MediaSource.cpp.
* update server/WebApi.cpp.
|
2022-06-11 04:31:06 +00:00 |
|
xiongziliang
|
b23cbaa0f8
|
on_publish hook新增continue_push_ms参数,用于断连续推延时控制
|
2022-06-04 11:06:35 +08:00 |
|
mtdxc
|
6cab65f7b0
|
支持SDP munging方式simulicast (#1639)
|
2022-05-18 11:26:57 +08:00 |
|
mtdxc
|
301cbf0a83
|
支持多个rtc候选地址 (#1622)
* 支持多个rtc候选地址
* fixed missing extern_ips check
|
2022-05-12 15:21:09 +08:00 |
|
xiongziliang
|
d66845425c
|
完善ipv6网络适配
|
2022-05-08 00:26:01 +08:00 |
|
xiongziliang
|
fb003714e9
|
新增webrtc datachannel功能: #1216
|
2022-05-04 22:15:21 +08:00 |
|
xiongziliang
|
1ed793fe97
|
初步实现webrtc datachannel sdp握手
|
2022-04-03 17:12:23 +08:00 |
|
xiongziliang
|
4dc621e1bb
|
转协议选项抽象为ProtocolOption对象
|
2022-03-12 13:24:23 +08:00 |
|
ziyue
|
d88fe077ed
|
on_publish hook新增originType字段
|
2022-03-02 18:03:44 +08:00 |
|
xiongziliang
|
497bf289b6
|
完善rtc网络迁移日志打印
|
2022-02-27 21:40:32 +08:00 |
|
xiongziliang
|
f8ef5579ee
|
整理webrtc相关日志
|
2022-02-27 21:26:49 +08:00 |
|
xiongziliang
|
ba5ca8f5c7
|
修复rtp seq重复导致rtp nack重传列队异常的bug
|
2022-02-27 02:07:25 +08:00 |
|
夏楚
|
3a2a60c40a
|
防止开启模拟丢包代码时,生成twcc rtcp紊乱的bug
|
2022-02-27 00:33:39 +08:00 |
|