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Step 1: Use FFMpeg to push rtmp stream and webRTC stream to SRS separately. When pushing webRTC stream, use FFMpeg's metaRTC plugin.
Step 2: Access port 8080 and modify the webRTC address for playback.
Expect
The expected latency for video playback using webRTC streaming is lower than that of video playback using RTMP streaming, because when streaming with RTMP, the transcoding process in SRS takes approximately 150ms. However, in reality, the latency for webRTC streaming is mostly consistent with that of RTMP streaming, and sometimes even higher.
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使用RTMP推流到SRS, 和使用webRTC推流到SRS, 时延一致未降低。
Using RTMP to push stream to SRS and using webRTC to push stream to SRS, the latency remains consistent without reduction.
Aug 3, 2023
Description
Please description your issue here
SRS Version: 4.0-r4
SRS Log:
RTMP push SRS log
WebRTC push SRS log
Replay
Please describe how to replay the bug?
Step 1: Use FFMpeg to push rtmp stream and webRTC stream to SRS separately. When pushing webRTC stream, use FFMpeg's metaRTC plugin.
Step 2: Access port 8080 and modify the webRTC address for playback.
Expect
The expected latency for video playback using webRTC streaming is lower than that of video playback using RTMP streaming, because when streaming with RTMP, the transcoding process in SRS takes approximately 150ms. However, in reality, the latency for webRTC streaming is mostly consistent with that of RTMP streaming, and sometimes even higher.
TRANS_BY_GPT3
The text was updated successfully, but these errors were encountered: