I have a client using webrtc to send a stream through janode( videoroom plugin) by socket (ws:host:4444) and then janode will forward the rtp to a streaming.
And the user will watch the stream via the second streaming example (streaming plugin) ( ws:host:4445)
But it seems that during the rtp video relay, there are a lot of lost frames and the video keeps jerking. Is there any way to increase streaming latency to reduce frame loss (no stuttering).
Thanks,
Thiên
Is the publisher’s stream okay? You can check it by setting up a subscriber in the room or getting a mjr recording of it and replaying.
Are the videoroom and streaming instances on the same host? If not there might be some networking issue between the videoroom and the streaming nodes. Try a packet capture on the streaming mountpoint and check with Wireshark if everything looks good.
Sorry it took so long to reply. Recently I was looking for a way to save a video as mjr and use ffmpeg to convert to mp4. I noticed the video stuttering and losing frames right from the broadcast (publisher’s stream). Is there any way to improve this work? Is it possible to delay the video playback a bit (5~10 seconds) to ensure the frames are fully loaded?
Thanks,
Thiên