Live Stream getting freeze

[mojostreaming]

Hello Max
We are having a issue of live stream freeze while streaming. Is there any config settings that we can make to overcome this live stream freeze issue?
Something like increasing amount of: video_incoming_buffer_size
Or any thing else than can help to fix this issue?

 

[Max]

Good day.
Please clarify the case:
- what WCS build do you use?
- how do you publish a stream: WebRTC, RTMP, captured RTSP or RTMP stream from other server?
- how do you play the stream: via WebRTC, RTMP, HLS, RTSP?
- what is the stream parameters: resolution, FPS, bitrate?
- what is the stream publishing source: webcam, screen sharing?

Is there any config settings that we can make to overcome this live stream freeze issue?

Usually, stream freezes due to publisher or player channel issues, when stream resolution/bitrate does not fit to the channel bandwidth, or there is packets loss.
Please try to reduce stream publishing resolution (for example, from 1080p to 480p) an bitrate (for example, from 2500 kbps to 1000 kbps). Also, please try to use another network to publish/play a stream, for example, use Wi-Fi instead of mobile network, or wired connection instead of Wi-Fi.
You can switch to TCP publishing/playback to prevent packet loss:

session.createStream({
    name: streamName,
    display: localVideo,
    cacheLocalResources: true,
    receiveVideo: false,
    receiveAudio: false,
    transport: "TCP"
    ...
}).publish();

session.createStream({
    name: streamName,
    display: localVideo,
    transport: "TCP"
    ...
}).play();

You can also check the channel quality using Media Devices example as described here.

Something like increasing amount of: video_incoming_buffer_size

This setting affects audio and video synchronization only

 

[mojostreaming]

Hello Max

For the questions you have asked , and I will follow the suggested steps and check if it resolve the issue.

- what WCS build do you use?
=>version == 5.1
- how do you publish a stream: WebRTC, RTMP, captured RTSP or RTMP stream from other server?
=> WebRTC
- how do you play the stream: via WebRTC, RTMP, HLS, RTSP?
=> HLS
- what is the stream parameters: resolution, FPS, bitrate?
=>I'm using default values for these.
- what is the stream publishing source: webcam, screen sharing?
=> webcam

 

[Max]

- what WCS build do you use?
=>version == 5.1

Please update to the latest build from this page and check if problem still persists. If yes, collect a report as described here including client debug logs and send using this private form.

 

[mojostreaming]

Please update to the latest build from this page and check if problem still persists. If yes, collect a report as described here including client debug logs and send using this private form.

We already updated to the the latest version but there were issues coming with latest version so we had to downgrade to 5.1

 

[junkship]

Hi Max
As to MojoStreaming's reply...
We downgraded from the latest version after chatting to you on the thread:
WebSocket connection to 'wss://host:8443/' failed: Error during WebSocket handshake: Unexpected response code: 302
See #19
Thanks

 

[Max]

Have you tried the solution we recommended in this post?
We still recommend you to update WCS to 5.2 because 5.1 is obsolete and is not supported now. The common recommendations for both versions are in this post, but channel quality detection is not supported in 5.1.

 

[mojostreaming]

Have you tried the solution we recommended in this post?
We still recommend you to update WCS to 5.2 because 5.1 is obsolete and is not supported now. The common recommendations for both versions are in this post, but channel quality detection is not supported in 5.1.

Yes I have tried the solution recommended on that post. but it did not fixed the issues so I had to downgrade.

 

[mojostreaming]

Good day.
Please clarify the case:
- what WCS build do you use?
- how do you publish a stream: WebRTC, RTMP, captured RTSP or RTMP stream from other server?
- how do you play the stream: via WebRTC, RTMP, HLS, RTSP?
- what is the stream parameters: resolution, FPS, bitrate?
- what is the stream publishing source: webcam, screen sharing?

Usually, stream freezes due to publisher or player channel issues, when stream resolution/bitrate does not fit to the channel bandwidth, or there is packets loss.
Please try to reduce stream publishing resolution (for example, from 1080p to 480p) an bitrate (for example, from 2500 kbps to 1000 kbps). Also, please try to use another network to publish/play a stream, for example, use Wi-Fi instead of mobile network, or wired connection instead of Wi-Fi.
You can switch to TCP publishing/playback to prevent packet loss:

session.createStream({
    name: streamName,
    display: localVideo,
    cacheLocalResources: true,
    receiveVideo: false,
    receiveAudio: false,
    transport: "TCP"
    ...
}).publish();

session.createStream({
    name: streamName,
    display: localVideo,
    transport: "TCP"
    ...
}).play();

I have tried these settings and set the bitrate settings as mentioned below in the flashphoner properties. Resolution is: 1280x720. But stream getting blurred while watching the live stream.
webrtc_cc_min_bitrate = 1500
webrtc_cc_max_bitrate = 2000

 

[junkship]

And we have to have high quality streaming - I would ultimately like to stream in 4K

 

[Max]

Yes I have tried the solution recommended on that post. but it did not fixed the issues so I had to downgrade.

Please clarify what exactly does not work. Provide SSH access using this form if necessary.

And we have to have high quality streaming - I would ultimately like to stream in 4K

All the fixes concerning 4K streaming are in 5.2. 5.1 branch was not even tested for 4K streaming, so you should upgrade.
Please read this article about 4K streaming issue. Briefly, 4K requires:
- TCP transport
- bitrate 5-10 Mbps
- channel bandwidth at least 20 Mbps, wired channel is desirable
- if transcoding is necessary (to play as HLS with FPS and keyframe equalizing for example), it should be done on separate server