Identify and fix issues that may occur after VideoWhisper solution was configured correctly, tested and running.
If you don’t have it installed, yet get a plan for a turnkey solution, in example for HTML5 Videochat .
Before troubleshooting make sure you have latest plugins, solution installed so issues are not related to older versions. As technology, browsers, streaming servers update, solution is also updated to match and older versions may no longer work.
Streaming issues can have various causes: setup configuration settings, user internet connection to streaming server, network conditions and suitability of streaming protocol, browser type and version.
Here’s some possible issues and steps to identify their cause and possible fixes:
Video pixelation, low quality
1. In HTML5 Videochat app , toggle Settings and check selected streaming resolution & bitrate in broadcast panel and make sure these are not configured to low values. Higher bitrate and resolution should provide higher quality. Maximum bitrate is limited by license and hosting plan.
2. Also check real streaming bitrate measurements. Toggle Settings to get measurements both for Broadcast and Playback panels.
+ Verify that measured bitrate is close to selected bitrate.
+ Use Chrome on PC as that also provides WebRTC statistics like packet loss, latency, jitter.
3. Try adjusting streaming bitrate and see if real bitrate is achieved based on new settings. Maximum bitrate is limited by license and hosting plan.
– WebRTC also adapts quality depending on available connection and network conditions for UDP.
Congested networks and Wi-Fi / mobile depending on signal may produce packet loss when using WebRTC UDP. Packet loss results in interruptions, pixelation, lower quality, automated bitrate downgrade (forced by browser).
4. Try RTMP TCP broadcasting with OBS / GoCoder or other encoders, as mentioned below. TCP resends packets, fixing signal issues related to WiFi or mobile connection.
5. If issues occur both for WebRTC and RTMP streaming, measure your internet connection (see instructions below).
Broadcaster streaming interruptions, frequent errors, slow website while streaming
Some broadcasters may experience issues due to their internet connection speed, location (very far from streaming server). Having a lower connection requires adjusting maximum streaming bitrate, so it doesn’t consume all available bandwidth.
1) Do a speed test from broadcasting location to a location near streaming server.
1. Go to https://www.speedtest.net .
2. Change Server and search for a server in Beauharnois (North America).
3. Press GO to start measurement.
3. Get measurement link from top left icon and share with our staff.
Broadcaster upload connection needs to handle video + audio stream and also other interactions and web requests.
2) In some network conditions UDP streaming may not work at all or provide low bitrate and reliability (showing as pixelation, interruptions).
Broadcaster can download OBS for PC / GoCoder for mobile per instructions in Broadcast tab to broadcast with RTMP TCP instead of WebRTC UDP.
Connection to sever is high and streaming quality is low/DISRUPTED, although configured high bitrate in settings
– Check live bitrate stats in HTML5 Videochat app.
– Try OBS / GoCoder RTMP streaming.
If connection bitrate is high and live streaming bitrate is lower than configured, issue could be related to network conditions and WebRTC protocol streaming over UDP. For higher quality and reliability, broadcasting is possible using a RTMP TCP app like OBS for desktop or GoCoder mobile, directly to streaming server without depending on web browser. RTMP stream is delivered to site users as HTML5 HLS.
Broadcaster browser streaming failure error, Retry message, permission errors, camera not available in list
– Make sure you are loading site over HTTPS (required to publish camera).
– Restart browser.
– Test with a different browser: Chrome, Firefox, Brave, Safari.
Try the Brave browser (Chrome privacy focused fork).
Intermittent connection issues associated with slow site or intermittent 503 web errors
Web hosting resources may be underpowered for site complexity and load.
-Try reducing site complexity (by removing plugins) and resource load per request.
-Upgrade to a higher plan from HTML5 WebRTC Relay Hosting .
Broadcasting stream not connecting or disconnects
Check if bitrate (video + audio) is within plan hosting limits. Trying to broadcast higher bitrate will result in automated stream rejection and short cooldown while all connection attempts are rejected.
See Client Upload (kbps) for your plan at HTML5 WebRTC Relay Hosting and configure lower.
Certain users only can’t stream
– Browser issue: Upgrade browser to latest version or try a different HTML5 browser like Brave browser . Browser must support latest WebRTC features and codecs. Older browsers and versions will not work (in Windows use Edge not Internet Explorer).
– Network Issue: Try a different network protocol: Broadcaster can download OBS for PC / GoCoder for mobile per instructions in Broadcast tab to broadcast with RTMP TCP instead of WebRTC UDP.
– Firewall Issue: If user has a firewall can temporary disable it to identify if that is the cause. Ports and protocols required depend on method of streaming, server and site configuration.
Visitors can’t connect or stream (without login)
Visitor only issues are most likely related to cache and site serving static content to their requests.
In WP Super Cache you can disable cache for site visitors that have cookies.
– Broadcaster needs to select correct input device (microphone). when broadcasting. Open broadcast tab to make changes to input devices/settings.
– Viewers need to use “Tap for Sound” button to enable audio.
Browsers require user interaction to allow automated video playback with sound. Sometimes playback is not permitted at all and application will show a “Tap to Play” button.
This is a browser feature / restriction.
– Try reloading stream or page. Depending on network conditions and browser, audio stream may sometimes be missing from WebRTC UDP broadcast. Try OBS/GoCoder streaming over RTMP TCP for increased reliability.
Webcam or Microphone Not Accessible
The web based applications use devices provided by system / browser to WebRTC . Try the WebRTC samples to check available devices. If system / browser does not make it accessible for WebRTC usage, the WebRTC based application can’t access it.
-Try a different browser.
-Try broadcasting with OBS with settings from Broadcast tab.
-If available, try the legacy Flash based applications.
-For more details about browser WebRTC support, check with hardware provider support.
General tests to identify nature of issue:
– Try live demos . If issues do not occur in the live demos, cause may be related to your installation. If same issues occur, these may be related to client browser / connection.
– Update browsers to latest versions. Try different browsers to identify if issue is specific to a certain browser and its plugins. Try opening a new tab/window in privacy mode without plugins if possible to test same browser without plugins.
– Try from a different computer / mobile device, if available, to identify if issue is specific to device.
– Try from a different ISP (in example access from a mobile device with WiFi turned off to use the mobile network) to identify if issue is specific to connection.
– Have a partner, friend from a different location test to identify if issue is specific to your testing environment.