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This issue has been fixed in macOS Big Sur 11.1 (Safari 14.0.2), iOS 14.3 and iPadOS 14.3.

Wrong thread?

Looks like they're on it: Safari Technology Preview 116 is generally okay! Only right after starting the stream there is still a brief moment of this distortion, after that it remains stable.

That's exactly the issue I have, thanks for pointing me to this bug report. The test case provided there is spot on: https://jsfiddle.net/juberti/bnmfvjo0/

In recent releases of Apple software (macOS Big Sur using Safari 14.0.1, iOS 14.2 and iPadOS 14.2) the audio of my RTSP stream starts heavily distorted when using WebRTC (Opus audio). After some time audio becomes clean but it can go back to the distorted state later on. I already tried changing...

Thank you for integrating this workaround into Web SDK!

I updated to 5.2.820 and it works, let's hope the fix will stick! Thank you!

Link sent!

I updated from 5.2.784 to 5.2.798 and now the same RTSP stream fails, is this again an SDP parsing issue? The SDP has not changed. o=- 18467 41 IN IP4 0.0.0.0 c=IN IP4 0.0.0.0 a=control:* a=range:npt=0- t=0 0 m=video 0 RTP/AVP 96 a=rtpmap:96 H264/90000 a=fmtp:96 profile-level-id=0...

The freezes were caused by wrong buffer management in the encoder. The I-frames were sometimes too big and therefore arrived too late (buffer underrun situation), causing the browser to freeze the video for the duration of the whole GOP. It was fixed by new firmware from the encoder manufacturer...

Thanks for pointing me to that thread! In comment #16 and #25 a workaround is described and demoed: So I went ahead to implement this workaround in flashphoner.js around line 14507: //create offer and set local sdp connection.createOffer(constraints).then(function (offer) { // workaround...

Link sent!

It makes no difference whether or opus_formats=stereo=1 exists in flashphoner.properties. Safari and Chrome are always mono and Firefox is always stereo.

I updated to 5.2.784 and set up the custom SDP files for MSE and HLS as per your postings. I can confirm audio is now stereo and the quality is great when streaming to MSE and HLS from my RTSP source. However, over WebRTC (using Opus) audio is still mono. How do I set the Opus encoder to output...

Today I updated to 5.2.784 and I can confirm that HLS from an RTSP source now works as expected. Thank you!

Hi, My RTSP stream plays fine in the embedded player example page (WebRTC). Now I am trying to access this RTSP stream as HLS. From the HLS player examples I understand that I should be able to get the M3U8 playlist at https://<WCS_IP>:8445/<URL-encoded-RTSP-URL>/<URL-encoded-RTSP-URL>.m3u8...

My source is an RTSP stream with stereo AAC audio. I want to achieve high quality stereo audio encoding in the direction of server to browser. The AAC and Opus bitrates are configurable in flashphoner.properties, but it's still mono. How can I set the Opus and AAC encoders to stereo? And it...

I think the underlying problem is with keepalive. It seems Safari is trying to keep the connection alive in a way that is confusing Web Call Server. Using Chrome the connection is kept alive correctly but using Safari the connection is either dropped (in case WCS is running on Java 8) or hangs...

I resolved this by downgrading from Java 11 (jre-default on Debian 10) to Java 8: https://linuxize.com/post/install-java-on-debian-10/#installing-openjdk-8 https://linuxize.com/post/install-java-on-debian-10/#set-the-default-version

My Safari browser (version 13.1.1 (15609.2.9.1.2) on macOS 10.15.5) sends HTTPS traffic that causes permanently high CPU usage on the server, until I restart WCS. It looks like it happens when the HTTPS connections are closed after timeout. I just go to https://<WCS_IP>:8444/ and around 30...