Recent content by Radu

Just solved the error by removing the audio constraints: let audioConstraints ={ bitrate: 44100 } is now just: let audioConstraints=true

Getting the following error on Chrome (OSX Mojave) only using Chrome Version 70.0.3538.67 (Official Build) (64-bit). Anyone experienced anything related, is it a config issue from the server-side? Failed to parse SessionDescription: a=rtpmap:111 opus/48000/2 Invalid value: 2. Settings codecs...

Ah great! Thanks!

Is there any changelog for 5.1? The server.properties line: wss.port = 443 was not being taken into consideration: [root@phoner01 FlashphonerWebCallServer]# netstat -nlp | grep java tcp 0 0 0.0.0.0:843 0.0.0.0:* LISTEN 1282/java tcp 0...

My bad, found the update process. However switching from 5.0.33 to 5.1 does not work with the ./webcallserver update?

Ah great, we are already running a licensed WCS 5.0.2616 in production. Any instructions for a smooth upgrade? Also are there any potentially breaking changes between the two?

Hi Max, Some months ago you said it will be possible to do a pull-based rtmp live streaming in the browsers using websockets. Basically the client would say: "play rtmp://wowza-server/app/id.mp4" and the WCS5 would fetch the rtmp stream (h264+AAC), repackage it for MSE and play it in-browser...

PS Sound is fine when I add the "muted" attribute to the underlying video element from the webrtc stream but then audio from my end doesn't go through.

Hi Max, I have managed to set up a test case for you. There are issues playing the audio file (using AudioContext) after the WebRTC session is started. Please do the following steps (I'm using an iPhone 7 - iOS 11.1) - on an iphone 7 Safari go to https://www.imc-post.com/phoner/rtc.html -...

How can I force the audio bitrate to 44100?

I think it's related to this: https://github.com/Jam3/ios-safe-audio-context More on the issue: https://stackoverflow.com/questions/26336040/how-to-fix-changing-sample-rate-bug Will let you know if it fixes the problem.

I noticed a strange issue when doing WebRTC to RTMP from my iPhone. I believe this is more of an issue with Safari, but when I start streaming all other sounds played on the page are glitchy and sound is very bad. I added a simple mp3 audio tag to test and it plays well before I start the...

Sorry, please delete this, it's working well.

I'm trying to disable audio when publishing: constraints = { audio: false, video: { width: 320, height: 240 } } session.createStream({ name: streamName, display: @localVideo, cacheLocalResources: true, constraints: constraints...

Ah yes, it's the RTMP pulling that we need. Ok, great! We just purchased a monthly subscription to use the WebRTC -> RTMP (Wowza) publishing but we are looking forward for the pulling part. Will that be provided under the same product license?