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Perhaps there were some recording fixes or media packets parsing fixes between the latest build and the biuld you've used. The changes may not affect webm container directly.

Please look at those buugs description: https://bugs.webkit.org/show_bug.cgi?id=232381 https://bugs.webkit.org/show_bug.cgi?id=231505 The only 100% working solution is to use VP8 for publishing/playing. Another solution which AppRTC uses is to enable screen orientation support cvoExtension...

We checked your report. For futher reports, please use report.sh script to collect all files needed. Unfortunately, we do not see any playback tries for stream nat that you've marked as problem in report comments. Only publishing present in logs. Also, seems like this stream publishing is failed...

We've tested SIP calls with our WCS instance and with your WCS instance of course. The problem you describe is not reproducing in our tests: audio quality looks good for both sides (calling from Phone Min example to Bria softphone), no delays, losses or dashes. Voice seems to be clear (and music...

Добрый день. Мы сообщаем о текущем статусе здесь, в данном топике. Форум используется как публичный багтрекер. По тикету WCS-3379 промежуточные результаты пока такие: при выставлении targetSDK: 31 при сборке Android SDK и примеров публикация с камер работает, проблема воспроизводится только со...

Good day. Please provide more details: - what WCS build do you use? - how do you publish a stream which fails to play as RTMP? Please also reproduce the issue and collect a report as described here including a problem stream name, then send it using this form.

Good day. Have you updated WCS at this time? If yes, did rollback to previous version help? Please also clarify what WCS build do you use? Also please chaek if ffmpeg was updated. If there are no changes in software, seems like the problem is in stream published and recorded. If the problem is...

A client should pass authetication parameters in RTMP URL rmtp://mydomain.com:1935/live?username=user&password=pwd/streamKey To handle those parameters, you should use REST hook /connect. You backend server will receive a custom field with parameters passed, for example: POST...

We recommend you to update WCS at least to the latest build available in AWS Marketplace (5.2.944) or to the latest build from this page because there were a lot of fixes since build you use including RTMP streaming fixes.

We tried to test SIP calls with credentials you've provided. First, you SIP PBX does not support SIP TCP signaling. In this case, you should exclude unneeded codecs for SIP call to be established properly: allow_outside_codecs=false codecs_exclude_sip=mpeg4-generic,flv,mpv,h264,vp8,opus,speex16...

Firefox should be on other side: iOS Safari - Win Firefox for example In iOS, all the browsers are just Safari wrappers.

Good day. We've refactored Camera Manager example slightly to make third party filter library integration easier. Actually, you should implement 3 methods: initFilter, applyFilter, destroyFilter. See source code on GitHub and Camera Manager example description. By default, GPU image library is...

Good day. You should pass authetication parameters in RTMP URL and parse them on a separate backend server using REST hooks. Please read details here.

Good day. Please clarify what WCS build do you use? Also, please try to test with our demo server rtmp://demo.flashphoner.com:1935/live/streamKey. If the problem is not reproducing, please update your WCS to the latest build 5.2.1106 as described here.

The lines you've removed will be added again on server restart if your hosting provider exposes Amazon-like API because this is necessary settings if your instance is behind NAT. Please make sure your SIP PBX is in the same region. Please also check the call with our demo server as we...

client_mode=false rtc_ice_add_local_component=true These two parameters say that you are on Amazon EC2 instance. 1. Make sure your instance is running in the same region as region of your SIP PBX. 2. Make sure your instance has enough resources. For example, t3.micro is not a proper server for...

Yes, in our examples, storyboards are used. WebRTC library itself which is a basement for WCS iOS SDK is build in good old fashioned ObjectiveC by Google, so we provide a Swift wrapper. So you can build application interface with SwiftUI using a portions of code from our examples to publish or...

Добрый день. Проверили логи. По логам, происходило следующее: 1. В 17:04:08,024 публикация потока с камеры в микшер 5792gXDnIsKnlj5iwFv остановилась, т.к. в течение минут не было видеотрафика от клиента. В 17:04:15,127 поток был выведен из микшера. Поток экрана при этом оставался в микшере. 2. В...

Good day. Seems like this is a codec or bandwidth issue. Please check what audio codec is set in softphone parameters of the other side. Try to set up Opus codec if your SIP PBX supports that. You can also pass an additional parameters in INVITE SDP to manage bandwidth. Please read details here.

Это означает, что есть транскодинг или ресемплинг звука на Edge. Почитайте, пожалуйста, здесь по настройке проброса аудио через CDN. D Вашем случае на HLS edge нужно пробрасывать звук в кодеке AAC (mpeg4-generic). Также для HLS по умолчанию используется звук со следующими частотами дискретизации...