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Good day. You can publish a stream from VOD using REST API. Please read details here POST /rest-api/vod/startup HTTP/1.1 HOST: localhost:8081 content-type: application/json content-length: 83 { "uri":"vod-live://sample.mp4", "localStreamName": "test" }

Добрый день. Протестировали автоматическую ретрансляцию потоков на RTMP сервер, в сборке 5.2.1110 добавили возможность указать параметры аутентификации rtmp_push_auto_start_url=rtmp://user:password@rtmp.server.com:1935/live На текущий момент этот функционал может использоваться для...

We checked your server. Seems like you've tried to update WCS to the latest build, but you've not stopped previous WCS process. Also, your server have 251 Gb RAM, and you've trying to launch WCS with Java Heap 220 Gb and Wowza with Java heap 200 Gb simultaneously. The sum 220 + 200 is more than...

Добрый день. Скорее всего, проблема в том, что Вы пытаетесь добавить поток без видео: В этом случае фоновая картинка отрисовываться не будет, т.к. поток будет добавлен в микшер без видеосоставляющей. Для таких потоков можно управлять только цветом фона прямоугольника. Кроме того, обратите...

We also have tested the latest build 5.2.1109, and the vulnerability cannot be exploited, please see test results.

Good day. According to CVE description https://cve.mitre.org/cgi-bin/cvename.cgi?name=CVE-2021-44228 In case of WCS, an external attacker cannot control log messages or log message parameters because the only log configuration file is log4j.properties file which is placed in...

Please provide SSH access to your server and WCS dashboard access using this form.

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.