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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...

Please check media ports configuration as we mentioned above. Also check server log after container start: are there any port binding errors? Read also this article about WebRTC ports forwarding to docker instance.

First, please create your own topic for futher questions. You can capture RTSP streams and record them using REST API Stream capture from the IP camera management by REST API Stream recording on demand

Good day. Please read the following docs: Overview iOS SDK Swift framework integration iOS SDK Swift framework examples

Good day. If you've extended media ports range, and using CDN (or WebRTC pull/push), you should also set a new range for WCS agent ports which are used for that purpose, for example: media_port_from=20000 media_port_to=40000 wcs_agent_port_from=44001 wcs_agent_port_to=55000 See also the...

We've checked the player by link: 1. Publish a stream using Two Way Streaming example 2. Play this stream in player Seems like player contains the recommended modifications and audio is playing correctly:

Good day. If you wat to just test the camera, then publish a stream, please look at this camera test example: Testing microphones and cameras. The example shows how to stop test by clicking Stop test button, you can stop camera test then start streaming in this case.

Yes. Most of European locales force decimal point to be comma, so you should set locale to en_US.