I am using socketIO and nodejs express server how do I leave the video conference and end the call
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I am working on an application where host needs to mute other participants. I have tried using remotestream.muteAudio() api. But this doesn't mute the audio at source. Which means with this API, if host mutes Participant A it is muted only for the host. But participant B can still hear Participant A.
I went through the API documentation and couldn't find the solution.
https://docs.agora.io/en/faq/API%20Reference/web/interfaces/agorartc.stream.html#muteaudio
Can this be achieved by any ways?
Hi there you need to use the Agora RTM SDK to do that.
You can create an RTM channel with the same name as the RTC channel. Alternatively, You can use P2P messaging(Also present in RTM) if that suits your use case better.
Then the host can send a message to the concerned user to mute. On receiving this message, the user can call the muteAudio on their side, thereby muting the audio at source. All of this would be done programmatically.
I want to create a video chat with WebRTC, but i have no idea about this. I need my own WebRTC server to establish a video call from a PC browser to another browser on PC or on android device, how should I do?
Luckily we live in 2018 where most of the stuff already implemented. There are bunch of WebRTC Video Chat providers that provide API, SDK and docs for integration.
I used ConnectyCube in many of my applications. They provide WebRTC Video Calling functionality for iOS, Android and Web(JS).
iOS WebRTC (video chat) guide https://developers.connectycube.com/ios/videocalling
Android WebRTC (video chat) guide https://developers.connectycube.com/android/videocalling
Javascript WebRTC (video chat) guide https://developers.connectycube.com/js/videocalling
WebRTC features supported:
1-1 video chat
Group video chat
WebRTC based
VP8/H264 video codecs supported
Mute/Unmute audio/video stream
Switch video input devices (cameras)
Video recording
The whole list of supported features
Highly recommend to try something like this and do not waste time on implementing everything from scratch by yourself.
I have been reading a lot of SIP/WebRTC documentation and it is not clear to me whether a video call between an intercom device (SIP) and a web client (WebRTC) can be established.
I have researched and played with various VoIP server/frameworks like OverSIP, Asterisk, Kamailio, JSSIP, PJSIP, FreeSwitch, dratchio ... and all I have got is to receive the SIP device call in the server.
The basic flow for the PoC would be as follows:
SIP device (video door entry) initiates call to the server
Server receive the call
The client can contest the videocall using a web page.
The complete flow would be as follows (always the same flow):
SIP device (video door entry) initiates call to the server
Server determine the destination client
Server make a temporal webpage to contest the videocall
Server sends the URL using a notification to client's mobile device (or clients mobiles, in plural)
The mobile receive the notification and open the web page showing the video call
The customer can answer the video call from the web
It's possible? Is there an example or tutorial for help?
Thanks!
I am having a requirement to broadcast live audio on my website. Scenario is
1) one user will talk/sing in my application and
2) his followers will have to listen that live audio instantly in the same application.
Can you please suggest me any 3rd party libraries for this?
Note: I am developing my web application in ASP.Net MVC5.
Thanks in advance.
Towards that goal, if you plan on using WebRTC, you'll need to add media servers that route media around.
Look at Kurento, Jitsi or Janus for open source alternatives - especially if you plan on adding video support to it.
You can also try using Asterisk or FreeSWITCH - these are telephony based solutions but can be re-purposed to use WebRTC and offer a conference call like experience where you mute everyone and have a single speaker "broadcast" his audio to all the rest.
How to disable video streaming while video streaming is already invoked? Like in Facebook Messenger, it can turn video call to voice call only.
At the moment, you'll have to tear down and recreate the whole connection with audio only, as Kurento does not support renegotiation. We are working on it, though!