CloudWebRTC
CloudWebRTC
You are right, these two functions are related and should need to be modified at the same time.
I think this is not in the scope of this plugin. The current sip protocol is only used for invite sessions. It is a one-to-one model. To implement a conference...
@mroshini `grammar_parser.dart` is generated from `grammar.peg` This file cannot be modified manually because the amount of code is too large, unless we rewrite a sip header parser, maybe we can...
> Please support null safety @a42279395 You can fund anyone to complete this task because this upgrade has a lot of work, Maybe @mroshini is already doing it, you can...
Did you use the audio-only mode to call on flutter web?
Excluding other errors, I think the problem is caused by audio stream rendering on the web. In the case of rendering a video stream, an `html.VideoElement` will be created in...
This is a good proposal. I will upgrade this repo later(I currently cannot maintain open source projects full-time). Of course, I will be happy to accept a PR and review...
This plug-in currently uses webrtc as the media stack. It supports opus, PCMU, and PCMA encoding. As you mentioned, G729, GSM, Speex, and AMR encoding require custom development of libwebrtc,...
Just wait for this PR to land https://github.com/flutter-webrtc/flutter-webrtc/pull/290
The current version should work on webrtc plan-b, but it needs some testing. It fully supports unified-plan and still needs to wait https://github.com/flutter-webrtc/flutter-webrtc/pull/610 Landing