Hi,
@cloudwebrtc is Safari (mobile/desktop) supported?
And should we use adapter.js for web? (https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API/adapter.js)
Thanks.
@cloudwebrtc is Safari (mobile/desktop) supported?
And should we use adapter.js for web? (https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API/adapter.js)
Thanks.
Hi everyone, can someone help? I got this error while debug, /Users/dauken/FlutterDev/flutter/.pub-cache/hosted/pub.dartlang.org/flutter_webrtc-0.8.1/ios/Classes/FlutterRTCMediaStream.m:3:9: fatal error: 'WebRTC/WebRTC.h' file not found
Anyone has screen sharing functionality on flutter webRTC mobile? Please share the code if you have.
@hardikdabhi2 https://github.com/flutterjanus/flutter_janus_client you can have that with janus server
@shivanshtalwar0 because i need to implement conference call.
@shivanshtalwar0 can you please share me server?
let me check sir
will it work on mac system sir?
Sorry for too many questions.
Hello all. I'm using the callkeep plugin to handle some calls. I've run into an issue in which the phone app seems to start with the speakers enabled, whether the audio actually comes out from them or not (see image) .
Has anyone dealt with this before?
1 reply
I am facing a problem in getting the remote stream on devices when they are on different networks. Currently, I am testing on the google's STUN server
If anyone using janus webrtc server and is looking for best typescript wrapper library then you guys can play and use https://github.com/flutterjanus/JanusJs
Hi I'm building a flutter app for our webrtc project and I've got weird but interesting issue I don't understand. For some reason native webrtc cannot receive my audio streams... the same audio works fine in flutter web (I can see data received in webrtc-stats) but on macOS or iOS under the ssrc for my audio it looks like nothing is received. To make it slightly more interesting I can receive video normally. Any idea what could be wrong?
Hello. Not sure if this might be the right place to ask, but seeing as the callkeep plugin points to webrtc, I thought someone could help. On Android, I have no issues as the OS's call screen shows correctly and I can handle every event with very little trouble. However, on iOS, while the OS's incoming call notification shows correctly and I can answer it, the call screen seems to show for just a split second before going back to my app.
For a workaround, I have a custom screen to handle the call events, but that seems to have another set of issues:
- When I try to mute/unmute the microphone through callkeep's events, it doesn't seem to have much of an effect, even when the event seems to go through with no errors showing in the console.
- When using the hold/unhold event, the former works just fine as no audio is received or sent; however, while the latter seems to correctly call the event (like the muting mentioned above), I still cannot receive or send any audio.
Does anyone know if there is a way to make use of iOS's calling screen show like Android's does? And/or has anyone dealt with the couple of issues mentioned above?
@hatembz you can use janus sip plugin to achieve that , here's the demo:- https://janus.conf.meetecho.com/siptest.html
Hi again. I've managed to handle the previously mentioned issue after some further digging. However, I'm now dealing with something different. Using
janus_client, I'm handling SIP calls to my app. Through the use of a background service, I can keep the app listening when the app is in the background and when the phone is locked. On Android, the handling works just fine as I can still receive and answer calls with mostly no issues. On iOS, while the app is still able to receive calls and answer them, I'm able to send audio to the caller, but I'm unable to receive audio from the caller. Does anyone know what could be a possible cause for this?
I have been trying to get something working on windows but have been getting an exception on createLocalMediaStream method
is there a way to make the app always registered to the sip even if the app is terminated
Hello all. Using
callkeep and janus_client, we're trying to do calls over SIP. We've just noticed that trying to call numbers such as 911 takes us to the phone app of the device instead of trying to directly call. I've read up something about the way emergency numbers that are handled in such a way by the OS itself, as well as a list of numbers labeled as emergency as seen here on pages 19 and 20. However, there is a mention of the number 08, and yet our app doesn't quite behave the same on different devices: on a Moto G 9 Plus, dialing this number (or any number starting with 08) seems to work just fine with callkeep, as the app at least tries to start the call; on Samsung devices, it takes the user to the device's phone app (like I mentioned at the start). Has anyone dealt with something similar to this?
@cloudwebrtc gonna provide few PR's soon. In particular frame capture and recording. But my audio recording implementation is very hacky, includes usage of private headers
Hi, I'm experimenting with flutter web rtc and after long days of trial and error the best I achieved is a healthy connection but no video feed from remote: would someone please have a look https://github.com/bmd007/wonderland/blob/master/dance_partner_finder/lib/bloc/video_chat/video_chat_bloc.dart ?
this link relates to a block in a bigger app. But this bloc is all responsible for the web rtc part. The related widget is https://github.com/bmd007/wonderland/blob/master/dance_partner_finder/lib/video_chat_widget.dart
When I run the code locally, I don't get any error. But I only see the local stream. The remote stream doesn't show up.