Improve permission handling in call flow
Improve permission handling by asking the user to give mic and camera permissions before sending `CallBegin` or `CallAccept` for the caller and receiver respectively.
Followed the flow described here: https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API/Connectivity#session_descriptions
CallProvider:
- Change functions order to place listeners under the function that sends the corresponding WebSocket message.
- Replace `Default` CallStatus with `Loading` for when asking user permissions before sending the `CallBegin`/`CallAccept` message.
- Remove `localStream` and `remoteStream` from `CallContext`. They are now available only in `WebRtcContext`.
- Replace `setAudioStatus` and `setVideoStatus` with `setIsAudioOn` and `setIsVideoOn`. A `useEffect` is now used to disable the tracks when the audio/video status changes.
WebRtcProvider:
- Move WebRTC connection close logic to WebRtcProvider
- Remove `webRtcConnection` from `WebRtcContext`. `WebRtcProvider` is now in charge of setting everything related to the WebRTC Connection.
UI:
- Add `CallPermissionDenied` page for when permissions are denied.
- Rework `CallPending` to display `Loading...` when waiting for user permissions
Change-Id: I48153577cca4c73cdb9b81d2fa78cfdfe2e06d69
diff --git a/client/src/components/ConversationView.tsx b/client/src/components/ConversationView.tsx
index 064efcf..e412cca 100644
--- a/client/src/components/ConversationView.tsx
+++ b/client/src/components/ConversationView.tsx
@@ -43,9 +43,8 @@
const ConversationHeader = () => {
const { account } = useAuthContext();
- const { conversation } = useContext(ConversationContext);
+ const { conversation, conversationId } = useContext(ConversationContext);
const { t } = useTranslation();
- const { conversationId } = useContext(ConversationContext);
const members = conversation.getMembers();
const adminTitle = conversation.infos.title as string;