using System.Text.Json;
using System.Text.Json.Serialization;
using RelayClient.Crypto;
using RelayShared.Rtc;
using RelayShared.Services;
namespace RelayClient.Services;
///
/// The bridge between the C# WebSocket pipe and the JavaScript WebRTC engine
/// running inside the HybridWebView (which is shown when a Voice channel is open).
///
/// Outbound (JS → C# → server): the WebView JS calls into C# via SendRtcSignal(json).
/// We deserialise to RtcSignalMessage, encrypt with the server's public key, wrap in
/// SocketRtcSignalMessage, and send through the WebSocket.
///
/// Inbound (server → C# → JS): the WebSocket fires EncryptedRtcSignalReceived. MainPage
/// hands it to HandleIncomingRtcSignalAsync, which decrypts with the user's private key
/// and calls back into JS via hybridWebView.InvokeJavaScriptAsync("testIndex", …).
///
/// JoinRtcChannel / LeaveRtcChannel just send WsAction control messages; presence tracking
/// happens server-side in RtcChannelPresenceService.
///
public sealed class RtcBridgeService
{
/// The currently-signed-in username. Stamped onto outgoing RTC signals.
private readonly string _username;
/// The shared WebSocket to RelayServer. Outbound RTC signals ride on this.
private readonly RelaySocketClient _socket;
/// The MAUI HybridWebView that hosts the JS WebRTC engine. We push JS calls into it.
private readonly HybridWebView _hybridWebView;
/// Lazy view into MainPage._currentChannelId so we always have the current voice channel.
private readonly Func _getCurrentChannelId;
/// Diagnostic logger that surfaces messages back to the WebView UI. Used for status/error reporting.
private readonly Action _sendRawToWebView;
/// Captures collaborators. MainPage constructs this once and never replaces it.
public RtcBridgeService(string username, RelaySocketClient socket, HybridWebView hybridWebView,
Func getCurrentChannelId, Action sendRawToWebView)
{
_username = username;
_socket = socket;
_hybridWebView = hybridWebView;
_getCurrentChannelId = getCurrentChannelId;
_sendRawToWebView = sendRawToWebView;
}
/// Sends RtcJoin for the currently-selected channel. Server-side, this triggers the Speak permission check and presence registration.
public Task JoinRtcChannel()
{
var channelId = _getCurrentChannelId();
if (string.IsNullOrWhiteSpace(channelId))
return Task.CompletedTask;
_socket.SendRtcJoinChannel(channelId);
return Task.CompletedTask;
}
/// Sends RtcLeave for the currently-selected channel. Clears server-side voice presence so peers stop seeing us.
public void LeaveRtcChannel()
{
var channelId = _getCurrentChannelId();
if (string.IsNullOrWhiteSpace(channelId))
return;
_socket.SendRtcLeaveChannel(channelId);
}
///
/// Called from JavaScript (via the HybridWebView bridge) when the WebRTC engine wants to
/// send an SDP offer/answer or ICE candidate to other peers. Parses the JSON, fills in
/// missing ChannelId/From, encrypts with the server's public key, ships as
/// SocketRtcSignalMessage. The server then forwards it (re-encrypted per-recipient) to
/// every other session in the same voice channel.
///
public void SendRtcSignal(string json)
{
if (string.IsNullOrWhiteSpace(_socket.ServerPublicKey))
{
_sendRawToWebView("SendRtcSignal failed: server public key not loaded.");
return;
}
RtcSignalMessage? rtcSignal;
try
{
rtcSignal = JsonSerializer.Deserialize(json);
}
catch (Exception ex)
{
_sendRawToWebView("SendRtcSignal failed to parse RTC signal: " + ex.Message);
return;
}
if (rtcSignal is null)
return;
rtcSignal.ChannelId ??= _getCurrentChannelId();
rtcSignal.From ??= _username;
// _sendRawToWebView($"RTC_SIGNAL file: {JsonSerializer.Serialize(rtcSignal)}");
if (string.IsNullOrWhiteSpace(rtcSignal.ChannelId))
{
_sendRawToWebView("SendRtcSignal failed: missing channel id.");
return;
}
var outgoingJson = JsonSerializer.Serialize(rtcSignal);
try
{
var encrypted = E2EeHelper.EncryptForRecipient(outgoingJson, _socket.ServerPublicKey);
var payload = new SocketRtcSignalMessage
{
Type = SignalType.EncryptedSignal,
SenderUsername = _username,
ChannelId = rtcSignal.ChannelId,
CipherText = encrypted.CipherText,
Nonce = encrypted.Nonce,
Tag = encrypted.Tag,
EncryptedKey = encrypted.EncryptedKey
};
_socket.SendJson(payload);
_sendRawToWebView($"SendRtcSignal sent: {rtcSignal.Type} -> {rtcSignal.To}");
}
catch (Exception ex)
{
_sendRawToWebView("SendRtcSignal failed: " + ex.Message);
}
}
/// JS bridge: returns the current voice-channel roster as JSON. Hits ServerAPI's REST endpoint, not the WebSocket.
public async Task GetRtcParticipants()
{
var channelId = _getCurrentChannelId();
if (string.IsNullOrWhiteSpace(channelId))
return "[]";
var participants = await ServerAPI.GetRtcParticipantsAsync(channelId);
return JsonSerializer.Serialize(participants ?? []);
}
///
/// MainPage hands incoming SocketRtcSignalMessage frames here. Filters out our own
/// frames, validates the channel scope, decrypts with the user's private key, parses to
/// RtcSignalMessage, then pushes into the JS RTC engine via SendRtcSignalToJsAsync.
///
public async Task HandleIncomingRtcSignalAsync(SocketRtcSignalMessage payload)
{
// _sendRawToWebView("HandleIncomingRtcSignal called");
var currentChannelId = _getCurrentChannelId();
if (payload.ChannelId != currentChannelId)
{
_sendRawToWebView("Channel id does not match");
return;
}
if (payload.SenderUsername == _username)
{
_sendRawToWebView("Received own message");
return;
}
string decryptedJson;
try
{
var privateKey = KeyStorage.LoadPrivateKey(_username);
decryptedJson = E2EeHelper.DecryptForRecipient(
new EncryptedPayload
{
CipherText = payload.CipherText,
Nonce = payload.Nonce,
Tag = payload.Tag,
EncryptedKey = payload.EncryptedKey
},
privateKey
);
}
catch (Exception ex)
{
_sendRawToWebView("RTC decrypt failed: " + ex.Message);
return;
}
RtcSignalMessage? rtcSignal;
try
{
rtcSignal = JsonSerializer.Deserialize(decryptedJson);
// _sendRawToWebView($"Received Encrypted Signal: [{rtcSignal.From}]: {rtcSignal.Offer}");
}
catch (Exception ex)
{
_sendRawToWebView("RTC signal parse failed: " + ex.Message);
return;
}
if (rtcSignal is null)
{
_sendRawToWebView("rtcSignal is null");
return;
}
if (!string.IsNullOrWhiteSpace(rtcSignal.To) &&
!string.Equals(rtcSignal.To, _username, StringComparison.OrdinalIgnoreCase))
{
_sendRawToWebView($"Ignoring RTC signal meant for {rtcSignal.To}");
return;
}
// _sendRawToWebView("Received encrypted RTC signal: " + decryptedJson);
await SendRtcSignalToJsAsync(rtcSignal);
}
///
/// Pushes the current username and channelId into JS globals (window.setUsername, window.setChannelId).
/// Called whenever the user switches voice channels OR the JS engine reports rtc_page_ready.
///
public Task PushRtcContextToJsAsync()
{
MainThread.BeginInvokeOnMainThread(async () =>
{
var usernameJson = JsonSerializer.Serialize(_username);
var channelIdJson = JsonSerializer.Serialize(_getCurrentChannelId());
await _hybridWebView.EvaluateJavaScriptAsync($"window.setUsername({usernameJson})");
await _hybridWebView.EvaluateJavaScriptAsync($"window.setChannelId({channelIdJson})");
});
return Task.CompletedTask;
}
///
/// Final hop: hands a decrypted RtcSignalMessage off to the JS engine via
/// hybridWebView.InvokeJavaScriptAsync("testIndex", …). SDP strings have their newlines
/// escaped as "(rn)" because the JSON marshalling otherwise breaks them.
///
private Task SendRtcSignalToJsAsync(RtcSignalMessage data)
{
if (data.Type == "rtc_offer" || data.Type == "rtc_answer")
{
data.Sdp = data.Sdp.Replace("\r\n", "(rn)");
}
MainThread.BeginInvokeOnMainThread(async () =>
{
try
{
// await _hybridWebView.InvokeJavaScriptAsync("testIndex", [JsonSerializer.Serialize(data)], [RtcJsType.Default.String]);
await _hybridWebView.InvokeJavaScriptAsync("testIndex", [data], [RtcJsType.Default.RtcSignalMessage]);
#region OldDebugger
// var jsArg = JsonSerializer.Serialize(data);
//
// await _hybridWebView.EvaluateJavaScriptAsync($@"
// try {{
// window.HybridWebView.SendRawMessage('C# eval entered');
//
// if (!window.RelaySocket) {{
// window.HybridWebView.SendRawMessage('window.RelaySocket missing');
// }} else if (typeof window.RelaySocket.receiveRtcSignal !== 'function') {{
// window.HybridWebView.SendRawMessage('RelaySocket.receiveRtcSignal missing');
// }} else {{
// window.HybridWebView.SendRawMessage('Calling RelaySocket.receiveRtcSignal');
// window.RelaySocket.receiveRtcSignal({jsArg});
// }}
// }} catch (err) {{
// window.HybridWebView.SendRawMessage('RTC JS dispatch failed: ' + err);
// }}
// ");
#endregion
}
catch (Exception ex)
{
_sendRawToWebView("SendRtcSignalToJsAsync failed: " + ex.Message);
}
});
return Task.CompletedTask;
}
}
[JsonSourceGenerationOptions(WriteIndented = false)]
[JsonSerializable(typeof(RtcDescription))]
[JsonSerializable(typeof(List))]
[JsonSerializable(typeof(RtcSignalMessage))]
[JsonSerializable(typeof(IceCandidate))]
[JsonSerializable(typeof(List))]
[JsonSerializable(typeof(string))]
internal partial class RtcJsType : JsonSerializerContext
{
}