From dd32bb2ccbbb7d32ff458387eacb3463382b066c Mon Sep 17 00:00:00 2001
From: bangae1 <bangae2@gmail.com>
Date: Fri, 19 Sep 2025 20:25:02 +0900
Subject: [PATCH] first

---
 .gitignore                  |   2 +
 call.html                   |  18 ++++++
 src/main.js                 |  23 +++++++
 src/webrtc/audioRecorder.js |  43 ++++++++++++
 src/webrtc/uiController.js  |  52 +++++++++++++++
 src/webrtc/webrtcManager.js | 126 ++++++++++++++++++++++++++++++++++++
 6 files changed, 264 insertions(+)
 create mode 100644 .gitignore
 create mode 100644 call.html
 create mode 100644 src/main.js
 create mode 100644 src/webrtc/audioRecorder.js
 create mode 100644 src/webrtc/uiController.js
 create mode 100644 src/webrtc/webrtcManager.js
diff --git a/.gitignore b/.gitignore
new file mode 100644
index 0000000..29b636a
--- /dev/null
+++ b/.gitignore
@@ -0,0 +1,2 @@
+.idea
+*.iml
\ No newline at end of file
diff --git a/call.html b/call.html
new file mode 100644
index 0000000..ba74328
--- /dev/null
+++ b/call.html
@@ -0,0 +1,18 @@
+<!DOCTYPE html>
+<html lang="en">
+<head>
+    <meta charset="UTF-8">
+    <title>Title</title>
+</head>
+<body>
+<div id="webrtc-controls">
+    <input type="text" id="peer-id-input" placeholder="상대방 ID 입력">
+    <button id="call-btn">전화걸기</button>
+    <button id="hangup-btn">끊기</button>
+    <button id="record-btn">녹음시작</button>
+    <button id="stop-record-btn">녹음중지</button>
+    <audio id="audio-playback" controls></audio>
+</div>
+</body>
+<script type="module" src="/src/main.js"></script>
+</html>
\ No newline at end of file
diff --git a/src/main.js b/src/main.js
new file mode 100644
index 0000000..05b15c0
--- /dev/null
+++ b/src/main.js
@@ -0,0 +1,23 @@
+// main.js — 기존 웹 페이지에서 <script type="module"> 로 로드
+
+import { WebRTCManager } from '/client/src/webrtc/webrtcManager.js';
+import { AudioRecorder } from '/client/src/webrtc/audioRecorder.js';
+import { UIController } from '/client/src/webrtc/uiController.js';
+
+// DOM 요소 연결 (기존 웹에 있는 버튼/요소 선택)
+const callButton = document.getElementById('call-btn');
+const hangupButton = document.getElementById('hangup-btn');
+const recordButton = document.getElementById('record-btn');
+const stopRecordButton = document.getElementById('stop-record-btn');
+const audioPlayback = document.getElementById('audio-playback');
+
+// 인스턴스 생성
+const rtcManager = new WebRTCManager('http://localhost:3001');
+const recorder = new AudioRecorder();
+const ui = new UIController({ callButton, hangupButton, recordButton, stopRecordButton, audioPlayback });
+
+// UI 이벤트 바인딩
+ui.bindEvents(rtcManager, recorder);
+
+// 필요한 경우 초기화
+rtcManager.init();
\ No newline at end of file
diff --git a/src/webrtc/audioRecorder.js b/src/webrtc/audioRecorder.js
new file mode 100644
index 0000000..dac22a3
--- /dev/null
+++ b/src/webrtc/audioRecorder.js
@@ -0,0 +1,43 @@
+// audioRecorder.js
+
+export class AudioRecorder {
+    constructor() {
+        this.mediaRecorder = null;
+        this.audioChunks = [];
+        this.onRecordingComplete = null; // 외부 콜백
+    }
+
+    start(stream) {
+        this.audioChunks = [];
+        const options = { mimeType: 'audio/webm;codecs=opus' };
+        this.mediaRecorder = new MediaRecorder(stream, options);
+
+        this.mediaRecorder.ondataavailable = (event) => {
+            if (event.data.size > 0) {
+                this.audioChunks.push(event.data);
+            }
+        };
+
+        this.mediaRecorder.onstop = () => {
+            const audioBlob = new Blob(this.audioChunks, { type: 'audio/webm' });
+            const audioUrl = URL.createObjectURL(audioBlob);
+            if (this.onRecordingComplete) {
+                this.onRecordingComplete(audioUrl, audioBlob);
+            }
+        };
+
+        this.mediaRecorder.start();
+        console.log('⏺️ Recording started');
+    }
+
+    stop() {
+        if (this.mediaRecorder && this.mediaRecorder.state !== 'inactive') {
+            this.mediaRecorder.stop();
+            console.log('⏹️ Recording stopped');
+        }
+    }
+
+    isRecording() {
+        return this.mediaRecorder?.state === 'recording';
+    }
+}
\ No newline at end of file
diff --git a/src/webrtc/uiController.js b/src/webrtc/uiController.js
new file mode 100644
index 0000000..3fc5aa4
--- /dev/null
+++ b/src/webrtc/uiController.js
@@ -0,0 +1,52 @@
+// uiController.js (수정본)
+
+export class UIController {
+    constructor({ callButton, hangupButton, recordButton, stopRecordButton, audioPlayback, peerIdInput }) {
+        this.callButton = callButton;
+        this.hangupButton = hangupButton;
+        this.recordButton = recordButton;
+        this.stopRecordButton = stopRecordButton;
+        this.audioPlayback = audioPlayback;
+        this.peerIdInput = peerIdInput; // ✅ 상대방 ID 입력 필드
+    }
+
+    bindEvents(rtcManager, recorder) {
+        // 통화 시작
+        this.callButton.onclick = () => {
+            const peerId = this.peerIdInput.value.trim();
+            if (!peerId) return alert('상대방 ID를 입력하세요');
+            rtcManager.createOffer(peerId);
+        };
+
+        // 통화 종료
+        this.hangupButton.onclick = () => rtcManager.hangup();
+
+        // 녹음 시작
+        this.recordButton.onclick = () => {
+            const stream = rtcManager.remoteStream || rtcManager.localStream;
+            if (!stream) return alert('스트림이 없습니다');
+            recorder.start(stream);
+        };
+
+        // 녹음 중지
+        this.stopRecordButton.onclick = () => {
+            recorder.stop();
+        };
+
+        // 녹음 완료 시 재생
+        recorder.onRecordingComplete = (audioUrl, audioBlob) => {
+            this.audioPlayback.src = audioUrl;
+            this.audioPlayback.play();
+
+            // 필요시 서버 업로드
+            // uploadRecording(audioBlob);
+        };
+
+        // 원격 스트림 수신 시 자동 재생
+        rtcManager.onRemoteStream = (stream) => {
+            const audio = new Audio();
+            audio.srcObject = stream;
+            audio.play().catch(e => console.warn('자동 재생 실패:', e));
+        };
+    }
+}
\ No newline at end of file
diff --git a/src/webrtc/webrtcManager.js b/src/webrtc/webrtcManager.js
new file mode 100644
index 0000000..4015536
--- /dev/null
+++ b/src/webrtc/webrtcManager.js
@@ -0,0 +1,126 @@
+// webrtcManager.js (수정본 — STUN 없이 ICE만)
+
+export class WebRTCManager {
+    constructor(signalUrl) {
+        this.pc = null;
+        this.localStream = null;
+        this.remoteStream = null;
+        this.onRemoteStream = null;
+        this.signalUrl = signalUrl; // 시그널링 서버 주소
+        this.ws = null;
+        this.myId = null;
+        this.peerId = null; // 상대방 ID (수동 입력 또는 자동 할당)
+    }
+
+    async init() {
+        await this.requestMicPermission()
+        this.setupSignaling();
+    }
+
+    async requestMicPermission() {
+        try {
+            const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+            console.log('✅ 마이크 접근 성공');
+            stream.getTracks().forEach(track => track.stop()); // 임시로 열었다가 닫음
+            return true;
+        } catch (err) {
+            console.error('❌ 마이크 권한 거부 또는 장치 없음:', err.message);
+            alert('마이크 접근이 필요합니다. 브라우저 설정에서 마이크 권한을 허용해주세요.');
+            return false;
+        }
+    }
+
+    setupSignaling() {
+        this.ws = new WebSocket(this.signalUrl);
+
+        this.ws.onopen = () => {
+            console.log('🔌 시그널링 서버 연결됨');
+        };
+
+        this.ws.onmessage = (event) => {
+            const msg = JSON.parse(event.data);
+            if (msg.type === 'your-id') {
+                this.myId = msg.id;
+                console.log('내 ID:', this.myId);
+            } else if (msg.type === 'offer') {
+                this.setRemoteDescription(msg.sdp);
+                this.addIceCandidateQueue.forEach(c => this.pc.addIceCandidate(c));
+                this.addIceCandidateQueue = [];
+            } else if (msg.type === 'answer') {
+                this.setRemoteDescription(msg.sdp);
+            } else if (msg.type === 'candidate') {
+                this.addIceCandidate(msg.candidate);
+            }
+        };
+    }
+
+    createPeerConnection() {
+        // ✅ STUN 서버 없이 빈 설정 — 인트라넷 내부 IP로 직접 연결
+        this.pc = new RTCPeerConnection({});
+
+        this.localStream.getTracks().forEach(track => {
+            this.pc.addTrack(track, this.localStream);
+        });
+
+        this.pc.ontrack = (event) => {
+            this.remoteStream = event.streams[0];
+            if (this.onRemoteStream) this.onRemoteStream(this.remoteStream);
+        };
+
+        // ICE Candidate → 시그널링으로 전송
+        this.pc.onicecandidate = (event) => {
+            if (event.candidate) {
+                console.log('📤 ICE Candidate:', event.candidate.candidate);
+                this.ws.send(JSON.stringify({
+                    target: this.peerId,
+                    type: 'candidate',
+                    candidate: event.candidate
+                }));
+            }
+        };
+    }
+
+    async createOffer(targetId) {
+        this.peerId = targetId;
+        this.createPeerConnection();
+        const offer = await this.pc.createOffer();
+        await this.pc.setLocalDescription(offer);
+
+        this.ws.send(JSON.stringify({
+            target: targetId,
+            type: 'offer',
+            sdp: offer
+        }));
+        console.log('📤 Offer 전송 완료');
+    }
+
+    async setRemoteDescription(sdp) {
+        await this.pc.setRemoteDescription(new RTCSessionDescription(sdp));
+        if (sdp.type === 'offer') {
+            const answer = await this.pc.createAnswer();
+            await this.pc.setLocalDescription(answer);
+            this.ws.send(JSON.stringify({
+                target: this.peerId || 'unknown',
+                type: 'answer',
+                sdp: answer
+            }));
+        }
+    }
+
+    async addIceCandidate(candidate) {
+        if (this.pc && this.pc.remoteDescription) {
+            await this.pc.addIceCandidate(new RTCIceCandidate(candidate));
+        } else {
+            // SDP 설정 전이라면 큐에 저장
+            this.addIceCandidateQueue = this.addIceCandidateQueue || [];
+            this.addIceCandidateQueue.push(candidate);
+        }
+    }
+
+    hangup() {
+        this.pc?.close();
+        this.localStream?.getTracks().forEach(t => t.stop());
+        this.ws?.close();
+        console.log('📞 통화 종료');
+    }
+}
\ No newline at end of file
