diff --git a/app/providers/CallProvider/CallProvider.tsx b/app/providers/CallProvider/CallProvider.tsx
index b6c61d2..f5781fb 100644
--- a/app/providers/CallProvider/CallProvider.tsx
+++ b/app/providers/CallProvider/CallProvider.tsx
@@ -92,10 +92,27 @@ export function CallProvider(props : CallProviderProps) {
              * Другая сторона отправила нам ICE кандидата для установления WebRTC соединения
              */
             const candidate = JSON.parse(packet.getSdpOrCandidate());
+            console.info(candidate);
             await peerConnectionRef.current?.addIceCandidate(new RTCIceCandidate(candidate));
             info("Received WebRTC ICE candidate and added to peer connection");
             return;
         }
+        if(signalType == WebRTCSignalType.OFFER){
+            /**
+             * SFU сервер отправил нам оффер, например при renegotiation, нам нужно его принять и 
+             * отправить ответ (ANSWER)
+             */
+             const sdp = JSON.parse(packet.getSdpOrCandidate());
+             await peerConnectionRef.current?.setRemoteDescription(new RTCSessionDescription(sdp));
+             let answer = await peerConnectionRef.current?.createAnswer();
+             await peerConnectionRef.current?.setLocalDescription(answer);
+             let answerSignal = new PacketWebRTC();
+             answerSignal.setSignalType(WebRTCSignalType.ANSWER);
+             answerSignal.setSdpOrCandidate(JSON.stringify(answer));
+             send(answerSignal);
+             info("Received WebRTC offer, set remote description and sent answer");
+             return;
+        }
     }, [activeCall, sessionKeys, callState]);
 
     usePacket(26, async (packet: PacketSignalPeer) => {
@@ -201,6 +218,34 @@ export function CallProvider(props : CallProviderProps) {
                     { urls: 'stun:stun.l.google.com:19302' }
                 ]
             });
+            /**
+             * Подписываемся на ICE кандидат
+             */
+            peerConnectionRef.current.onicecandidate = (event) => {
+                if(event.candidate){
+                    let candidateSignal = new PacketWebRTC();
+                    candidateSignal.setSignalType(WebRTCSignalType.ICE_CANDIDATE);
+                    candidateSignal.setSdpOrCandidate(JSON.stringify(event.candidate));
+                    send(candidateSignal);
+                }
+            }
+            /**
+             * Соединение установлено, можно начинать звонок, переходим в активное состояние звонка
+             */
+            peerConnectionRef.current.onconnectionstatechange = () => {
+                console.info("Peer connection state changed: " + peerConnectionRef.current?.connectionState);
+                if(peerConnectionRef.current?.connectionState == "connected"){
+                    setCallState(CallState.ACTIVE);
+                    info("WebRTC connection established, call is active");
+                }
+            }
+
+            peerConnectionRef.current.ontrack = (event) => {
+                /**
+                 * При получении медиа-трека с другой стороны
+                 */
+                console.info("TRACK RECV!!!!!");
+            }
             /**
              * Запрашиваем Аудио поток с микрофона и добавляем его в PeerConnection, чтобы другая сторона могла его получить и воспроизвести, 
              * когда мы установим WebRTC соединение