diff --git a/submodules/TgVoipWebrtc/Impl/Controller.mm b/submodules/TgVoipWebrtc/Impl/Controller.mm
index 808c4aa7ac..e042e528ad 100644
--- a/submodules/TgVoipWebrtc/Impl/Controller.mm
+++ b/submodules/TgVoipWebrtc/Impl/Controller.mm
@@ -20,9 +20,9 @@ Controller::Controller(bool is_outgoing, const EncryptionKey& encryption_key, si
 : thread(rtc::Thread::Create())
 , connector(std::make_unique<Connector>(std::make_unique<Layer92>(encryption_key, is_outgoing)))
 , state(State::Starting)
-, is_outgoing(is_outgoing)
 , last_recv_time(rtc::TimeMillis())
 , last_send_time(rtc::TimeMillis())
+, is_outgoing(is_outgoing)
 , init_timeout(init_timeout * 1000)
 , reconnect_timeout(reconnect_timeout * 1000)
 , local_datasaving(false)
@@ -37,9 +37,6 @@ Controller::Controller(bool is_outgoing, const EncryptionKey& encryption_key, si
 Controller::~Controller() {
     thread->Invoke<void>(RTC_FROM_HERE, [this]() {
         media = nullptr;
-#ifdef TGVOIP_PREPROCESSED_OUTPUT
-        preproc = nullptr;
-#endif
         connector = nullptr;
     });
 }
@@ -111,11 +108,14 @@ void Controller::NewMessage(const message::Base& msg) {
             }
         });
         if (!webrtc::RtpUtility::RtpHeaderParser(msg_rtp.data.data(), msg_rtp.data.size()).RTCP()) {
+            //printf("rtp received size %d\n", (int)(msg_rtp.data.size()));
             last_recv_time = rtc::TimeMillis();
             if (state == State::Reconnecting) {
                 state = State::Established;
                 SignalNewState(state);
             }
+        } else {
+            //printf("rtcp received size %d\n", (int)(msg_rtp.data.size()));
         }
     } else if (msg.ID == message::tBufferOverflow ||
                msg.ID == message::tPacketIncorrect ||
diff --git a/submodules/TgVoipWebrtc/Impl/Endpoint.cpp b/submodules/TgVoipWebrtc/Impl/Endpoint.mm
similarity index 99%
rename from submodules/TgVoipWebrtc/Impl/Endpoint.cpp
rename to submodules/TgVoipWebrtc/Impl/Endpoint.mm
index 4e26c0d941..7ef75142c0 100644
--- a/submodules/TgVoipWebrtc/Impl/Endpoint.cpp
+++ b/submodules/TgVoipWebrtc/Impl/Endpoint.mm
@@ -91,6 +91,7 @@ void EndpointRelayObfuscatedTcp::Close(rtc::AsyncPacketSocket *, int) {
 void EndpointRelayObfuscatedTcp::RecvPacket(rtc::AsyncPacketSocket *socket, const char *data, size_t packet_len,
         const rtc::SocketAddress& remote_addr, const int64_t& packet_time_us) {
     EndpointBase::RecvPacket(socket, data, packet_len, remote_addr, packet_time_us);
+    
     do {
         if (in_remains > in_buffer->Length())
             break;
diff --git a/submodules/TgVoipWebrtc/Impl/MediaEngineWebrtc.mm b/submodules/TgVoipWebrtc/Impl/MediaEngineWebrtc.mm
index 1229fc91b0..d72078c985 100644
--- a/submodules/TgVoipWebrtc/Impl/MediaEngineWebrtc.mm
+++ b/submodules/TgVoipWebrtc/Impl/MediaEngineWebrtc.mm
@@ -177,9 +177,9 @@ MediaEngineWebrtc::MediaEngineWebrtc(bool outgoing, bool send, bool recv)
                 rtc::scoped_refptr<webrtc::ObjCVideoTrackSource> objCVideoTrackSource(new rtc::RefCountedObject<webrtc::ObjCVideoTrackSource>());
                 _nativeVideoSource = webrtc::VideoTrackSourceProxy::Create(signaling_thread.get(), worker_thread.get(), objCVideoTrackSource);
                 
-                codec.SetParam(cricket::kCodecParamMinBitrate, 300000);
-                codec.SetParam(cricket::kCodecParamStartBitrate, 300000);
-                codec.SetParam(cricket::kCodecParamMaxBitrate, 600000);
+                codec.SetParam(cricket::kCodecParamMinBitrate, 32);
+                codec.SetParam(cricket::kCodecParamStartBitrate, 300);
+                codec.SetParam(cricket::kCodecParamMaxBitrate, 1000);
                 
 #if TARGET_IPHONE_SIMULATOR
 #else
@@ -242,9 +242,9 @@ MediaEngineWebrtc::MediaEngineWebrtc(bool outgoing, bool send, bool recv)
                 //send_parameters.options.auto_gain_control = params.auto_gain_control;
                 //send_parameters.options.highpass_filter = false;
                 //send_parameters.options.typing_detection = false;
-                send_parameters.max_bandwidth_bps = 300000;
-                send_parameters.rtcp.reduced_size = true;
-                send_parameters.rtcp.remote_estimate = true;
+                //send_parameters.max_bandwidth_bps = 800000;
+                //send_parameters.rtcp.reduced_size = true;
+                //send_parameters.rtcp.remote_estimate = true;
                 video_channel->SetSendParameters(send_parameters);
                 
                 video_channel->SetVideoSend(ssrc_send_video, NULL, _nativeVideoSource.get());
@@ -270,15 +270,15 @@ MediaEngineWebrtc::MediaEngineWebrtc(bool outgoing, bool send, bool recv)
     if (recv) {
         for (auto codec : videoCodecs) {
             if (codec.id == 96 && codec.name == cricket::kH264CodecName) {
-                codec.SetParam(cricket::kCodecParamMinBitrate, 300000);
-                codec.SetParam(cricket::kCodecParamStartBitrate, 300000);
-                codec.SetParam(cricket::kCodecParamMaxBitrate, 600000);
+                codec.SetParam(cricket::kCodecParamMinBitrate, 32);
+                codec.SetParam(cricket::kCodecParamStartBitrate, 300);
+                codec.SetParam(cricket::kCodecParamMaxBitrate, 1000);
                 
                 cricket::VideoRecvParameters recv_parameters;
                 recv_parameters.codecs.emplace_back(codec);
                 recv_parameters.extensions.emplace_back(webrtc::RtpExtension::kTransportSequenceNumberUri, extension_sequence);
-                recv_parameters.rtcp.reduced_size = true;
-                recv_parameters.rtcp.remote_estimate = true;
+                //recv_parameters.rtcp.reduced_size = true;
+                //recv_parameters.rtcp.remote_estimate = true;
                 video_channel->AddRecvStream(cricket::StreamParams::CreateLegacy(ssrc_recv_video));
                 video_channel->SetRecvParameters(recv_parameters);
                 
@@ -324,11 +324,11 @@ void MediaEngineWebrtc::SetNetworkParams(const MediaEngineWebrtc::NetworkParams&
 //        send_parameters.max_bandwidth_bps = 16000;
     send_parameters.rtcp.reduced_size = true;
     send_parameters.rtcp.remote_estimate = true;
-    voice_channel->SetSendParameters(send_parameters);
+    //voice_channel->SetSendParameters(send_parameters);
 }
 
 void MediaEngineWebrtc::SetMute(bool mute) {
-    voice_channel->SetAudioSend(ssrc_send, !mute, nullptr, &audio_source);
+    //voice_channel->SetAudioSend(ssrc_send, !mute, nullptr, &audio_source);
 }
 
 void MediaEngineWebrtc::AttachVideoView(VideoMetalView *videoView) {
diff --git a/submodules/TgVoipWebrtc/Impl/TgVoip.mm b/submodules/TgVoipWebrtc/Impl/TgVoip.mm
index 0b4e5d87eb..f1046f705c 100644
--- a/submodules/TgVoipWebrtc/Impl/TgVoip.mm
+++ b/submodules/TgVoipWebrtc/Impl/TgVoip.mm
@@ -151,6 +151,9 @@ public:
             }
             controller_->AddEndpoint(addr, endpoint.peerTag, type);
         }
+        /*rtc::SocketAddress addr("192.168.8.118", 7325);
+        unsigned char peerTag[16];
+        controller_->AddEndpoint(addr, peerTag, Controller::EndpointType::P2P);*/
 
         setNetworkType(initialNetworkType);