diff --git a/submodules/TgVoipWebrtc/tgcalls b/submodules/TgVoipWebrtc/tgcalls
index 8ec8837f16..37b8dcc71a 160000
--- a/submodules/TgVoipWebrtc/tgcalls
+++ b/submodules/TgVoipWebrtc/tgcalls
@@ -1 +1 @@
-Subproject commit 8ec8837f169a3594ad457c8ee47f9ecb4d576275
+Subproject commit 37b8dcc71a77fb93fa23edef306b27269ee9da94
diff --git a/third-party/webrtc/BUILD b/third-party/webrtc/BUILD
index 2fc4db27e5..4ea2f56d5d 100644
--- a/third-party/webrtc/BUILD
+++ b/third-party/webrtc/BUILD
@@ -728,7 +728,6 @@ webrtc_sources = [
     "pc/sdp_utils.cc",
     "pc/session_description.cc",
     "pc/simulcast_description.cc",
-    "pc/srtp_filter.cc",
     "pc/srtp_session.cc",
     "pc/srtp_transport.cc",
     "pc/track_media_info_map.cc",
@@ -912,7 +911,6 @@ webrtc_sources = [
     "modules/audio_coding/neteq/normal.cc",
     "modules/audio_coding/neteq/packet.cc",
     "modules/audio_coding/neteq/packet_buffer.cc",
-    "modules/audio_coding/neteq/post_decode_vad.cc",
     "modules/audio_coding/neteq/preemptive_expand.cc",
     "modules/audio_coding/neteq/random_vector.cc",
     "modules/audio_coding/neteq/red_payload_splitter.cc",
@@ -1737,7 +1735,6 @@ webrtc_sources = [
     "modules/audio_coding/neteq/normal.h",
     "modules/audio_coding/neteq/packet.h",
     "modules/audio_coding/neteq/packet_buffer.h",
-    "modules/audio_coding/neteq/post_decode_vad.h",
     "modules/audio_coding/neteq/preemptive_expand.h",
     "modules/audio_coding/neteq/random_vector.h",
     "modules/audio_coding/neteq/red_payload_splitter.h",
@@ -2139,7 +2136,6 @@ webrtc_sources = [
     "pc/sdp_utils.h",
     "pc/session_description.h",
     "pc/simulcast_description.h",
-    "pc/srtp_filter.h",
     "pc/srtp_session.h",
     "pc/srtp_transport.h",
     "pc/track_media_info_map.h",
@@ -2244,7 +2240,6 @@ webrtc_sources = [
     "call/rtp_stream_receiver_controller_interface.h",
     "api/audio/echo_control.h",
     "api/crypto/frame_encryptor_interface.h",
-    "api/crypto_params.h",
     "api/neteq/neteq_controller_factory.h",
     "api/transport/rtp/dependency_descriptor.h",
     "api/transport/rtp/dependency_descriptor.cc",
@@ -2802,11 +2797,9 @@ webrtc_sources = [
     "pc/simulcast_sdp_serializer.h",
     "stats/attribute.cc",
     "api/stats/attribute.h",
-    "stats/rtc_stats_member.cc",
     "api/environment/environment.h",
     "api/location.h",
     "api/ref_count.h",
-    "api/stats/rtc_stats_member.h",
     "common_video/h265/h265_inline.h",
     "modules/rtp_rtcp/source/frame_object.h",
     "logging/rtc_event_log/events/rtc_event_log_parse_status.h",
@@ -2837,8 +2830,6 @@ webrtc_sources = [
     "modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.h",
     "modules/video_coding/timing/inter_frame_delay_variation_calculator.h",
     "video/receive_statistics_proxy.h",
-    "api/video_codecs/h265_profile_tier_level.cc",
-    "modules/rtp_rtcp/source/rtp_packetizer_h265.cc",
     "api/rtp_sender_interface.cc",
     "modules/audio_processing/agc2/speech_level_estimator.cc",
     "modules/rtp_rtcp/source/flexfec_03_header_reader_writer.cc",
@@ -2846,6 +2837,7 @@ webrtc_sources = [
     "modules/audio_processing/agc2/input_volume_stats_reporter.cc",
     "media/base/media_channel_impl.cc",
     "p2p/base/stun_dictionary.cc",
+    "call/create_call.h",
     "call/create_call.cc",
     "modules/rtp_rtcp/source/frame_object.cc",
     "modules/video_coding/timing/decode_time_percentile_filter.cc",
@@ -2854,13 +2846,22 @@ webrtc_sources = [
     "video/frame_dumping_encoder.cc",
     "modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.cc",
     "video/receive_statistics_proxy.cc",
+    "api/transport/bandwidth_estimation_settings.h",
+    "api/video_codecs/h265_profile_tier_level.h",
+    "api/video_codecs/h265_profile_tier_level.cc",
+    "modules/rtp_rtcp/source/rtp_packetizer_h265.h",
+    "modules/rtp_rtcp/source/rtp_packetizer_h265.cc", 
     "modules/rtp_rtcp/source/video_rtp_depacketizer_h265.h",
     "modules/rtp_rtcp/source/video_rtp_depacketizer_h265.cc",
-    "modules/video_coding/codecs/h265/include/h265_globals.h",
-    "modules/rtp_rtcp/source/rtp_format_h265.h",
-    "modules/rtp_rtcp/source/rtp_format_h265.cc",
-    "modules/video_coding/h265_vps_sps_pps_tracker.h",
+    "modules/video_coding/include/video_error_codes_utils.h",
+    "modules/video_coding/include/video_error_codes_utils.cc",
+    "modules/rtp_rtcp/source/rtp_packet_h265_common.h",
+    "api/video_codecs/video_decoder_factory.cc",
     "modules/video_coding/h265_vps_sps_pps_tracker.cc",
+    "modules/video_coding/h265_vps_sps_pps_tracker.h",
+    "modules/video_coding/codecs/h265/include/h265_globals.h",
+    "common_video/h265/legacy_bit_buffer.cc",
+    "common_video/h265/legacy_bit_buffer.h",
 ]
 
 ios_objc_sources = [
@@ -3082,6 +3083,9 @@ ios_sources = [
     "objc/api/peerconnection/RTCPeerConnectionFactoryOptions+Private.h",
     "objc/api/peerconnection/RTCConfiguration+Native.h",
     "objc/api/peerconnection/RTCPeerConnectionFactoryOptions.h",
+    "objc/api/peerconnection/RTCRtpCodecCapability.h",
+    "objc/api/peerconnection/RTCRtpCodecCapability.mm",
+    "objc/api/peerconnection/RTCRtpCodecCapability+Private.h",
     "objc/native/api/network_monitor_factory.h",
     "objc/native/api/network_monitor_factory.mm",
     "objc/native/src/objc_network_monitor.h",
diff --git a/third-party/webrtc/webrtc b/third-party/webrtc/webrtc
index 94a1503c3d..e3dfe44d1a 160000
--- a/third-party/webrtc/webrtc
+++ b/third-party/webrtc/webrtc
@@ -1 +1 @@
-Subproject commit 94a1503c3d426e29ec8676a0caa994e98ed61205
+Subproject commit e3dfe44d1aea15a1aabded19d71884412e772245