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Question: There is no sound when using gaming headphones. Is there a way to solve this? #240
Description
Here is an example of such a problem with LIBERTY GAMER-SPLENDOR headphones. I'm attaching a part of the log that shows what's going on.
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-26] WARN - (openssl_stream_adapter.cc:148): WebRTC-ForceDtls13:
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-26] WARN - (openssl_stream_adapter.cc:148): WebRTC-ForceDtls13:
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-26] INFO - (dtls_transport.cc:497): DtlsTransport[0|1|__]: DTLS setup complete, dtls_in_stun: 0
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-24] INFO - (sdp_offer_answer.cc:3092): Session: 4959460482492424513 Old state: have-local-offer New state: stable
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-25] INFO - (channel.cc:544): Channel enabled: {mid: 0, media_type: audio}
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-25] INFO - (channel.cc:879): Changing voice state, recv=1 send=0 for {mid: 0, media_type: audio}
2025-12-09 12:58:20,244 57424 [root - Thread-24] INFO - [PeerConnectionObserver] > Signaling change: STABLE
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-25] INFO - (channel.cc:949): Setting remote voice description for {mid: 0, media_type: audio}
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-25] INFO - (webrtc_voice_engine.cc:1313): WebRtcVoiceMediaChannel::SetSenderParameters: {codecs: [AudioCodec[111:opus:48000:0:2], AudioCodec[110:telephone-event:48000:0:1]], extensions: [], extmap-allow-mixed: false, max_bandwidth_bps: -1, mid: 0, options: AudioOptions {}, rtcp: {reduced_size:0, remote_estimate:0}}
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-25] INFO - (audio_send_stream.cc:213): AudioSendStream::ConfigureStream: {rtp: {ssrc: 1615178193, extmap-allow-mixed: true, extensions: [], c_name: qd+qHwrY/ug6BLle}, rtcp_report_interval_ms: 5000, send_transport: (Transport), min_bitrate_bps: 64000, max_bitrate_bps: 64000, has audio_network_adaptor_config: false, has_dscp: false, send_codec_spec: {nack_enabled: false, enable_non_sender_rtt: false, cng_payload_type: <unset>, red_payload_type: <unset>, payload_type: 111, format: {name: opus, clockrate_hz: 48000, num_channels: 2, parameters: {maxaveragebitrate: 64000, maxplaybackrate: 48000, minptime: 10, ptime: 20, useinbandfec: 1}}}}
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-25] INFO - (audio_send_stream.cc:213): AudioSendStream::ConfigureStream: {rtp: {ssrc: 1615178193, extmap-allow-mixed: false, extensions: [], c_name: qd+qHwrY/ug6BLle}, rtcp_report_interval_ms: 5000, send_transport: (Transport), min_bitrate_bps: 64000, max_bitrate_bps: 64000, has audio_network_adaptor_config: false, has_dscp: false, send_codec_spec: {nack_enabled: false, enable_non_sender_rtt: false, cng_payload_type: <unset>, red_payload_type: <unset>, payload_type: 111, format: {name: opus, clockrate_hz: 48000, num_channels: 2, parameters: {maxaveragebitrate: 64000, maxplaybackrate: 48000, minptime: 10, ptime: 20, useinbandfec: 1}}}}
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-25] INFO - (audio_send_stream.cc:213): AudioSendStream::ConfigureStream: {rtp: {ssrc: 1615178193, mid: 0, extmap-allow-mixed: false, extensions: [], c_name: qd+qHwrY/ug6BLle}, rtcp_report_interval_ms: 5000, send_transport: (Transport), min_bitrate_bps: 64000, max_bitrate_bps: 64000, has audio_network_adaptor_config: false, has_dscp: false, send_codec_spec: {nack_enabled: false, enable_non_sender_rtt: false, cng_payload_type: <unset>, red_payload_type: <unset>, payload_type: 111, format: {name: opus, clockrate_hz: 48000, num_channels: 2, parameters: {maxaveragebitrate: 64000, maxplaybackrate: 48000, minptime: 10, ptime: 20, useinbandfec: 1}}}}
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-25] INFO - (webrtc_voice_engine.cc:1762): WebRtcVoiceMediaChannel::SetMaxSendBitrate.
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-25] INFO - (webrtc_voice_engine.cc:1290): Setting voice channel options: AudioOptions {}
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-25] INFO - (webrtc_voice_engine.cc:603): WebRtcVoiceEngine::ApplyOptions: AudioOptions {aec: 1, agc: 1, ns: 1, hf: 1, audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, }
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-25] INFO - (audio_device_impl.cc:829): virtual webrtc::AudioDeviceModuleImpl::BuiltInAECIsAvailable
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-25] INFO - (audio_device_impl.cc:832): output: 1
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-25] INFO - (audio_device_impl.cc:837): virtual webrtc::AudioDeviceModuleImpl::EnableBuiltInAEC(1)
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-25] INFO - (audio_device_impl.cc:840): output: 0
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-25] INFO - (webrtc_voice_engine.cc:669): Disabling EC since built-in EC will be used instead
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-25] INFO - (audio_device_impl.cc:845): virtual webrtc::AudioDeviceModuleImpl::BuiltInAGCIsAvailable
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-25] INFO - (audio_device_impl.cc:848): output: 0
2025-12-09 12:58:20,244 57424 [WebRTC.Native - Thread-25] INFO - (audio_device_impl.cc:861): virtual webrtc::AudioDeviceModuleImpl::BuiltInNSIsAvailable
2025-12-09 12:58:20,245 57425 [WebRTC.Native - Thread-25] INFO - (audio_device_impl.cc:864): output: 0
2025-12-09 12:58:20,245 57425 [WebRTC.Native - Thread-25] INFO - (audio_processing_impl.cc:700): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_amplifier: { enabled: 0, fixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 1 }, echo_canceller: { enabled: 0, mobile_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 1, level: High }, transient_suppression: { enabled: 0 }, gain_controller1: { enabled: 1, mode: AdaptiveAnalog, target_level_dbfs: 3, compression_gain_db: 9, enable_limiter: 1, analog_gain_controller { enabled: 1, startup_min_volume: 0, clipped_level_min: 70, enable_digital_adaptive: 1, clipped_level_step: 15, clipped_ratio_threshold: 0.1, clipped_wait_frames: 300, clipping_predictor: { enabled: 0, mode: 0, window_length: 5, reference_window_length: 5, reference_window_delay: 5, clipping_threshold: -1, crest_factor_margin: 3, use_predicted_step: 1 }}}, gain_controller2: { enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, headroom_db: 5, max_gain_db: 50, initial_gain_db: 15, max_gain_change_db_per_second: 6, max_output_noise_level_dbfs: -50 }, input_volume_control : { enabled 0}}
2025-12-09 12:58:20,245 57425 [WebRTC.Native - Thread-25] INFO - (webrtc_voice_engine.cc:1304): Set voice send channel options. Current options: AudioOptions {aec: 1, agc: 1, ns: 1, hf: 1, audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, }
2025-12-09 12:58:20,245 57425 [WebRTC.Native - Thread-25] INFO - (webrtc_voice_engine.cc:2335): AddRecvStream: {id:a0;ssrcs:[3930923763];ssrc_groups:;cname:RKvD0n2Sq8jK1Sz7;stream_ids:3ovvVp1y2dfAt2JYgzsG0zoF1wHOhDpa;}
2025-12-09 12:58:20,245 57425 [WebRTC.Native - Thread-25] INFO - (delay_manager.cc:55): Delay manager config: quantile=0.95 forget_factor=0.983 start_forget_weight=2 resample_interval_ms=500 use_reorder_optimizer=1 reorder_forget_factor=0.9993 ms_per_loss_percent=20
2025-12-09 12:58:20,245 57425 [WebRTC.Native - Thread-25] INFO - (neteq_impl.cc:165): NetEq config: sample_rate_hz=48000, max_packets_in_buffer=200, min_delay_ms=0, enable_fast_accelerate=false, enable_muted_state=true, enable_rtx_handling=false
2025-12-09 12:58:20,245 57425 [WebRTC.Native - Thread-25] INFO - (audio_receive_stream.cc:120): AudioReceiveStreamImpl: 3930923763
2025-12-09 12:58:20,245 57425 [WebRTC.Native - Thread-25] INFO - (audio_receive_stream.cc:192): AudioReceiveStreamImpl::Start: 3930923763
2025-12-09 12:58:20,245 57425 [WebRTC.Native - Thread-25] INFO - (audio_device_impl.cc:775): virtual webrtc::AudioDeviceModuleImpl::Playing
2025-12-09 12:58:20,245 57425 [WebRTC.Native - Thread-25] INFO - (audio_device_impl.cc:712): virtual webrtc::AudioDeviceModuleImpl::InitPlayout
2025-12-09 12:58:20,245 57425 [WebRTC.Native - Thread-25] INFO - (audio_device_impl.cc:738): virtual webrtc::AudioDeviceModuleImpl::PlayoutIsInitialized
2025-12-09 12:58:20,250 57430 [WebRTC.Native - Thread-25] INFO - (audio_device_core_win.cc:1894): nChannels=2, nSamplesPerSec=48000 is not supported. Closest match: nChannels=8, nSamplesPerSec=48000
2025-12-09 12:58:20,251 57431 [WebRTC.Native - Thread-25] INFO - (audio_device_core_win.cc:1894): nChannels=1, nSamplesPerSec=48000 is not supported. Closest match: nChannels=8, nSamplesPerSec=48000
2025-12-09 12:58:20,252 57432 [WebRTC.Native - Thread-25] INFO - (audio_device_core_win.cc:1894): nChannels=2, nSamplesPerSec=44100 is not supported. Closest match: nChannels=8, nSamplesPerSec=48000
2025-12-09 12:58:20,252 57432 [WebRTC.Native - Thread-25] INFO - (audio_device_core_win.cc:1894): nChannels=1, nSamplesPerSec=44100 is not supported. Closest match: nChannels=8, nSamplesPerSec=48000
2025-12-09 12:58:20,253 57433 [WebRTC.Native - Thread-25] INFO - (audio_device_core_win.cc:1894): nChannels=2, nSamplesPerSec=16000 is not supported. Closest match: nChannels=8, nSamplesPerSec=48000
2025-12-09 12:58:20,254 57434 [WebRTC.Native - Thread-25] INFO - (audio_device_core_win.cc:1894): nChannels=1, nSamplesPerSec=16000 is not supported. Closest match: nChannels=8, nSamplesPerSec=48000
2025-12-09 12:58:20,255 57435 [WebRTC.Native - Thread-25] INFO - (audio_device_core_win.cc:1894): nChannels=2, nSamplesPerSec=96000 is not supported. Closest match: nChannels=8, nSamplesPerSec=48000
2025-12-09 12:58:20,255 57435 [WebRTC.Native - Thread-25] INFO - (audio_device_core_win.cc:1894): nChannels=1, nSamplesPerSec=96000 is not supported. Closest match: nChannels=8, nSamplesPerSec=48000
2025-12-09 12:58:20,256 57436 [WebRTC.Native - Thread-25] INFO - (audio_device_core_win.cc:1894): nChannels=2, nSamplesPerSec=32000 is not supported. Closest match: nChannels=8, nSamplesPerSec=48000
2025-12-09 12:58:20,257 57437 [WebRTC.Native - Thread-25] INFO - (audio_device_core_win.cc:1894): nChannels=1, nSamplesPerSec=32000 is not supported. Closest match: nChannels=8, nSamplesPerSec=48000
2025-12-09 12:58:20,258 57438 [WebRTC.Native - Thread-25] INFO - (audio_device_core_win.cc:1894): nChannels=2, nSamplesPerSec=8000 is not supported. Closest match: nChannels=8, nSamplesPerSec=48000
2025-12-09 12:58:20,258 57438 [WebRTC.Native - Thread-25] INFO - (audio_device_core_win.cc:1894): nChannels=1, nSamplesPerSec=8000 is not supported. Closest match: nChannels=8, nSamplesPerSec=48000
2025-12-09 12:58:20,262 57442 [WebRTC.Native - Thread-25] WARN - Error: (audio_device_core_win.cc:1981): IAudioClient::Initialize() failed:
2025-12-09 12:58:20,263 57443 [WebRTC.Native - Thread-25] WARN - Error: (audio_device_core_win.cc:1981): IAudioClient::Initialize() failed:
2025-12-09 12:58:20,263 57443 [WebRTC.Native - Thread-25] WARN - Error: (audio_device_core_win.cc:1981): IAudioClient::Initialize() failed:
2025-12-09 12:58:20,268 57448 [WebRTC.Native - Thread-25] WARN - Error: (audio_device_core_win.cc:4162): Core Audio method failed (hr=-2004287480)
2025-12-09 12:58:20,268 57448 [WebRTC.Native - Thread-25] WARN - Error: (audio_device_core_win.cc:4162): Core Audio method failed (hr=-2004287480)
2025-12-09 12:58:20,268 57448 [WebRTC.Native - Thread-25] WARN - Error: (audio_device_core_win.cc:4162): Core Audio method failed (hr=-2004287480)
2025-12-09 12:58:20,268 57448 [WebRTC.Native - Thread-25] WARN - Error: (audio_device_core_win.cc:4165): Error details: ꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪ
2025-12-09 12:58:20,268 57448 [WebRTC.Native - Thread-25] WARN - Error: (audio_device_core_win.cc:4165): Error details: ꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪ
2025-12-09 12:58:20,268 57448 [WebRTC.Native - Thread-25] WARN - Error: (audio_device_core_win.cc:4165): Error details: ꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪꪪ
2025-12-09 12:58:20,268 57448 [WebRTC.Native - Thread-25] INFO - (audio_device_impl.cc:718): output: -1
2025-12-09 12:58:20,268 57448 [WebRTC.Native - Thread-25] INFO - (channel.cc:766): Add remote ssrc: 3930923763 to {mid: 0, media_type: audio}
2025-12-09 12:58:20,268 57448 [WebRTC.Native - Thread-25] INFO - (channel.cc:879): Changing voice state, recv=1 send=0 for {mid: 0, media_type: audio}
2025-12-09 12:58:20,268 57448 [WebRTC.Native - Thread-24] INFO - (peer_connection.cc:1867): Changing IceConnectionState 0 => 1
2025-12-09 12:58:20,269 57449 [WebRTC.Native - Thread-26] INFO - (connection.cc:260): Conn[fd3c5080:0:Net[Intel(R):192.168.0.x/32:Ethernet:id=1]:Fx+67lAp:1:0:relay:udp:77.223.98.x:65158->qLMK+XIC:1:659136:host:udp:185.99.140.x:21490|C--W|-|0|0|2830967613702143|-]: Connection created
2025-12-09 12:58:20,269 57449 [root - Thread-24] INFO - [PeerConnectionObserver] > Ice connection change: CHECKING
2025-12-09 12:58:20,269 57449 [WebRTC.Native - Thread-26] INFO - (turn_port.cc:1622): Port[f26d2730:0:1:0:relay:Net[Intel(R):192.168.0.x/32:Ethernet:id=1]]: TURN create permission request sent, id=484d74434b43756a737a4f4f
2025-12-09 12:58:20,269 57449 [WebRTC.Native - Thread-26] INFO - (p2p_transport_channel.cc:1425): Channel[0|1|__]: Created connection with origin: 2, total: 1
2025-12-09 12:58:20,269 57449 [WebRTC.Native - Thread-26] INFO - (connection.cc:260): Conn[8afae290:0:Net[Intel(R):192.168.0.x/32:Ethernet:id=1]:S+8jNHH7:1:0:relay:udp:80.249.148.x:65425->qLMK+XIC:1:659136:host:udp:185.99.140.x:21490|C--W|-|0|0|2830967613702655|-]: Connection created
2025-12-09 12:58:20,269 57449 [WebRTC.Native - Thread-26] INFO - (turn_port.cc:1622): Port[f26d1c90:0:1:0:relay:Net[Intel(R):192.168.0.x/32:Ethernet:id=1]]: TURN create permission request sent, id=59462f716f6174656552554d
2025-12-09 12:58:20,269 57449 [WebRTC.Native - Thread-26] INFO - (p2p_transport_channel.cc:1425): Channel[0|1|__]: Created connection with origin: 2, total: 2
2025-12-09 12:58:20,269 57449 [WebRTC.Native - Thread-26] INFO - (connection.cc:260): Conn[fceb2120:0:Net[Intel(R):192.168.0.x/32:Ethernet:id=1]:4ywg+sMd:1:0:host:udp:192.168.0.x:60473->qLMK+XIC:1:659136:host:udp:185.99.140.x:21490|C--W|-|0|0|2830971808005631|-]: Connection created
2025-12-09 12:58:20,269 57449 [WebRTC.Native - Thread-26] INFO - (p2p_transport_channel.cc:1425): Channel[0|1|__]: Created connection with origin: 2, total: 3
2025-12-09 12:58:20,269 57449 [WebRTC.Native - Thread-26] INFO - (p2p_transport_channel.cc:1874): Channel[0|1|__]: Transport channel state changed from 0 to 1
2025-12-09 12:58:20,269 57449 [WebRTC.Native - Thread-26] INFO - (jsep_transport_controller.cc:1344): 0 Transport 1 state changed. Check if state is complete.
2025-12-09 12:58:20,269 57449 [WebRTC.Native - Thread-26] INFO - (jsep_transport_controller.cc:1344): 0 Transport 1 state changed. Check if state is complete.
2025-12-09 12:58:20,269 57449 [WebRTC.Native - Thread-26] INFO - (p2p_transport_channel.cc:1671): Channel[0|1|__]: Have a pingable connection for the first time; starting to ping.
2025-12-09 12:58:20,269 57449 [WebRTC.Native - Thread-24] INFO - (sdp_offer_answer.cc:2238): Processing the MSIDs for MID=0 (streams=[3ovvVp1y2dfAt2JYgzsG0zoF1wHOhDpa]).
2025-12-09 12:58:20,269 57449 [WebRTC.Native - Thread-24] INFO - (sdp_offer_answer.cc:2248): Processing the addition of a remote track for MID=0.
2025-12-09 12:58:20,269 57449 [WebRTC.Native - Thread-24] INFO - (rtp_transceiver.cc:503): Changing transceiver (MID=0) current direction from <not set> to kSendRecv.
2025-12-09 12:58:20,269 57449 [WebRTC.Native - Thread-26] INFO - (connection.cc:1607): Conn[fceb2120:0:Net[Intel(R):192.168.0.x/32:Ethernet:id=1]:4ywg+sMd:1:0:host:udp:192.168.0.x:60473->qLMK+XIC:1:659136:host:udp:185.99.140.x:21490|C--W|-|0|0|2830971808005631|-]: Sent STUN BINDING request, id=76436e6f4b56656656304845, use_candidate=1, nomination=0
2025-12-09 12:58:20,270 57450 [WebRTC.Native - Thread-25] INFO - (webrtc_voice_engine.cc:2497): SetBaseMinimumPlayoutDelayMs() to 0 for recv stream with ssrc 3930923763
2025-12-09 12:58:20,270 57450 [WebRTC.Native - Thread-25] INFO - (webrtc_voice_engine.cc:2448): WRVMC::SetOutputVolume({ssrc=3930923763}, {volume=1.00})
2025-12-09 12:58:20,270 57450 [WebRTC.Native - Thread-25] INFO - (webrtc_voice_engine.cc:2458): WRVMC::SetOutputVolume => (stream with SSRC 3930923763 now uses volume 1.00)
2025-12-09 12:58:20,270 57450 [root - Thread-24] INFO - [PeerConnectionObserver] > Add track, rtp receiver: dev.onvoid.webrtc.RTCRtpReceiver@25f2ea4a, media streams: [Ldev.onvoid.webrtc.media.MediaStream;@53765d1
2025-12-09 12:58:20,270 57450 [WebRTC.Native - Thread-24] INFO - (peer_connection.cc:1886): Changing standardized IceConnectionState 0 => 1
The audio interface is initialized, but informs webrtc that it operates at a frequency of 48000 and 8 channels. Webrtc selects the frequencies available to it 48000, 44100, 16000, 96000, 32000, 8000, but without success. As a result, IAudioClient returns an initialization error.
It is unclear to me whether this can be fixed and, most importantly, where. The error indicates that this is happening inside the native part of webrtc, because apparently the selected device is used as a multimedia role, not communication. Maybe someone has some ideas on this? Thank you in advance!
Activity
devopvoid commentedon Aug 15, 2026
Hi, sorry I'm late, I have no idea why you experience such issues. My best bet would be https://groups.google.com/g/discuss-webrtc or opening a bug ticket within WebRTC itself.
Stasyanych commentedon Aug 17, 2026
Thank you for your answer!
devopvoid commentedon Sep 13, 2026
One concrete thing worth trying, based on the log: the device is reporting itself to WASAPI as an 8-channel endpoint, and
IAudioClient::Initialize()fails for every channel count/sample-rate combination WebRTC's Windows Core Audio backend tries (audio_device_core_win.cc) because none of them match that fixed shared-mode format. This is a common quirk with gaming headsets that expose a virtual "7.1 surround" endpoint by default — Windows won't let a shared-mode client open it with a different channel count.If your headset software has a way to switch its Windows output endpoint to plain stereo (most of these have a toggle for "stereo" vs. "surround/gaming" mode, or expose a separate 2-channel device alongside the virtual surround one), selecting that device instead should let
IAudioClient::Initializesucceed. Since this is Windows' own shared-mode format negotiation inside libwebrtc's ADM (not something webrtc-java's JNI layer controls), I don't have a fix on our side beyond that — closing this out as answered, but feel free to reopen if the device-switch route doesn't pan out and you find something more specific.devopvoid commentedon Oct 11, 2026
A patch is in development. Reopening.
devopvoid commentedon Oct 12, 2026
@Stasyanych Can you test this PR 358?