[Voice Room] Segmentation fault Error

[knassar702]

Steps to reproduce

1. go to group
2. start/join a voice room

description

when i try to join a voice room in my group , i got Segmentation fault in the terminal and GUI exit

Configuration

• Operating system: Void Linux
• Version of Telegram Desktop: 2.8.10

Installation source (Linux Only) - the official website / GitHub releases / flatpak / snap / distribution package:

• XBPS (official void package manager)

Logs:

(telegram-desktop:26272): Telegram-WARNING **: 20:51:36.393: Application was built without embedded fonts, this may lead to font issues.
QStandardPaths: XDG_RUNTIME_DIR not set, defaulting to '/tmp/runtime-knassar702'
QStandardPaths: XDG_RUNTIME_DIR not set, defaulting to '/tmp/runtime-knassar702'
[ALSOFT] (EE) Failed to set real-time priority for thread: Operation not permitted (1)
error: : cannot open
error: : cannot open
error: : cannot open
Invalid return value 0 for stream protocol
Invalid return value 0 for stream protocol
[ALSOFT] (EE) Failed to set real-time priority for thread: Operation not permitted (1)
(field_trial.cc:140): Setting field trial string:WebRTC-Audio-Allocation/min:32kbps,max:32kbps/WebRTC-Audio-OpusMinPacketLossRate/Enabled-1/WebRTC-TaskQueuePacer/Enabled/WebRTC-VP8ConferenceTemporalLayers/1/WebRTC-Audio-MinimizeResamplingOnMobile/Enabled/
(openssl_key_pair.cc:38): Making key pair
(openssl_key_pair.cc:91): Returning key pair
(audio_processing_impl.cc:277): Injected APM submodules:
Echo control factory: 0
Echo detector: 0
Capture analyzer: 0
Capture post processor: 1
Render pre processor: 0
(openssl_certificate.cc:59): Making certificate for WebRTC
(openssl_certificate.cc:109): Returning certificate
(dtls_srtp_transport.cc:62): Setting RTCP Transport on null transport 0
(dtls_srtp_transport.cc:67): Setting RTP Transport on null transport 0
(audio_device_buffer.cc:64): AudioDeviceBuffer::ctor
(audio_device_buffer.cc:180): SetRecordingSampleRate(48000)
(audio_device_buffer.cc:186): SetPlayoutSampleRate(48000)
(audio_device_buffer.cc:200): SetRecordingChannels(1)
(audio_device_buffer.cc:206): SetPlayoutChannels(2)
(basic_port_allocator.cc:375): Start getting ports with turn_port_prune_policy 0
(basic_port_allocator.cc:375): Start getting ports with turn_port_prune_policy 0
(p2p_transport_channel.cc:532): Set backup connection ping interval to 25000 milliseconds.
(p2p_transport_channel.cc:541): Set ICE receiving timeout to 2500 milliseconds
(p2p_transport_channel.cc:548): Set ping most likely connection to 1
(p2p_transport_channel.cc:555): Set stable_writable_connection_ping_interval to 2500
(p2p_transport_channel.cc:568): Set presume writable when fully relayed to 0
(p2p_transport_channel.cc:586): Set regather_on_failed_networks_interval to 8000
(p2p_transport_channel.cc:593): Set receiving_switching_delay to 1000
(p2p_transport_channel.cc:466): Set ICE ufrag: p873 pwd: ZUaqf4IX8PbGQblts0XEaPx5 on transport transport
(dtls_srtp_transport.cc:62): Setting RTCP Transport on transport transport 0
(dtls_srtp_transport.cc:67): Setting RTP Transport on transport transport 43c35000
(webrtc_voice_engine.cc:269): WebRtcVoiceEngine::WebRtcVoiceEngine
(webrtc_video_engine.cc:629): WebRtcVideoEngine::WebRtcVideoEngine()
(webrtc_voice_engine.cc:291): WebRtcVoiceEngine::Init
Segmentation fault

[ilya-fedin]

Please don't report package-specific bugs here, use your distro bugtracker.

[github-actions]

This issue has been automatically locked since there has not been any recent activity after it was closed. Please open a new issue for related bugs.