I’m building a plugin to integrate libmediasoupclient into Unreal Engine. It makes use of the Unreal built-in webRTC module. When loading mediasoup device by calling mediasoup::Device::Load(...), Unreal crashes with this report:
Abort signal received
ucrtbase
ucrtbase
UnrealEditor_ShibuChat_Win64_DebugGame!rtc::webrtc_checks_impl::UnreachableCodeReached()
UnrealEditor_ShibuChat_Win64_DebugGame!rtc::webrtc_checks_impl::FatalLog()
UnrealEditor_ShibuChat_Win64_DebugGame!cricket::WebRtcVoiceEngine::Init()
UnrealEditor_ShibuChat_Win64_DebugGame!cricket::CompositeMediaEngine::Init()
UnrealEditor_ShibuChat_Win64_DebugGame!cricket::ChannelManager::Create()
UnrealEditor_ShibuChat_Win64_DebugGame!std::vector<std::unique_ptr<cricket::VoiceChannel,std::default_delete<cricket::VoiceChannel> >,std::allocator<std::unique_ptr<cricket::VoiceChannel,std::default_delete<cricket::VoiceChannel> > > >::erase()
UnrealEditor_ShibuChat_Win64_DebugGame!rtc::Thread::Restart()
UnrealEditor_ShibuChat_Win64_DebugGame!rtc::Thread::QueuedTaskHandler::OnMessage()
UnrealEditor_ShibuChat_Win64_DebugGame!rtc::Thread::Dispatch()
UnrealEditor_ShibuChat_Win64_DebugGame!rtc::Thread::ProcessMessages()
UnrealEditor_ShibuChat_Win64_DebugGame!rtc::Thread::PreRun()
kernel32
ntdll
From which I think cricket::WebRtcVoiceEngine::Init() causes the crash.
This is the function I call to load mediasoup device:
static decltype(auto) CreateAndLoadDevice(const FString& InRtpCapabilityJsonString) noexcept
{
mediasoupclient::Device device;
device.Load(nlohmann::json::parse(StringCast<ANSICHAR>(*InRtpCapabilityJsonString).Get()));
return device;
}
This is the RTP capabilities I use for the input which I get with socket.io.
{
"codecs":
[
{
"channels": 2,
"clockRate": 48000,
"kind": "audio",
"mimeType": "audio/opus",
"parameters": {},
"preferredPayloadType": 100,
"rtcpFeedback":
[
{
"parameter": "",
"type": "transport-cc"
}
]
},
{
"clockRate": 90000,
"kind": "video",
"mimeType": "video/VP8",
"parameters":
{
"x-google-start-bitrate": 1000
},
"preferredPayloadType": 101,
"rtcpFeedback":
[
{
"parameter": "",
"type": "nack"
},
{
"parameter": "pli",
"type": "nack"
},
{
"parameter": "fir",
"type": "ccm"
},
{
"parameter": "",
"type": "goog-remb"
},
{
"parameter": "",
"type": "transport-cc"
}
]
},
{
"clockRate": 90000,
"kind": "video",
"mimeType": "video/rtx",
"parameters":
{
"apt": 101
},
"preferredPayloadType": 102,
"rtcpFeedback": []
}
],
"headerExtensions":
[
{
"direction": "sendrecv",
"kind": "audio",
"preferredEncrypt": false,
"preferredId": 1,
"uri": "urn:ietf:params:rtp-hdrext:sdes:mid"
},
{
"direction": "sendrecv",
"kind": "video",
"preferredEncrypt": false,
"preferredId": 1,
"uri": "urn:ietf:params:rtp-hdrext:sdes:mid"
},
{
"direction": "recvonly",
"kind": "video",
"preferredEncrypt": false,
"preferredId": 2,
"uri": "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id"
},
{
"direction": "recvonly",
"kind": "video",
"preferredEncrypt": false,
"preferredId": 3,
"uri": "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id"
},
{
"direction": "sendrecv",
"kind": "audio",
"preferredEncrypt": false,
"preferredId": 4,
"uri": "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"
},
{
"direction": "sendrecv",
"kind": "video",
"preferredEncrypt": false,
"preferredId": 4,
"uri": "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"
},
{
"direction": "recvonly",
"kind": "audio",
"preferredEncrypt": false,
"preferredId": 5,
"uri": "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"
},
{
"direction": "sendrecv",
"kind": "video",
"preferredEncrypt": false,
"preferredId": 5,
"uri": "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"
},
{
"direction": "sendrecv",
"kind": "video",
"preferredEncrypt": false,
"preferredId": 6,
"uri": "http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07"
},
{
"direction": "sendrecv",
"kind": "video",
"preferredEncrypt": false,
"preferredId": 7,
"uri": "urn:ietf:params:rtp-hdrext:framemarking"
},
{
"direction": "sendrecv",
"kind": "audio",
"preferredEncrypt": false,
"preferredId": 10,
"uri": "urn:ietf:params:rtp-hdrext:ssrc-audio-level"
},
{
"direction": "sendrecv",
"kind": "video",
"preferredEncrypt": false,
"preferredId": 11,
"uri": "urn:3gpp:video-orientation"
},
{
"direction": "sendrecv",
"kind": "video",
"preferredEncrypt": false,
"preferredId": 12,
"uri": "urn:ietf:params:rtp-hdrext:toffset"
},
{
"direction": "sendrecv",
"kind": "audio",
"preferredEncrypt": false,
"preferredId": 13,
"uri": "http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time"
},
{
"direction": "sendrecv",
"kind": "video",
"preferredEncrypt": false,
"preferredId": 13,
"uri": "http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time"
}
]
}
I tried to look around in the Pixel Streaming codes but couldn’t find any call of that webRTC function.
Does anyone know what’s going on?