Trace/breakpoint trap in GetNativeRtpCapabilities

mediasoup-demo
1

I was testing the c++ demo , but I keep getting Trace/breakpoint trap from GetNativeRtpCapabilities
GitHub - versatica/mediasoup-broadcaster-demo: mediasoup broadcaster demo (libmediasoupclient). is there any was changed in the functions?

json Handler::GetNativeRtpCapabilities(const PeerConnection::Options* peerConnectionOptions)
	{
		MSC_TRACE();std::unique_ptr<PeerConnection::PrivateListener> privateListener(new PeerConnection::PrivateListener());
		std::unique_ptr<PeerConnection> pc(new PeerConnection(privateListener.get(), peerConnectionOptions));
		(void)pc->AddTransceiver(cricket::MediaType::MEDIA_TYPE_AUDIO);
		(void)pc->AddTransceiver(cricket::MediaType::MEDIA_TYPE_VIDEO);
		webrtc::PeerConnectionInterface::RTCOfferAnswerOptions options;

		// May throw.
		auto offer                 = pc->CreateOffer(options);
		auto sdpObject             = sdptransform::parse(offer);
		auto nativeRtpCapabilities = Sdp::Utils::extractRtpCapabilities(sdpObject);

		return nativeRtpCapabilities;
	}
2

Nothing coming to my mind that can cause this due to the code in libmediasoupclient.

3

@jmillan thanks for your input. Do you think the issue could be from the demo itself?

I added a breakpoint. When this line get called this->device.load(routerRtpCapabilities)

Then it just call auto nativeRtpCapabilities = Handler::GetNativeRtpCapabilities(peerConnectionOption)

routerRtpCapabilities returned from the server.

Where do you think I should dig deeper? I feel I could be looking into wrong direction.

Thanks!

4

What is the problem you are having?

5

I have experienced multiple issues, when I fix one I run into another one :frowning:
I was able to compile the demo, but when I run it I kept getting the error below

Fatal error in: …/…/media/engine/adm_helpers.cc, line 39

last system error: 88

Check failed: 0 == adm->Init() (0 vs. -1)

Failed to initialize the ADM.Aborted

I searched for days trying to find a fix for this issue and tried many solutions dealing with audio on a linux box, tried pulseaudio, pipewire and installed libwebrtc-audio-processing-dev, but did not go anywhere.

I tried to disable the audio init from webrtc and from the demo, but I kept getting
Trace/breakpoint trap when getting nativeRtpCapabilities get called

I reached out to webrtc google group and I was told to update the libwebrtc because m94 was very old.

I built multiple versions of libwebrc, but mediasoup client does not compile with any version above m98, when I tried to build mediasoupclient using m105, I was getting some errors.
/webrtc/mediasoup/libmediasoupclient/src/Handler.cpp:212:88: error: no matching function for call to ‘mediasoupclient::PeerConnection::AddTransceiver(webrtc::MediaStreamTrackInterface*&, webrtc::RtpTransceiverInit&)’

right now I am stuck between google and mediasoup where google folks think m94 is old and unable to help and mediasoupclient does not work with recent libwebrtc. It is not clear if you are planning to update the client to m105

I was hoping that I can build the demo as I thought it is fastest way to evaluate mediasoup, but it is very problematic more I initially thought