Unable to Receive Audio Track on C++ WebRTC Receiver (Video Works)

[Agneya Singh]

Hi everyone,

I'm working on a custom WebRTC application where one peer is a browser based sender, and the other peer is a C++ native receiver.

Goal

I want to receive both audio and video tracks from the remote peer in my C++ application. The video stream is working correctly. I can attach a VideoSink and process frames as expected. However, audio is not being received even though the browser is sending audio.
For audio I am able to reach onTrack on the C++ application but I cannot reach the OnData.

1. JavaScript (sender):

I'm sending both audio and video from the browser like this:

stream.getAudioTracks().forEach(track => {
  this.pc.addTrack(track, stream);
});
stream.getVideoTracks().forEach(track => {
  this.pc.addTrack(track, stream);
});

  On the JS side, chrome://webrtc-internals shows that audio packets are being sent (framessent number increases, packetssent number increases)

2. C++ application

In my C++ Conductor::OnTrack() handler, I'm successfully receiving the video track, but the audio sink never gets triggered. Here's my code:

void Conductor::OnTrack(rtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver)
{
    auto track = transceiver->receiver()->track();
    printf("\n Inside Conductor::OnTrack"); fflush(stdout);

    if (track->kind() == webrtc::MediaStreamTrackInterface::kVideoKind) {
        rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track =
            static_cast<webrtc::VideoTrackInterface*>(track.get());

        remote_video_sink_.reset(new VideoFrameExtractor());
        rtc::VideoSinkWants wants;
        wants.rotation_applied = true;
        video_track->AddOrUpdateSink(remote_video_sink_.get(), wants);
        std::cout << "OnTrack Video track received and sink attached.\n";
    }

    else if (track->kind() == webrtc::MediaStreamTrackInterface::kAudioKind) {
        rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track =
            static_cast<webrtc::AudioTrackInterface*>(track.get());

        std::cout << "[OnTrack] Audio track received:\n";
        std::cout << "  ID      : " << audio_track->id() << "\n";
        std::cout << "  State   : " << (audio_track->state() == webrtc::MediaStreamTrackInterface::TrackState::kLive ? "Live" : "Ended") << "\n";
        std::cout << "  Enabled : " << (audio_track->enabled() ? "true" : "false") << "\n";

        remote_audio_sink_.reset(new AudioSinkImpl());
        audio_track->AddSink(remote_audio_sink_.get());

        printf("\n [OnTrack] Audio track attached.\n"); fflush(stdout);
    }
}

OnTrack() is triggered for both audio and video.
Audio track appears to be live and enabled.
I'm using a custom AudioSinkImpl, but its OnData() callback is never triggered.
STUN and ICE are working (I receive remote video).


Questions:

1. Is there anything else I need to enable in C++ to activate audio decoding or sink processing?

2. Are there known conditions under which AudioTrack::AddSink() would fail to deliver audio?

3. Is it possible that the audio is not being routed because of codec negotiation? (I am using opus 48000 Hz and 16bit)