I am using LibJitsi to setup an RTP media stream to send DTMF tones through my SIP stack.
When I am using the MediaService to create the media stream I am able to establish the RTP Session if I instantiate it with a MediaDevice where the MediaDevice is setup like:
MediaDevice device = mediaService.getDefaultDevice(MediaType.AUDIO, MediaUseCase.CALL);
AudioMediaStream mediaStream = (AudioMediaStream) mediaService.createMediaStream(device);
However, if I try to use the media service to create the stream by just using the audio MediaType I cannot get the RTP Session to connect once I attempt to start the stream.
I don’t have an issue using the device but that seems to default the capture device to the microphone on my computer and I am not sure if that will interfere with me sending the DTMF tones with MediaStream::startSendingDTMF. Right now I am not seeing any errors with the approach but I also cannot clearly hear the DTMF tones on the line either.
I also have some concerns about how this will work once deployed to my server since I have no idea what the default device will pull back.
Is this approach necessary to establish an RTP session and if so, does the microphone being the capture device have any effect on my ability to send DTMF tones over the media stream?