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first of all, thanks to your great work.
Im currently using the library from peers in a software project and it works good so far. I am able to initiate a phonecall and so on.
My question now: Is it possible to send an audio file to the callee after he picked up the phonecall. I don't want to initiate a usual phonecall. It should just play the audio file, and wait till the callee hung up.
I tried to figure out how to get the audio file into a RTP Packet and then how to send this packet to the callee after he picked up the phonecall. But i m stuck here. I could not find an option to send the RTP packet.
Any hint or information would be great to get this to work.
Thank you for your time and best regards
you should replace access to net.sourceforge.peers.media.SoundManager with a FileManager or something like that; This file should be encoded as raw data with the same format as microphone (linear PCM 8kHz, 16 bits signed, mono-channel, little endian). You can transcode your file using audacity and "export data". Thus you will just have to replace SoundManager with FileManager and read raw data from file in this new class.
Let me know if it's not clear.
Thanks a lot, for your quick reply.
I guess this should be clear now :) I ll give it a try and let you know how it worked.
I too get this problem. every time i am getting some noise only. in my project i am using freeTTS for generating speech.
I had a similar issue a couple of days back. As yohannmartineau correctly specifies, it must be linear PCM 8kHz, 16 bits signed, mono-channel, little endian. FreeTTS sends you big endian.
Fix that and you will be fine :-)