VoIP RTP Streaming from/to server (in Java) to/from android

My target is to have a push-to-talk voice app in GSM/UMTS/LTE networks; initially I wanted use multicast addresses and peer-to-peer without overload the server; unfortunatly, after deep investigation, I discovered that multicast is not allowed in GSM/UMTS/LTE networks, therefore I have to use a central server in order to bounce the VoIP packets. I have to send VoIP from 1 android client to a server (PC), and the same VoIP from the server to N android clients. The server is in Java, it has to receive the VoIP packets from the transmitter Android, and then it has to send the VoIP packets to other N clients; therefore the server is a bouncer of VoIP packets. My app, on both android and server side, is almost ready, what is missing is the code for the VoIP streaming. I need mainly a Java method, to be included in the server java code, that receives the VoIP packets from 1 android client and resend the same VoIP packets to N android clients. The parameters of the method will be: - IP address of android client VoIP sender (the transmitter) - port of android client VoIP sender - arraylist containing the N ip addresses of the N android client VoIP receivers - arraylist containing the N ports of the N android client VoIP receiver Of course the method has to take into account and solve the tipical VoIP issues: delivery speed, FEC, loss concealment, jitter buffering, etc... I also read that there are some technics to have a dynamic jitter buffer so to have dynamically the right balance between quality and latency (in stackoverflow I saw also some code example of dynamic jitter buffer...). The software developper must have deep experience in VoIP and Java, and will have also to suggest me the right/suitable parameters to be used (type of codecs, etc...) for VoIP tuning. Please find below the code (for android) I use to send the VoIP packets from the transmitter android client to the server. It works because I don't have problems when I use it for individual call sending the VoIP packets directly from Android to Android; the code to receive in android the packets from server is very similar, so I don't recopy it. As you can see I use [login to view URL] . You will have to modify this code if needed in order to be adapted to the server code. // ANDROID CODE USED TO SEND VOIP TO SERVER //Attribute definition private static final AudioCodec myAudioCodec_COSTANTE = [login to view URL] ; private static final int myAudioGroupTX_COSTANTE = AudioGroup.MODE_NORMAL ; private static final int myAudioGroupRX_COSTANTE = AudioGroup.MODE_NORMAL ; private static final int myRtpStreamTX_COSTANTE = RtpStream.MODE_SEND_ONLY ; private static final int myRtpStreamRX_COSTANTE = RtpStream.MODE_RECEIVE_ONLY ; private static final int myAudioManagerTX_COSTANTE = AudioManager.MODE_IN_COMMUNICATION; private static final int myAudioManagerRX_COSTANTE = AudioManager.MODE_IN_COMMUNICATION; //Method called for VoIP trasmission myAudioStream = new AudioStream(localClientIP); myAudioGroup = new AudioGroup(); myAudioManager = (AudioManager) [login to view URL](Context.AUDIO_SERVICE); [login to view URL](myAudioGroupTX_COSTANTE); [login to view URL](null); [login to view URL](myAudioCodec_COSTANTE); [login to view URL](myRtpStreamTX_COSTANTE); [login to view URL](ipaddress_Server, port_Server) [login to view URL](myAudioGroup); [login to view URL](myAudioManagerTX_COSTANTE); [login to view URL](false); [login to view URL](false);