NED File src/inet/applications/voipstream/VoIPStreamSender.ned

Name	Type	Description
VoIPStreamSender	simple module	VoIPStreamSender accepts an audio file and a destination IP address/port as input, and will transmit the file's contents as voice traffic over UDP n times (by default once). For transmission, the audio is resampled at the specified frequency and depth, and encoded with the specified codec at the specified bit rate, and chopped into packets that each carry specified number of milliseconds of voice. Those values come from module parameters. Packets that are all silence (all samples are below a given threshold in absolute value) are transmitted as special "silence" packets. The module does not simulate any particular VoIP protocol (e.g. RTP), but instead accepts a "header size" parameter that can be set accordingly.

Source code

//
// Copyright (C) 2005 M. Bohge ([email protected]), M. Renwanz
// Copyright (C) 2010 Zoltan Bojthe
//
// This program is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public License
// as published by the Free Software Foundation; either version 2
// of the License, or (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
// GNU Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public License
// along with this program; if not, see <http://www.gnu.org/licenses/>.
//


package inet.applications.voipstream;

import inet.applications.contract.IUDPApp;


//
// VoIPStreamSender accepts an audio file and a destination IP address/port as
// input, and will transmit the file's contents as voice traffic over UDP n
// times (by default once). For transmission, the audio is resampled at the
// specified frequency and depth, and encoded with the specified codec at the
// specified bit rate, and chopped into packets that each carry specified
// number of milliseconds of voice. Those values come from module parameters.
// Packets that are all silence (all samples are below a given threshold in
// absolute value) are transmitted as special "silence" packets. The module
// does not simulate any particular VoIP protocol (e.g. RTP), but instead
// accepts a "header size" parameter that can be set accordingly.
//
simple VoIPStreamSender like IUDPApp
{
    parameters:
        int localPort;
        int destPort;
        string destAddress;
        double startTime @unit(s) = default(0s);
        int voipHeaderSize @unit(B);
        int voipSilenceThreshold;
        int sampleRate @unit(Hz) = default(8000Hz);
        string codec = default("g726");         // used by ffmpeg::avcodec_find_encoder_by_name()
        int compressedBitRate @unit(bps) = default(40kbps);
        double packetTimeLength @unit(s) = default(20ms);
        string soundFile;                       // file name of input audio file
        int repeatCount = default(1);
        string traceFileName = default("");     // file name to save output stream (wav), OFF when empty
        @signal[sentPk](type=inet::VoIPStreamPacket);
        @statistic[sentPk](title="packets sent"; source=sentPk; record=count,"sum(packetBytes)","vector(packetBytes)"; interpolationmode=none);
        @display("i=block/departure");
    gates:
        input udpIn;
        output udpOut;
}