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

Name	Type	Description
VoIPStreamReceiver	simple module	VoIPStreamReceiver listens on an UDP port, and expects to receive VoIP packets on it. The received voice is then saved into a result audio file that can be compared with the original for further evaluation. VoIP packets are numbered, and out-of-order packets are discarded (the corresponding voice interval will be recorded as silence into the file). VoIP packets that miss their deadlines will similarly be discarded. It is assumed that the audio is played back with delay (by default 20ms), which allows some jitter for the incoming packets. The resulting audio file is closed when the simulation completes (i.e. in the OMNeT++ finish() function). Only one voice session ("call") may be underway at a time.

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;


//
// VoIPStreamReceiver listens on an UDP port, and expects to receive VoIP packets on
// it. The received voice is then saved into a result audio file that can be
// compared with the original for further evaluation. VoIP packets are
// numbered, and out-of-order packets are discarded (the corresponding voice
// interval will be recorded as silence into the file). VoIP packets that miss
// their deadlines will similarly be discarded. It is assumed that the audio is
// played back with delay (by default 20ms), which allows some jitter for the
// incoming packets. The resulting audio file is closed when the simulation
// completes (i.e. in the OMNeT++ finish() function). Only one voice session
// ("call") may be underway at a time.
//
simple VoIPStreamReceiver like IUDPApp
{
    parameters:
        int localPort;
        double playoutDelay @unit(s) = default(20ms);
        string resultFile;
        @signal[rcvdPk](type=cPacket); // expected type=VoIPStreamPacket
        @signal[dropPk](type=cPacket);
        @signal[lostSamples](type=long);
        @signal[lostPackets](type=long);
        @signal[packetHasVoice](type=long);  // 1=yes, 0=no
        @signal[connState](type=long);  // 1=open, -1=close
        @signal[delay](type=simtime_t);  // total time lag of voice transmission
        @statistic[rcvdPk](title="packets received"; source=rcvdPk; record=count,"sum(packetBytes)","vector(packetBytes)"; interpolationmode=none);
        @statistic[dropPk](title="packets dropped"; source=dropPk; record=count,"sum(packetBytes)","vector(packetBytes)"; interpolationmode=none);
        @statistic[lostSamples](title="lost samples"; interpolationmode=none; record=vector,stats);
        @statistic[lostPackets](title="lost packets"; interpolationmode=none; record=vector,stats);
        @statistic[packetHasVoice](title="voice or silence packet"; record=mean,vector; interpolationmode=none);
        @statistic[delay](title="delay"; unit=s; record=vector,histogram; interpolationmode=none);
        @statistic[numActiveSessions](title="number of active sessions"; source="sum(connState)"; record=max,timeavg,vector; interpolationmode=sample-hold; );
        @statistic[numSessions](title="total number of sessions"; source="sum(connState+1)/2"; record=last);
        @display("i=block/arrival");
    gates:
        input udpIn;
        output udpOut;
}