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.

Inheritance diagram

The following diagram shows inheritance relationships for this type. Unresolved types are missing from the diagram.

Parameters

Name	Type	Default value	Description
localPort	int
destPort	int
destAddress	string
startTime	double	0s
voipHeaderSize	int
voipSilenceThreshold	int
sampleRate	int	8000Hz
codec	string	"g726"	used by ffmpeg::avcodec_find_encoder_by_name()
compressedBitRate	int	40kbps
packetTimeLength	double	20ms
soundFile	string		file name of input audio file
repeatCount	int	1
traceFileName	string	""	file name to save output stream (wav), OFF when empty

Properties

Name	Value	Description
display	i=block/departure

Gates

Name	Direction	Size	Description
udpIn	input
udpOut	output

Signals

Name	Type	Unit
sentPk	inet::VoIPStreamPacket

Statistics

Name	Title	Source	Record	Unit	Interpolation Mode
sentPk	packets sent	sentPk	count, sum(packetBytes), vector(packetBytes)		none

Source code

//
// 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;
}

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