VoipStreamSender

Package: inet.applications.voipstream

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.

VoipStreamSender

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
socketIn input
socketOut output

Signals

Name Type Unit
packetSent inet::Packet

Statistics

Name Title Source Record Unit Interpolation Mode
packetSent packets sent packetSent 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 IApp
{
    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[packetSent](type=inet::Packet);       // TODO first chunk is VoipStreamPacket
        @statistic[packetSent](title="packets sent"; source=packetSent; record=count,"sum(packetBytes)","vector(packetBytes)"; interpolationmode=none);
        @display("i=block/departure");
    gates:
        input socketIn @labels(UdpControlInfo/up);
        output socketOut @labels(UdpControlInfo/down);
}

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