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
timeToLive	int	-1	if not -1, set the TTL (IPv4) or Hop Limit (IPv6) field of sent packets to this value
dscp	int	-1	if not -1, set the DSCP field (on IPv4/IPv6) of sent packets to this value
tos	int	-1	if not -1, set the Type Of Service (IPv4) / Traffic Class (IPv6) field of sent packets to this value

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

Scheduled messages (observed)

msg	kind	ctrl	tags	msgname	context
Packet	0			SILENCE
Packet	0			VOICE
omnetpp::cMessage	0			sendVoIP

Direct method calls (observed)

call to	function	info
MessageDispatcher	inet::MessageDispatcher::arrived	arrived

Outgoing messages (observed)

gate	msg	kind	ctrl	destModule	tags
socketOut	Packet	0 (UdpCommandCode::UDP_C_DATA)		Udp	DispatchProtocolReq, L3AddressReq, L4PortReq, SocketReq
socketOut	Request	1 (UdpCommandCode::UDP_C_BIND)	UdpBindCommand	Udp	DispatchProtocolReq, SocketReq

Packet operations (observed)

chunkType	packetAction
BytesChunk	insertAtFront
VoipStreamPacket	insertAtFront

Shared Tagging operations (observed)

tagType	tagAction
DispatchProtocolReq	addTagIfAbsent
L3AddressReq	addTagIfAbsent
L4PortReq	addTagIfAbsent
SocketReq	addTagIfAbsent

Tagging operations (observed)

tagType	tagAction
inet::Ipv4InterfaceData	findTag

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
        int timeToLive = default(-1); // if not -1, set the TTL (IPv4) or Hop Limit (IPv6) field of sent packets to this value
        int dscp = default(-1); // if not -1, set the DSCP field (on IPv4/IPv6) of sent packets to this value
        int tos = default(-1); // if not -1, set the Type Of Service (IPv4) / Traffic Class (IPv6) field of sent packets to this value
        @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