Net::SIP::DTMF(3) DTMF RTP packet generating and extracting

SYNOPSIS


use Net::SIP::DTMF;
my $sub = dtmf_generator(
11, # event '#'
100, # duration 100ms
rfc2833_type => 101, # RTP type 101 for telephone-event/8000
# audio_type => 0, # RTP type 0 for PCMU/8000
);
while (...) {
my @pkt = $sub->($sequence,$timestamp,$srcid);
last if ! @pkt; # dtmf done
next if $buf[0] eq ''; # more coming, but no data this time (pause)
.. send @pkts ...
}
use Net::SIP::DTMF;
my $sub = dtmf_extractor(
rfc2833_type => 101, # RTP type 101 for telephone-event/8000
audio_type => 0, # RTP type 0 for PCMU/8000
);
while (...) {
if ( my ($event,$duration,$type) = $sub->($packet)) {
# event received
...
}
}

DESCRIPTION

This package provides functions for generating RTP packets containing DTMF events or extracting DTMF events from RTP packets.

FUNCTIONS

dtmf_generator ( EVENT, DURATION, %ARGS )
Generates a function which is used to generate packets for the given EVENT.

EVENT is the event numer (0..15) or undef if it should just generate silence or pause. DURATION is the time for the event in ms. ARGS contain information how the event should be packed: either as RFC2833 RTP event or as RTP audio PCMU/8000. %ARGS can be

rfc2833_type => rtp_type
Gives the RTP payload type number for rfc2833 RTP events (telephone-event/8000). If not given will try audio_type instead.
audio_type => rtp_type
Gives the RTP payload type number for PCMU/8000. If not given and no rfc2833_type given it will croak.
volume => volume
Sets the volume for RTP event according to rfc2833.

The generated function should then be called with SEQUENCE,TIMESTAMP,SRCID to generate the RTP packets and will return @RTP_PACKETS, with

() - DTMF event is finished
$RTP_PACKETS[0] eq '' - no RTP packet for this call (pause)
@RTP_PACKETS - RTP packets which can be send to the peer
dtmf_extractor ( %ARGS )
Generates a function which is used to extract DTMF events. Keys rfc2833_type and audio_type have the same meaning as in generate_dtmf. It will only attempt to extract DTMF events from rfc2833 RTP events or audio if the relevant rtp_type is given.

The function will be called with the RTP packet as the only argument and will return () if no new events where found or (EVENT,DURATION,TYPE) if an event finished, where DURATION is the duration in ms and TYPE is audio|rfc2833.

For performance reasons it is best to use only rfc2833 if the peer supports it.