
=head1 NAME

Net::SIP::Simple::Call - call context for L<Net::SIP::Simple>

=head1 SYNOPSIS

  my $call = $simple->invite(...);
  $call->reinvite(... );
  $call->bye();

=head1 DESCRIPTION

This package manages the call context for L<Net::SIP::Simple>,
e.g. (re-)invites on existing context etc.

=head1 CONSTRUCTOR

=over 4

=item new ( CONTROL, CTX, \%ARGS )

Creates a new L<Net::SIP::Simple::Call> object to control a call.
Usually called from B<invite> in L<Net::SIP::Simple>.

CONTROL is the L<Net::SIP::Simple> object managing the calls.

CTX is either an existing L<Net::SIP::Endpoint::Context> or the
SIP address of the peer which will be contacted in this call
or a hash which can be used to create the context. 
If no complete context is given missing information will be taken
from C<$call> if called as C<$call->new>.

%ARGS are used to describe the behavior of the call and will be
saved in the object as the connection parameter. The following
options are used in the connection parameter and can be given
in %ARGS:

=over 8

=item leg

Which leg should be used for the call (default is first leg in dispatcher).

=item sdp_on_ack

If given and TRUE it will not send the SDP body on INVITE request,
but on ACK. Mainly used for testing behavior of proxies in between
the two parties.

=item init_media

Callback used to initialize media for the connection, see method B<rtp>
in L<Net::SIP::Simple> and L<Net::SIP::Simple::RTP>.

Callback will be invoked with the call C<$self> and the connection
parameter as an argument (as hash reference).

=item rtp_param

Data for the codec used in the media specified by B<init_media> and
for the initialization of the default SDP data. This is an array reference
C<< [pt,size,interval,name] >> where B<pt> is the payload type, B<size> is the
size of the payload and B<interval> the interval in which the RTP packets
will be send. B<name> is optional and if given rtpmap and ptime entries will be added
to the SDP so that the name is associated with the given payload type.
The default is for PCMU/8000: C<< [0,160,160/8000] >>.

=item sdp

L<Net::SIP::SDP> object or argument for constructing this object.
If not given  it will create an SDP body with one RTP audio connection
unless it got first SDP data from the peer in which case it simply
matches them.

=item sdp_peer

Holds the L<Net::SIP::SDP> body send by the peer. Usually not set in
the constructor but can be accessed from callbacks.

=item media_lsocks

Contains a \@list of sockets for each media-line in the SDP. Each
item in this list is either a single socket (in case of port range 1)
or a \@list of sockets.

If B<sdp> is provided this parameter has to be provided too, e.g.
the package will not allocate the sockets described in the SDP packet.

=item media_ssocks

Sockets used for sending RTP data. If not given the socket for
sending RTP is the same as for receiving RTP, unless B<asymetric_rtp>
is specified.

=item asymetric_rtp

By default it will send the RTP data from the same port where it listens
for the data. If this option is TRUE it will allocate a different port
for receiving data. Mainly used for testing behavior of proxies in between
the two parties.

=item recv_bye

Callback usable by B<invoke_callback> in L<Net::SIP::Util> which will be
invoked, when the peer initiated the close of the connection using BYE
or CANCEL.

Argument for the callback will be a hash reference containing the connection
parameter.

=item send_bye

Callback usable by B<invoke_callback> in L<Net::SIP::Util> which will be
invoked, when the local side initiated the close of the connection using BYE
or CANCEL.

Argument for the callback will be a hash reference containing the connection
parameter merged with the parameter from the B<bye> method.

=item clear_sdp

If TRUE the keys media_lsocks, media_ssocks, sdp and sdp_peer will be cleared
on each new (re)INVITE request, so that it will allocate new sockets for RTP
instead of reusing the existing.

=item cb_final

Callback usable by B<invoke_callback> in L<Net::SIP::Util> which will be
invoked, when it received the final answer on locally created INVITE requests
(e.g. when it established the call by sending the ACK).

Callback will be invoked with C<< ( STATUS, SELF, %INFO ) >> where STATUS
is either 'OK' or 'FAIL' ('OK' if final response meant success, else 'FINAL'),
and %INFO contains more information, like C<< ( packet => packet ) >> for the 
packet containing the final answer or C<< ( code => response_code ) >> in case 
failures caused by an unsuccessful response.

=item cb_preliminary

Callback usable by B<invoke_callback> in L<Net::SIP::Util> which will be
invoked, when it received a preliminary response on locally created INVITE.

Callback will be invoked with C<< ( SELF, CODE, RESPONSE ) >> where
CODE is the response code and RESPONSE the L<Net::SIP::Response> packet.

=item cb_established

Callback usable by B<invoke_callback> in L<Net::SIP::Util> which will be
invoked, when it received the final answer on locally created INVITE requests.

Callback will be invoked with C<< ( 'OK', SELF ) >>.

=item cb_invite

Callback usable by B<invoke_callback> in L<Net::SIP::Util> which will be
invoked, when it received an INVITE request

Callback will be invoked with C<< ( SELF, REQUEST ) >> where REQUEST
is the L<Net::SIP::Request> packet for the INVITE.

=item cb_dtmf

Callback usable by B<invoke_callback> in L<Net::SIP::Util> which will be
invoked, when it received an DTMF event.

Callback will be invoked with C<< ( EVENT, DURATION ) >> where EVENT
is the event ([0-9A-D*#]) and DURATION the duration in ms.

Receiving DTMF needs to be supported by the active RTP handler set with 
B<init_media>. All builtin handlers from L<Net::SIP::Simple::RTP> are supported.
If no RTP handler is set up or if the RTP handler does not support DTMF
sending no DTMF will be received without any warning.

=item sip_header

A reference to a hash with additional SIP headers for the INVITE requests.

=item call_on_hold

This option causes the next SDP to have 0.0.0.0 as it's address to put
this side of the call on hold (will not receive data). This is a one-shot
option, e.g. needs to be set with B<set_param> or within B<reinvite>
each time the call should be put on hold.

=item ...

More parameters may be specified and are accessable from the callbacks.
For instance B<media_send_recv> in L<Net::SIP::Simple::RTP> uses a parameter
B<cb_rtp_done>. See there.

=back

=back

=head1 METHODS

=over 4

=item cleanup

Will be called to clean up the call. Necessary because callbacks etc can cause
cyclic references which need to be broken.
Calls B<rtp_cleanup> too.
Works by invoking all callbacks which are stored as \@list in C<< $self->{call_cleanup} >>.

This will called automatically at a clean end of a call (e.g. on BYE or CANCEL, either
issued locally or received from the peer). If there is not clean end and one wants
to destroy the call unclean one need to call this method manually.

=item rtp_cleanup

Cleanup of current RTP connection.
Works be invoking all callbacks which are stored as \@list in C<< $self->{rtp_cleanup} >>
(these callbacks are inserted by L<Net::SIP::Simple::RTP> etc).

=item get_peer

Returns peer of call, see B<peer> in L<Net::SIP::Endpoint::Context>.

=item reinvite ( %ARGS )

Creates a INVITE request which causes either the initial SDP session or an update of
the SDP session (reinvite). %ARGS will merged with the connection parameter, see description
on the constructor. 
Additionally using B<resp40x> an B<auth> as a parameter here would make sense 
if you want to habe full control about the authorization process.

Sets up callback for the connection, which will invoke B<cb_final> once the final
response for the INVITE was received and B<init_media> if this response was
successful.

If no B<cb_final> callback was given it will wait in the event loop until a final
response was received. Only in this case it will also use the param
B<ring_time> which specifies the time it will wait for a final response. If
no final response came in within this time it will send a CANCEL request for
this call to close it. In this case a callback specified with B<cb_noanswer>
will be called after the CANCEL was delivered (or delivery failed).

Returns the connection context as L<Net::SIP::Endpoint::Context> object.

This method is called within B<invite> in L<Net::SIP::Simple> after creating the new
L<Net::SIP::Simple::Call> object to create the first SDP session. Changes on the
SDP session will be done by calling this method on the L<Net::SIP::Simple::Call>
object C<$self>.

=item cancel ( %ARGS )

Closes a pending call by sending a CANCEL request.
Returns true if call was pending and could be canceled.

If %ARGS contains B<cb_final> it will be used as a callback and invoked once it gets
the response for the CANCEL (which might be a response packet or a timeout).
The rest of %ARGS will be merged with the connection parameter and given as an argument
to the B<cb_final> callback (as hash reference).

=item bye ( %ARGS )

Closes a call by sending a BYE request.
If %ARGS contains B<cb_final> it will be used as a callback and invoked once it gets
the response for the BYE (which might be a response packet or a timeout).
The rest of %ARGS will be merged with the connection parameter and given as an argument
to the B<cb_final> callback (as hash reference).

=item dtmf ( EVENTS, %ARGS )

Sends DTMF (dial tones) events to peer according to RFC2833 (e.g. as RTP
events). 

EVENTS is a string with the characters 0-9,A-D,*,#. These will be send as DTMF.
Any other characters in the string will lead to a pause in sending DTMF (e.g.
"123--#" will send "1","2,","3", then add to pauses and then send "#"). 

In %ARGS one can specify a B<duration> in ms (default 100ms) and a callback
B<cb_final> which is invoked with first argument 'OK', when all events are send. 
If no B<cb_final> callback is given the method will return only when all events 
are send.

One can also overwrite the automatic detection of the DTMF method using
B<methods> in %ARGS. Default is 'rfc2833,audio', with 'rfc2833' only one 
enforces the use of RTP events, and if the peer does not support it it will
croak. Setting to 'audio' will not fail from the client side, but the peer
might not look for DTMF inband data if it expects RTP events.

Sending DTMF needs to be supported by the active RTP handler set with
B<init_media>. All builtin handlers from L<Net::SIP::Simple::RTP> are supported.
If no RTP handler is set up or if the RTP handler does not support DTMF sending
no DTMF will be received without any warning.

=item receive ( ENDPOINT, CTX, ERROR, CODE, PACKET, LEG, FROM )

Will be called from the dispatcher on incoming packets. ENDPOINT is the
L<Net::SIP::Endpoint> object which manages the L<Net::SIP::Endpoint::Context> CTX
calling context for the current call. ERROR is an errno describing the
error (and 0|undef if no error). CODE is the numerical code from the packet if
a response packet was received. PACKET is the incoming packet,
LEG the L<Net::SIP::Leg> where it came in and FROM the C<< "ip:port" >> of the
sender. For more details see documentation to B<set_callback> in
L<Net::SIP::Endpoint::Context>.

If the incoming packet is a BYE or CANCEL request it will close the call and
invoke the B<recv_bye> callback.

If it is INVITE or ACK it will make sure that the RTP sockets are set up.
If receiving an ACK to the current call it will invoke the B<cb_established>
callback and also the B<init_media> callback which cares about setting up
the RTP connections (e.g produce and accept RTP traffic).

=item set_param ( %ARGS )

Changes param like B<init_media>, B<sdp_on_ack> on the current call. See the
constructor. This is useful if call consists of multiple invites with different
features.

=item get_param ( @KEYS )

Returns values for parameter @KEYS, pendant to B<set_param>
If there is only one key it will return the value as scalar, on multiple
keys it returns an array with all values.

=back
