Provided by: libnet-sip-perl_0.838-1_all 
NAME
Net::SIP::Simple::Call - call context for Net::SIP::Simple
SYNOPSIS
my $call = $simple->invite(...);
$call->reinvite(... );
$call->bye();
DESCRIPTION
This package manages the call context for Net::SIP::Simple, e.g. (re-)invites on existing context etc.
CONSTRUCTOR
new ( CONTROL, CTX, \%ARGS )
Creates a new Net::SIP::Simple::Call object to control a call. Usually called from invite in
Net::SIP::Simple.
CONTROL is the Net::SIP::Simple object managing the calls.
CTX is either an existing 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 $call if called as "$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:
leg Specifies which leg should be used for the call (default is first leg in dispatcher).
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.
init_media
Callback used to initialize media for the connection, see method rtp in Net::SIP::Simple and
Net::SIP::Simple::RTP.
Callback will be invoked with the call $self and the connection parameter as an argument (as
hash reference).
rtp_param
Data for the codec used in the media specified by init_media and for the initialization of
the default SDP data. This is an array reference "[pt,size,interval,name]" where pt is the
payload type, size is the size of the payload and interval the interval in which the RTP
packets will be send. 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: "[0,160,160/8000]". An alternative would be for example "[97,50,0.03,'iLBC/8000']"
for iLBC.
sdp 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.
sdp_peer
Holds the Net::SIP::SDP body send by the peer. Usually not set in the constructor but can be
accessed from callbacks.
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 sdp is provided this parameter has to be provided too, e.g. the package will not allocate
the sockets described in the SDP packet.
media_ssocks
Sockets used for sending RTP data. If not given the socket for sending RTP is the same as for
receiving RTP, unless asymetric_rtp is specified.
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.
dtmf_methods
If a DTMF callback is specified this is treated as a list of supported DTMF methods for
receiving DTMF. If not given it defaults to 'rfc2833,audio'.
recv_bye
Callback usable by invoke_callback in 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.
send_bye
Callback usable by invoke_callback in 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 bye method.
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.
cb_final
Callback usable by invoke_callback in 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 "( STATUS, SELF, %INFO )" where STATUS is either 'OK' or 'FAIL'
('OK' if final response meant success, else 'FINAL'), and %INFO contains more information,
like "( packet => packet )" for the packet containing the final answer or "( code =>
response_code )" in case failures caused by an unsuccessful response.
cb_preliminary
Callback usable by invoke_callback in Net::SIP::Util which will be invoked, when it received
a preliminary response on locally created INVITE.
Callback will be invoked with "( SELF, CODE, RESPONSE )" where CODE is the response code and
RESPONSE the Net::SIP::Response packet.
cb_established
Callback usable by invoke_callback in Net::SIP::Util which will be invoked, when it received
the final answer on locally created INVITE requests.
Callback will be invoked with "( 'OK', SELF )".
cb_invite
Callback usable by invoke_callback in Net::SIP::Util which will be invoked, when it received
an INVITE request
Callback will be invoked with "( SELF, REQUEST )" where REQUEST is the Net::SIP::Request
packet for the INVITE. If it returns a Net::SIP::Packet this will be used as response,
otherwise a default response with code 200 will be created.
cb_dtmf Callback usable by invoke_callback in Net::SIP::Util which will be invoked, when it received
an DTMF event.
Callback will be invoked with "( 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 init_media. All
builtin handlers from 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.
cb_notify
Callback usable by invoke_callback in Net::SIP::Util which will be invoked, when it received
an NOTIFY request
Callback will be invoked with "( SELF, REQUEST )" where REQUEST is the Net::SIP::Request
packet for the NOTIFY.
sip_header
A reference to a hash with additional SIP headers for the INVITE requests.
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
set_param or within reinvite each time the call should be put on hold.
... More parameters may be specified and are accessible from the callbacks. For instance
media_send_recv in Net::SIP::Simple::RTP uses a parameter cb_rtp_done. See there.
METHODS
cleanup
Will be called to clean up the call. Necessary because callbacks etc can cause cyclic references
which need to be broken. Calls rtp_cleanup too. Works by invoking all callbacks which are stored as
\@list in "$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.
rtp_cleanup
Cleanup of current RTP connection. Works be invoking all callbacks which are stored as \@list in
"$self->{rtp_cleanup}" (these callbacks are inserted by Net::SIP::Simple::RTP etc).
get_peer
Returns peer of call, see peer in Net::SIP::Endpoint::Context.
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 resp40x an 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 cb_final once the final response for the
INVITE was received and init_media if this response was successful.
If no 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 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 cb_noanswer will be called after the CANCEL
was delivered (or delivery failed).
Returns the connection context as Net::SIP::Endpoint::Context object.
This method is called within invite in Net::SIP::Simple after creating the new 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 Net::SIP::Simple::Call object $self.
cancel ( %ARGS )
Closes a pending call by sending a CANCEL request. Returns true if call was pending and could be
canceled.
If %ARGS contains 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 cb_final callback (as hash reference).
bye ( %ARGS )
Closes a call by sending a BYE request. If %ARGS contains 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 cb_final
callback (as hash reference).
request ( METHOD, BODY, %ARGS )
Will create a request with METHOD and BODY and wait for completion. If %ARGS contains cb_final it
will be used as a callback and invoked once it gets the response for the request (or timeout). The
rest of %ARGS will be used to create request (mostly for request header, see
Net::SIP::Endpoint::new_request)
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 duration in ms (default 100ms) and a callback cb_final which is invoked
with first argument 'OK', when all events are send. If no 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 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 init_media. All builtin
handlers from 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.
receive ( ENDPOINT, CTX, ERROR, CODE, PACKET, LEG, FROM )
Will be called from the dispatcher on incoming packets. ENDPOINT is the Net::SIP::Endpoint object
which manages the 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 Net::SIP::Leg where it came in
and FROM the "ip:port" of the sender. For more details see documentation to set_callback in
Net::SIP::Endpoint::Context.
If the incoming packet is a BYE or CANCEL request it will close the call and invoke the 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 cb_established callback and also the init_media callback which cares
about setting up the RTP connections (e.g produce and accept RTP traffic).
set_param ( %ARGS )
Changes param like init_media, sdp_on_ack on the current call. See the constructor. This is useful if
call consists of multiple invites with different features.
get_param ( @KEYS )
Returns values for parameter @KEYS, pendant to 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.
perl v5.40.0 2024-09-08 Net::SIP::Simple::Call(3pm)