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Net::SIP::Simple::Call(3) |
User Contributed Perl Documentation |
Net::SIP::Simple::Call(3) |
Net::SIP::Simple::Call - call context for Net::SIP::Simple
my $call = $simple->invite(...);
$call->reinvite(... );
$call->bye();
This package manages the call context for Net::SIP::Simple, e.g. (re-)invites on
existing context etc.
- 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.
- 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.
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