Steffen Ullrich > Net-SIP-0.67 > Net::SIP::Simple::Call

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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

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].

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.

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.

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 accessable 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.

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