Call Control Module

Dan Pascu

   <dan@ag-projects.com>

Edited by

Dan Pascu

   <dan@ag-projects.com>

   Copyright  2005-2008 Dan Pascu
   Revision History
   Revision $Revision: 1 $ $Date: 2008-12-17 13:05:29 +0200 (Wed,
                           17 Dec 2008) $
     __________________________________________________________

   Table of Contents

   1. Admin Guide

        1.1. Overview
        1.2. Description
        1.3. Features
        1.4. Dependencies

              1.4.1. OpenSIPS Modules
              1.4.2. External Libraries or Applications

        1.5. Exported parameters

              1.5.1. disable (int)
              1.5.2. socket_name (string)
              1.5.3. socket_timeout (int)
              1.5.4. signaling_ip_avp (string)
              1.5.5. canonical_uri_avp (string)
              1.5.6. diverter_avp_id (string)

        1.6. Exported Functions

              1.6.1. call_control()

   List of Examples

   1.1. Setting the disable parameter
   1.2. Setting the socket_name parameter
   1.3. Setting the socket_timeout parameter
   1.4. Setting the signaling_ip_avp parameter
   1.5. Setting the canonical_uri_avp parameter
   1.6. Setting the diverter_avp_id parameter
   1.7. Using the call_control function

Chapter 1. Admin Guide

1.1. Overview

   This module allows one to limit the duration of calls and
   automatically end them when they exceed the imposed limit. Its
   main use case is to implement a prepaid system, but it can also
   be used to impose a global limit on all calls processed by the
   proxy.

1.2. Description

   Callcontrol consists of 3 components:
     * The OpenSIPS call_control module
     * An external application called callcontrol which keeps
       track of the calls that have a time limit and automatically
       ends them when they exceed it. This application receives
       requests from OpenSIPS and makes requests to a rating
       engine (see below) to find out if a call needs to be
       limited or not. When a call ends (or is ended) it will also
       instruct the rating engine to debit the balance for the
       caller with the consumed amount. The callcontrol
       application is available from
       http://callcontrol.ag-projects.com/
     * A rating engine that is used to calculate the time limit
       based on the caller's credit and the destination price and
       to debit the caller's balance after a call ends. This is
       available as part of CDRTool from
       http://cdrtool.ag-projects.com/

   The callcontrol application runs on the same machine as
   OpenSIPS and they communicate over a filesystem socket, while
   the rating engine can run on a different host and communicates
   with the callcontrol application using a TCP connection.

   Callcontrol is invoked by calling the call_control() function
   for the initial INVITE of every call we want to apply a limit
   to. This will end up as a request to the callcontrol
   application, which will interogate the rating engine for a time
   limit for the given caller and destination. The rating engine
   will determine if the destination has any associated cost and
   if the caller has any credit limit and if so will return the
   amount of time he is allowed to call that destination.
   Otherwise it will indicate that there is no limit associated
   with the call. If there is a limit, the callcontrol application
   will retain the session and attach a timer to it that will
   expire after the given time causing it to call back to OpenSIPS
   with a request to end the dialog. If the rating engine returns
   that there is no limit for the call, the session is discarded
   by the callcontrol application and it will allow it to go
   proceed any limit. An appropriate response is returned to the
   call_control module that is then returned by the call_control()
   function call and allows the script to make a decision based on
   the answer.

1.3. Features

     * Very simple API consisting of a single function that needs
       to be called once for the first INVITE of every call. The
       rest is done automatically in the background using dialog
       callbacks.
     * Gracefully end dialogs when they exceed their time by
       triggering a dlg_end_dlg request into the dialog module,
       that will generate two BYE messages towards each endpoint,
       ending the call cleanly.
     * Allow parallel sessions using one balance per subscriber
     * Integrates with mediaproxy's ability to detect when a call
       does timeout sending media and is closed. In this case the
       dlg_end_dlg that is triggered by mediaproxy will end the
       callcontrol session before it reaches the limit and
       consumes all the credit for a call that died and didn't
       actually take place. For this mediaproxy has to be used and
       it has to be started by engage_media_proxy() to be able to
       keep track of the call's dialog and end it on timeout.
       Even when mediaproxy is unable to end the dialog because it
       was not started with engage_media_proxy(), the callcantrol
       application is still able to detect calls that did timeout
       sending media, by looking in the radius accounting records
       for entries recorded by mediaproxy for calls that did
       timeout. These calls will also be ended gracefully by the
       callcontrol application itself.

1.4. Dependencies

1.4.1. OpenSIPS Modules

   The following modules must be loaded before this module:
     * dialog module

1.4.2. External Libraries or Applications

   The following libraries or applications must be installed
   before running OpenSIPS with this module loaded:
     * None.

1.5. Exported parameters

1.5.1. disable (int)

   Boolean flag that specifies if callcontrol should be disabled.
   This is useful when you want to use the same OpenSIPS
   configuration in two different context, one using callcontrol,
   the other not. In the case callcontrol is disabled, calls to
   the call_control() function will return a code indicating that
   there is no limit associated with the call, allowing the use of
   the same configuration without changes.

   Default value is "0".

   Example 1.1. Setting the disable parameter
...
modparam("call_control", "disable", 1)
...

1.5.2. socket_name (string)

   It is the path to the filesystem socket where the callcontrol
   application listens for commands from the module.

   Default value is "/var/run/callcontrol/socket".

   Example 1.2. Setting the socket_name parameter
...
modparam("call_control", "socket_name", "/var/run/callcontrol/socket")
...

1.5.3. socket_timeout (int)

   How much time (in milliseconds) to wait for an answer from the
   callcontrol application.

   Default value is "500" (ms).

   Example 1.3. Setting the socket_timeout parameter
...
modparam("call_control", "socket_timeout", 500)
...

1.5.4. signaling_ip_avp (string)

   Specification of the AVP which holds the IP address from where
   the SIP signaling originated. If this AVP is set it will be
   used to get the signaling IP address, else the source IP
   address from where the SIP message was received will be used.
   This AVP is meant to be used in cases where there are more than
   one proxy in the call setup path and the proxy that actually
   starts callcontrol doesn't receive the SIP messages directly
   from the UA and it cannot determine the NAT IP address from
   where the signaling originated. In such a case attaching a SIP
   header at the first proxy and then copying that header's value
   into the signaling_ip_avp on the proxy that starts callcontrol
   will allow it to get the correct NAT IP address from where the
   SIP signaling originated.

   This is used by the rating engine which finds the rates to
   apply to a call based on caller's SIP URI, caller's SIP domain
   or caller's IP address (whichever yields a rate forst, in this
   order).

   Default value is "$avp(s:signaling_ip)".

   Example 1.4. Setting the signaling_ip_avp parameter
...
modparam("call_control", "signaling_ip_avp", "$avp(s:signaling_ip)")
...

1.5.5. canonical_uri_avp (string)

   Specification of the AVP which holds an optional application
   defined canonical request URI. When this is set, it will be
   used as the destination when computing the call price,
   otherwise the request URI will be used. This is useful when the
   username of the ruri needs to have a different, canonical form
   in the rating engine computation than it has in the ruri.

   Default value is "$avp(s:can_uri)".

   Example 1.5. Setting the canonical_uri_avp parameter
...
modparam("call_control", "canonical_uri_avp", "$avp(s:can_uri)")
...

1.5.6. diverter_avp_id (string)

   Specification of the id of an integer AVP which holds an
   optional application defined diverter SIP URI. When this is
   set, it will be used by the rating engine as the billing party
   when finding the rates to apply to a given call, otherwise, the
   caller's URI taken from the From field will be used. When set,
   this AVP should contain a value in the form "user@domain" (no
   sip: prefix should be used).

   This is useful when a destination diverts a call, thus becoming
   the new caller. In this case the billing party is the diverter
   and this AVP should be set to it, to allow the rating engine to
   pick the right rates for the call. For example, if A calls B
   and B diverts all its calls unconditionally to C, then the
   diverter AVP should the set to B's URI, because B is the
   billing party in the call not A after the call was diverted.

   Default value is "805".

   Example 1.6. Setting the diverter_avp_id parameter
...
modparam("call_control", "diverter_avp_id", 805)

route {
  ...
  # alice@example.com is paying for this call
  $avp(i:805) = "sip:alice@example.com";
  ...
}
...

1.6. Exported Functions

1.6.1. call_control()

   Trigger the use of callcontrol for the dialog started by the
   INVITE for which this function is called (the function should
   only be called for the first INVITE of a call). Further
   in-dialog requests will be processed automatically using
   internal bindings into the dialog state machine, allowing
   callcontrol to update its internal state as the dialog
   progresses, without any other intervention from the script.

   This function should be called right before the message is sent
   out using t_relay(), when all the request uri modifications are
   over and a final destination has been determined.

   This function has the following return codes:

     * +2 - call has no limit
     * +1 - call has limit and is traced by callcontrol
     * -1 - not enough credit to make the call
     * -2 - call is locked by another call in progress
     * -5 - internal error (message parsing, communication, ...)

   This function can be used from REQUEST_ROUTE.

   Example 1.7. Using the call_control function

...
if (is_avp_set("$avp(i:805)")) {
    # the diverter AVP is set, use it as billing party
    $avp(s:billing_party_domain) = $(avp(i:805){uri.domain});
} else {
    $avp(s:billing_party_domain) = $fd;
}

if (method==INVITE && !has_totag() &&
    is_domain_local("$avp(s:billing_party_domain)")) {
    call_control();
    switch ($retcode) {
    case 2:
        # Call with no limit
    case 1:
        # Call has limit and is under callcontrol management
        break;
    case -1:
        # Not enough credit (prepaid call)
        sl_send_reply("402", "Not enough credit");
        exit;
        break;
    case -2:
        # Locked by another call in progress (prepaid call)
        sl_send_reply("403", "Call locked by another call in progress");
        exit;
        break;
    default:
        # Internal error (message parsing, communication, ...)
        if (PREPAID_ACCOUNT) {
            xlog("Call control: internal server error\n");
            sl_send_reply("500", "Internal server error");
            exit;
        } else {
            xlog("L_WARN", "Cannot set time limit for postpaid call\n");
        }
    }
}
t_relay();
...
