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Oracle® Communications Services Gatekeeper Communication Service Guide
Release 5.0

Part Number E21362-01
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2 Parlay X 2.1 Audio Call/SIP

This chapter describes the Parlay X 2.1 Audio Call/Session Initiation Protocol (SIP) communication service in detail.

Overview of the Parlay X 2.1 Audio Call / SIP Communication Service

The Parlay X 2.1 Audio Call/SIP communication service exposes the Parlay X 2.1 Audio Call 2.1 application interfaces.

The communication service connects to a SIP-IMS network using Oracle Converged Application Server. Converged Application Server is collocated with Services Gatekeeper in the network tier.

For the exact version of the standards that the communication service supports for the application-facing interfaces and the network protocols, see the appendix on standards and specifications in Concepts Guide.

Using this communication service, an application can:

The Audio Call communication service can be used by applications to start an audio call using the Parlay X 2.1 Part 11 protocol.

The audio message content to be played must be defined in an audio fileformat stored at a URL available to the network.. Services Gatekeeper does not actually render the message. This is the responsibility of equipment that must be present on the target telecom network.

Audio Call/SIP Plug-in Application Requests

The Audio Call/SIP plug-in uses these requests that applications send to play or manage audio calls:

  • PlayAudioMessage: The application sends the URI of an audio file and the address of the terminal to the network. This request then returns a call correlator to the application and plays the audio file to the terminal.

  • EndMessage: The application send in the correlator of a PlayAudioMessage request to end that call immediately.

  • GetMessageStatus: The application sends in the correlator of a PlayAudioMessage request to get the status of that call. This method returns the status of the call (played, playing, pending, or error.

Audio Call/SIP Plug-in Call Flow

This is a sample Audio Call/SIP call flow:

  1. The application sends a Parlay X 2.1 PlayAudioCall request to the terminal address with a SIP INVITE message.

    This message does not contain a Session Description Protocol (SDP).

  2. The terminal returns a SIP 200 OK message to the application containing a valid SDP.

  3. The application's media server controller then processes the SDP and sends an ACK message back to the terminal address.

Application Interfaces

For information about the SOAP-based interface for the Parlay X 2.1 Audio Call/SIP communication service, see the discussion of Parlay X 2.1 Interfaces in Application Developer's Guide.

For information about the RESTful Audio Call interface, see the discussion of Audio Call in RESTful Application Developer's Guide.

The RESTful Service Short Messaging interfaces provide RESTful access to the same functionality as the SOAP-based interfaces. The internal representations are identical, and for the purposes of creating SLAs, reading CDRs, and so on, they are the same.

Events and Statistics

The Parlay X 2.1 Audio Call/SIP communication service generates Event Data Records (EDRs), Charging Data Records (CDRs), alarms, and statistics to assist system administrators and developers in monitoring the service.

For general information, see Appendix A, "Events, Alarms, and Charging."

Event Data Records

Table 2-1 lists the IDs of the EDRs created by the Parlay X 2.1 Audio Call/SIP communication service. This does not include EDRs created when exceptions are thrown.

Table 2-1 Event Types Generated by Parlay X 2.1 Audio Call/SIP

EDR ID Method Called







Charging Data Records

Audio Call/SIP-specific CDRs are generated under the following conditions:

  • When callConnected is sent from the network to Services Gatekeeper, indicating that the audio message has connected to the terminal. The CDR number is 405001; the class is

  • When callReleased is sent from the network to Services Gatekeeper, indicating that the audio message has completed playing. The CDR number is 405002; the class is


Table 2-2 maps methods invoked from either the application or the network to the transaction types collected by the Services Gatekeeper statistics counters.

Table 2-2 Methods and Transaction Types for Parlay X 2.1 Audio CalI/SIP

Method Transaction Type




For the list of alarms, see Alarm Handling Guide.

Managing Parlay X 2.1 Audio Call / SIP

This section describes the properties and workflow for setting up the Parlay X 2.1 Audio Call/SIP plug-in instance.

The Parlay 2.1 Audio Call/SIP plug-in supports sending an audio fileto a single terminal.

The Audio Call/SIP plug-in is usable with high availability systems only if your media server supports clustering. See your media server documentation for details.

This plug-in service does not support multiple instantiation using the Plug-in Manager. You can create only one instance by using the Plug-in Manager. The plug-in instance is not automatically created when the plug-in service is started.

Properties for Parlay X 2.1 Audio Call/SIP

Table 2-3 lists the technical specifications for the communication service.

Table 2-3 Properties for Parlay X 2.1 Audio CalI/SIP

Property Description

Managed object in Administration Console

domain_name > OCSG > server_name > Communication Services > plugin_instance_id






Network protocol plug-in service ID


Network protocol plug-in instance ID

The ID is assigned when the plug-in instance is created. See "Managing and Configuring the Plug-in Manager" in System Administrator's Guide.

Supported Address Schemes

tel, sip

Application-facing interfaces


Service type


Exposes to the service communication layer a Java representation of:

Parlay X 2.1 Part 11: Audio Call

Interfaces with the network nodes using:

RFC 3261.

Deployment artifacts

Application-facing: wlng_at_audio_call_px21.ear

Network-facing: wlng_nt_audio_call_px21.ear

Configuration Workflow for Parlay X 2.1 Audio Call/SIP

Following is an outline for configuring the plug-in using the Administration Console or an MBean browser.

  1. Using the Administration Console or an MBean browser, select the MBean listed in the "Properties for Parlay X 2.1 Audio Call/SIP" section.

  2. Configure the behavior of the plug-in instance. See "Reference: Attributes for Parlay X 2.1 Audio Call/SIP" for more information.

  3. If required, create and load a node SLA. For details see “Defining Global Node and Service Provider Group Node SLAs” and “Managing SLAs” in the Accounts and SLAs Guide.

    It is not necessary to set up routing rules to the plug-in instance.

Reference: Attributes for Parlay X 2.1 Audio Call/SIP

This section describes the attributes for configuration and maintenance:

Attribute: FromAddress

Scope: Cluster

Unit: URI

The URI of the connecting subscriber

Attribute: MediaServerFactoryJNDI

Scope: Cluster

Format: Integer

Unit type: Seconds

Default Value: 900

Valid Range: 13600

The JNDI used by the media server driver instance

Attribute: RetentionDuration

Scope: Cluster

Format: String

Default Value: null, for the default instance

Time period to retain the audio call status before deleting it

Attribute: RouteURI

Scope: Cluster

Unit: URI

The network location of the terminal accepting the audio call