This chapter describes the Oracle Communications Services Gatekeeper Parlay X 2.1 Audio Call/Session Initiation Protocol (SIP) communication service in detail.
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 Parley X 2.1 Audio Call communication service supports for the application-facing interfaces and the network protocols, see Services Gatekeeper Statement of Compliance.
Using this communication service, an application can:
Set up a Parlay X 2.1 call between a terminal device and a media server to play an audio file (such as WAV).
Get the status for a Parlay X 2.1 Audio call (played, playing, pending, or error).
Explicitly end any of these audio calls.
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 file format 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.
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).
This is a sample Audio Call/SIP call flow:
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).
The terminal returns a SIP 200 OK message to the application containing a valid SDP.
The application's media server controller then processes the SDP and sends an ACK message back to the terminal address.
For information about the SOAP-based interface for the Parlay X 2.1 Audio Call/SIP communication service, see the discussion about Parlay X 2.1 Audio Call Interfaces in Services Gatekeeper Application Developer's Guide.
For information about the RESTful Audio Call interface, see the discussion about RESTful Audio Call in Services Gatekeeper 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.
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.
See "Events, Alarms, and Charging" for more information.
Table 5-1 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.
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 oracle.ocsg.plugin.audio_call.sip.south.call.AudioCallCdrContext.
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 oracle.ocsg.plugin.audio_call.sip.south.call.AudioCallCdrContext.
Table 5-2 maps methods invoked from either the application or the network to the transaction types collected by the Services Gatekeeper statistics counters.
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 file to 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.
Table 5-3 lists the technical specifications for the communication service.
Table 5-3 Properties for Parlay X 2.1 Audio CalI/SIP
| Property | Description |
|---|---|
|
Managed object in Administration Console |
To access this object, select domain_name, then OCSG, server_name, Communication Services, and then the plugin_instance_id, in that order. |
|
MBean |
Domain=wlng_nt_audio_call_px21#6.0 Name=wlng_nt InstanceName=Audio_Call_sip Type=Plugin_px21_audio_call_sip.AudioCallMBean Documentation: See the description of AudioCallMBean in the ”All Classes” section of Services Gatekeeper OAM Java API Reference. |
|
Network protocol plug-in service ID |
Plugin_px21_audio_call_sip |
|
Network protocol plug-in instance ID |
The ID is assigned when the plug-in instance is created. See the discussion about configuring and managing the plug-in manager in Services Gatekeeper System Administrator's Guide. |
|
Supported Address Schemes |
tel, sip |
|
Application-facing interfaces |
com.bea.wlcp.wlng.px21.plugin.AudioCallPlugin |
|
Service type |
AudioCall |
|
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 |
Following is an outline for configuring the plug-in using the Administration Console or an MBean browser.
Using the Administration Console or an MBean browser, find AudioCallMBean listed in the Table 5-3.
Configure the behavior of the plug-in instance using AudioCallMBean. For information on the AudioCallMBean methods and fields, see the "All Classes" section of Services Gatekeeper OAM Java API Reference.
If required, create and load a node SLA. For details see the discussions on defining global node and service provider group mode SLAs and managing SLAs in Services Gatekeeper Portal Developer's Guide.
It is not necessary to set up routing rules to the plug-in instance.