Release Note

Contents

Introduction

As of February 1, 2012, Genesys is no longer an affiliate of Alcatel-Lucent; any indication of such affiliation within Genesys products or packaging is no longer applicable. Please see the Genesys website at http://www.genesyslab.com for more details.

This release note applies to all 8.0 releases of the SIP Endpoint SDK.

• Links in the Contents section enable you to access information regarding a specific release. Releases are listed by version number rather than in date order. For this reason, a recent release may be listed after earlier releases, if the version number is lower. Except when otherwise noted in the information for a specific release, each release includes all of the features and corrections that were introduced for the applicable operating system at earlier dates, regardless of the version numbering.
• The Release Number table lists the supported operating systems for each release at a high level. For more detailed information, see the Genesys Supported Operating Environment Reference Manual.
• The Known Issues and Recommendations section is a cumulative list. It includes information on when individual items were found and, if applicable, corrected.
• The Corrections and Modifications section for each release may list additional issues that were corrected without first being documented as Known Issues.
• The Discontinued Support section is cumulative. It lists functionality that is no longer supported and the release number of the version in which support was discontinued.

Use of Third-Party Software

Genesys follows applicable third-party redistribution policies to the extent that Genesys solutions utilize functionality of commercial or non-commercial third parties. For specific information on any third-party software used in this product, see the Read Me.

Supported Operating Systems

New in This Release

This release is under shipping control. This section describes new features that were introduced in this release of the SIP Endpoint SDK.

• Windows 7 QoS support is now available using Window Administration tools.

• The SipEndpoint.config file has been updated so that users can configure audio in/out devices. Audio devices can be configured with the following priority logic:

  1. The SIP Endpoint SDK configures a headset audio device (both microphone and speaker should have the same name); if the headset device configuration fails then the default audio devices will be configured.

     <setting name="use_headset" value="1"/>

  2. The SIP Endpoint SDK configures audio devices individually (each type, such as microphone and speaker, can have a different name); if the audio device configuration fails then the default audio devices will be configured.

     <setting name="manual_audio_devices_configure" value="1"/>
<setting name="use_headset" value="0"/>

  3. The SIP Endpoint SDK configures default audio devices.

     <setting name="manual_audio_devices_configure" value="0"/>
<setting name="use_headset" value="0"/>

  If using the enable/disable Speaker Phone capability, the SIP Endpoint SDK will toggle between devices that are internally assigned as appropriate Speaker Phone devices and devices that were defined in SipEndpoint.config file. To get currently used devices, use the GetMicName() and GetSpeakerName() methods. (ER# 256816936)

• The SIP Endpoint SDK now provides support for the refer-enabled=true SIP Server option. (ER# 256816933)

• Added new properties for the SessionEventArgs(object sender, SessionEventArgs e) event that have the following key values:

  • RemotePartyAddress
  • DisconnectReason
  • StatusLineMsg
  • ResponseCode

  (ER# 256816939)

• First-party call control (1pcc) is now supported, using the ICallControl(callControl=endpoint.Resolve("ICallControl") as ICallControl;) interface and the following members:

  • Dial(destination);
  • Answer(sessionId);
  • Release(sessionId);
  • Retrieve(sessionId);
  • Hold(sessionId);
  (ER# 256816918)

Corrections and Modifications

This release includes the following corrections and modifications:

There are no corrections or modifications in this release.

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Supported Operating Systems

New in This Release

This is a hot fix for this product. This release does not contain new features or functionality.

Corrections and Modifications

This release includes the following corrections and modifications:

Softphones developed using the SIP Endpoint SDK now handles the following scenario correctly:

1. An agent receives a call and hangs up quickly.
2. The agent immediately receives the next call.

Previously, the application could exit unexpectedly due to an error accepting the second call in this scenario. (ER# 243684483)

Calls using DTMF interactions are now handled correctly. Previously, sending digits as a DTMF interaction caused the softphone application to stop responding in some cases. (ER# 243565893)

The SIP Endpoint SDK no longer plays audio incorrectly through speakers when Windows is configured to play sounds through the headset. (ER# 245643947)

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Supported Operating Systems

New in This Release

This release is under shipping control. This section describes new features that were introduced in the initial 8.0 release of the SIP Endpoint SDK.

• This is the first release for this product.

• This release of the SIP Endpoint SDK provides support for:

  • Windows Server 2003, Windows XP, Windows Vista, and Windows 7
  • .NET Framework 3.5 SP1
  • Genesys CIM Platform 8.0
  • Genesys SIP Solution 8.0
  
  

• The SIP Endpoint SDK includes the telephony functionality and features described below, for both inbound and outbound activity:

  • Basic call functionality (answer, dial, hold and retrieve, or release calls).
  • Call transfers and conferences, including both blind and two-step.
  • SIP and RTP media termination at the desktop.
  • Ability to mute or unmute the microphone from the desktop.
  • Third-party call control (3PCC) through SIP Server.
  • First-party and third-party DTMF support.
  • The RTP port range can be used for sending and receiving media. (Note: The minimum port range is 10.)
  • Quality of Service (QoS) Type of Service (TOS) bits can be used to control voice quality for further routing purposes. (Supported for Windows XP only.)
  
  

• The SIP Endpoint SDK consists of strongly-named assemblies.

• The SIP Endpoint SDK supports data compression using the following audio codecs:

  • G.729b (for up to five channels)
  • G.711a
  • G.711u
  • G.722
  • GSM
  
  

• Audio devices such as headsets, microphones, and speakers are auto-detected.

  Note: The SIP Endpoint only recognizes one input device and one output device. For predictable results, and to ensure that the SIP Endpoint uses the intended audio devices, Genesys recommends using only one audio input/output pair.

• Applications that use the SIP Endpoint SDK can detect the status of a USB headset to determine whether a headset is plugged or unplugged.

• Beep Tones Control is provided when using the SIP Endpoint SDK with both Genesys Configuration Server 8.0 and Genesys SIP Server 8.0. This feature allows you to play an audio file for an active call.

• The SIP Endpoint SDK supports logging.

Corrections and Modifications

There are no corrections or modifications for this product in the initial 8.0 release.

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Known Issues and Recommendations

This section provides the latest information on known issues and recommendations associated with this product.

The SIP Endpoint SDK can experience a problem establishing an audio path during internal (consultation) call in the following scenario:

1. An inbound call arrives at a routing point with a loaded routing strategy.
2. The call is routed to an Agent A using Interaction Workspace, and SIP Endpoint accepts the call.
3. The inbound call is put on hold. The customer listens to configured music while on hold.
4. Agent A makes and internal/consultation call to Agent B, located on a remote SIP Server.
5. Agent A does not hear ringing, even though it is provided by Stream Manager.
6. Once the call is established there is no audio path established; both agents only hear silence.

To prevent this issue, Genesys recommends that you use SIP Endpoint in dual-dialog mode in this scenario, by changing the dual-dialog-enabled parameter to true. (ER# 258360035)

The following list of codecs are supported by SIP Server in 1pcc mode, but not in 3pcc mode:

• G722_2
• G723
• G726
• BroadVoice32
• BroadVoice32 FEC
• DVI4
• DVI4 Wideband
• iLBC
• L16 PCM Wideband
• Speex
• Speex FEC
• Speex Wideband
• Speex Wideband FEC

(ER# 257586441)

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Discontinued Support

This section documents features that are no longer supported in this software. This cumulative list is in release-number order with the most recently discontinued features at the top of the list.

There are no discontinued features or functions in this release of the SIP Endpoint SDK.

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Internationalization

Information in this section is included for international customers.

There are no internationalization issues for this product.

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Additional Information

Additional information on Genesys Telecommunications Laboratories, Inc. is available on our Technical Support website. The following documentation (located on the Developer Documentation CD, the Technical Support website, and the Genesys SDK Documentation Wiki) also contains information about this software. Please consult the wiki pages that provide deployment information first.

• Genesys SDK Documentation Wiki - Further information about the SIP Endpoint SDK, including deployment guidelines, can be found from the SIP Endpoint SDK wiki pages.

• SIP Endpoint SDK 8.0 API Reference - This reference manual provides detailed information about the SDK, and usage guidelines.

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