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New In This Release

Warning
This documentation is outdated and is included for historical purposes only.


Check out the new features that have been added in the latest releases of SIP Endpoint SDK.

.NET

New in Release 8.1.2 for .NET

Release 8.1.200.25

  • SIP Endpoint SDK 8.1.2 for .NET Release 8.1.200.25 allows you to select an audio layer for WebRTC using the Windows environment variable GCTI_AUDIO_LAYER.

Release 8.1.200.23

  • SIP Endpoint SDK 8.1.2 for .NET Release 8.1.200.23 supports Genesys SIP Proxy in standalone mode.

Release 8.1.200.14

Note: SIP Endpoint SDK 8.1.2 for .NET Release 8.1.200.14 supports the following features. Additionally, support includes all new features supported by SIP Endpoint SDK 8.1.2 for Apple OS, with the exception of OS X 10.8 Mountain Lion.

  • IPv6 support
  • Forced DNS lookup for operating with SIP Cluster
  • Support for long device names for Windows Vista SP1 or later
  • Genesys SIP libraries and Google's WebRTC media stack are now used instead of the CounterPath SDK
  • Support for basic first-party call control features:
    • Dial
    • Answer
    • Hold
    • Retrieve
    • Release
  • VoIP monitoring of packet jitter and latency using RTP statistics
  • Support for In-Dialog SIP:INFO message exchange using customer-provided information for the Content-Type and Content headers
  • Access to raw video frames
  • Support for the following Plantronics headset models in the standard headset mode:
    • Savi W740 and W745
    • Savi W440
    • Blackwire C310/C320
  • Windows 8 compatibility
  • Support for Picture-in-Picture (PiP) display
  • Support for Microsoft's .NET WPF (Windows Presentation Foundation)
  • Support for the creation of RTCP Extended Reports (RFC 3611) and the ability to publish them according to RFC 6035 at the end of each call. Note: The 8.1.2 implementation of this feature includes only MOS values (along with PacketLoss and Delay statistics, which are used for MOS estimation).

New in Release 8.1.1 for .NET

Release 8.1.100.04

  • SIP Endpoint SDK for .NET now supports the collection of RTP audio and video statistics, giving users real-time access to these statistics during a call.

Release 8.1.100.03

  • SIP Endpoint SDK for .NET now supports the G.729b voice codec with the annexb=no extension.

Release 8.1.100.02

  • SIP Endpoint SDK for .NET now supports video using a new video control interface that alllows for the control of video source, frame events, window handles, and connectivity mode.
  • When using the SIP Endpoint SDK for .NET, you can now encrypt SIP messaging and the media channel using TLS and SRTP, respectively.
  • SIP Endpoint SDK for .NET now supports the RFC2617-style digest authentication that is currently used by SIP Server.

Apple OS

New in Release 8.1.2 for Apple OS

  • SIP Endpoint SDK 8.1.2 for Apple OS supports OS X Mountain Lion (10.8)
  • This release of SIP Endpoint SDK for Apple OS supports the following codecs:
    • G.711 (PCMA, PCMU)
    • G.722
    • iLBC (Internet Low Bitrate Codec)
    • iSAC (Internet Speech Audio Codec)
    • VP8 video
  • SIP Endpoint SDK 8.1.2 for Apple OS supports the following Plantronics headset models in the standard headset mode:
    • Savi W440
    • Savi W7xx
    • Blackwire C320

SIP Endpoint SDK 8.1.2 for Apple OS also supports the following features:

  • TLS 1.2 protocol (RFC 6176)
  • AGC (Automatic Gain Control)
  • MWI (Message Waiting Indicator)
  • You can now specify these behaviors when a SIP Endpoint user does not have a working USB headset:
    • Whether SIP Endpoint should automatically reject an incoming call
    • The SIP error code to be sent to the inviting party
  • INVITE messages now have an additional header that contains user data. This data can be obtained by using the following new method of GSSessionControlService:
- (GSResult) dialFrom:(id<GSConnection>)connection to:(NSString*)destination withData:(NSString *)data;
  • Hangup on RTP inactivity timeout
  • Configuration of:
    • RTP port ranges
    • SIP port ranges
  • Continuous Registration
  • DTMF tones can now be sent using SIP INFO
  • VoIP monitoring of packet jitter and latency using RTP statistics
  • NAT (Network Address Translation) traversal methods:
    • ICE (Interactive Connectivity Establishment)
    • STUN (Session Traversal Utilities for NAT)
    • TURN (Traversal Using Relay NAT)
  • Connection to Genesys SIP Cluster

In addition, this release of SIP Endpoint SDK for Apple OS uses a SIP stack that has been developed by Genesys.

New in Release 8.1.1 for Apple OS

  • SIP Endpoint SDK 8.1.1 for Apple OS supports OS X Lion (10.7)
  • This release of SIP Endpoint SDK for Apple OS supports the following voice codecs:
    • G.711 (PCMA, PCMU)
    • G.722
    • iLBC (Internet Low Bitrate Codec)
  • When devices and services use multiple voice codecs, the SIP Endpoint SDK for Apple OS supports the negotiation of the voice codec that will be used between them.
  • The SIP Endpoint SDK for Apple OS supports Quality of Service (QoS), which helps guarantee that packet traffic for a voice or other media connection will not be delayed or dropped due to interference.
  • The SIP Endpoint SDK for Apple OS supports additional security signaling and media encryption via SRTP (Secure Real-Time Transport Protocol).
  • The SIP Endpoint SDK for Apple OS supports both first party call control (1PCC) and third party call control (3PCC).
This page was last edited on July 26, 2016, at 21:31.
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