The DiaStar® Server (DSS) delivers a new level of signaling, media, audio, and video transcoding services to complement and enhance the Asterisk and FreeSWITCH platforms. DSS is compliant with the Woomera open protocol and can operate in conjunction with any project or product that implements a corresponding Woomera client. The DSS allows developers access to video conferencing, video processing, call progress analysis, SS7 and SIGTRAN signaling, while continuing to develop in their native development environment.
When added in a Client-Server architecture, the DSS can offer a variety of improvements to open source telephony projects.
- Scalability - Media functions can be either performed by the telephony application or offloaded to the DSS to enable higher channel counts than are possible in a single-server system
- Redundancy - A system can be configured without a single point of failure. If interrupted, the system switches to a backup component.
- Reliability - Dialogic technology is engineered to provide high reliability and availability. Adding a DSS might require some initial downtime to configure Asterisk, but adding chan_woomera clients for Asterisk should not require downtime.
In addtion, the DSS gives developers access to Dialogic's proven media processing software and PSTN network interface technology while still creating applications in a way that is native to the project in which they are working.
Highlights
- Client/server architecture - Allows for the resources of DSS to be shared by multiple Asterisk or FreeSWITCH systems
- Video Playback and Record - Allows developers to create Interactive Voice and Video Response (IVVR) systems for the delivery of complex instruction sets and other information best delivered as video or graphics
- Application control of video conferencing functions - Provides video conferencing with multiple screen layouts, user-types, and participant captions
- Real-time audio and video transcoding - Allows real-time audio and video conferencing among endpoints that support different codecs and enables storage of audio and video media in a standard format while allowing playback on a variety of devices
- SIGTRAN and SS7 signaling - Allows applications created using Asterisk or FreeSWITCH to be deployed in carrier environments where ISDN, Robbed-Bit, or other inband signaling is not available
- Real-time menu builder - Enables developers to build interactive video menus from inside the application, minimizing video production requirements and providing maximum flexibility in application implementations
- Abstracts and encapsulates media, signaling, and gateway functions - Allows developers to access Dialogic® functionality while working within their native programming environment
Components
DSS is developed as an open source project (www.projectdiastar.org) that also includes access to closed-source components from Dialogic that provide value added functionality. These components include certain Dialogic® boards that provide ISDN variants, T1/E1 CAS, SS7, and so on, as well as certain derivatives of Dialogic's industry-leading media processing software that can add functionality, such as call progress analysis and video.
Supports Dialogic® Perfect Call for Call Progress Analysis
Dialogic® Perfect Call provides call progress analysis that is intelligently tolerant of the wide variation in call progress signaling tones found in central offices and PBXs around the globe. It also offers accurate performance without complex programming. DSP-based algorithms are used to accurately discriminate live human speech from recorded human speech and network noise. User-defined tone detection templates can also be defined. Detection results are made available in Asterisk Dialplan as well as the Asterisk Manager Interface (AMI).
Release 2.4 Features and Enhancements
- Robust support of SIP video from the Asterisk Dialplan including:
- Play and record audio and video
- H.263 support
- H.264 support
- MPEG4 support
- Video transcoding between H.263, H.264 and MPEG-4
- Video transrating - Allows video playback at different frame rates
- Video scaling - Enables video playback on screens of different sizes
- SS7 support - Provides conventional SS7 support for carrier installations
- SIGTRAN support - Provides SS7 signaling over IP
- G.711, G.722, G.729 audio codecs - Enables flexible endpoint support
- HD Voice - Provides significantly superior natural sound and a dramatically increased sense of participation in live conversations and video conferences
- Video conferencing - Allows full duplex video connectivity for multi-party conferences
- Native media bridging - Allows media to be bridged across the DSS (when both parties in a session are connected through the DSS), reducing load on the Asterisk or FreeSWITCH system. Learn about the applications you can create with video capabilities. A white paper on video-enabled contact centers may also be helpful.
- 3G-324M - Adds support for video-enabled mobile applications using the 3G-324M protocol with the AMR-NB, H.263, H.264, and MPEG-4 codecs
- RTSP streaming - Enables on-demand relay of RTSP video streams from RTSP servers and security cameras
Features Planned for Future Releases
- Fax (V.17) and FoIP (T.38) support - Permits fax-capable applications in both TDM and IP environments
- Increased video resolution (VGA and higher)
For more technical specifications, see the datasheet.