Processing Telephony Voice Media and SIP Signaling
DSP-Empowered for Media and SIP, 480Chs Per Board
Widely Used In High-Capacity Call Center, UC and More
Features & Benefits
ALL-IN-ONE architecture of integrated SIP signaling and multimedia processing
Help VoIP developers to integrate all cost-effective, scalable, feature-rich IP telephony applications and value-added services in single box in IP networking enviroments.
Rich media processing: conferencing, compression, fax, echo cancellation, call control
Support for enhanced multimedia processing resources, including conferencing, IVR, fax, compression, echo canceller, call control , help developers develop feature-rich applications
Signaling protocols: SIP/MGCP
Offer robust SIP signaling technologies for service providers and applications developer to develop and deploy high capability, high performance and highly available enhanced services in PSTN and PLMN networks
Universal user-friendly SHCTI API supports for a range of calling features
Unified API architecture minimizes efforts on application development and deployment, and PSTN-based or SIP-based applications can be migrated among all Synway's hardware platforms
Optional form factor: PCI, PCI-X*, PCI-express* interface
Support existing or next-generation form factor of network infrastructure, server or chassis without altering application programming interfaces
Global approval by service provider, application developer and system integrator
Deployed into large-scale call center application, value-add service, unified messaging solution by world-class application developers and service providers
Scalable and upgradeable up to 120 ports per slot, maximum 1920 ports per system
Cost effective and scalable hardware for a broad range of applications, and specifically designed to fulfill demands in high capacity, highly available and redundant system architecture.
A single board has 32 IP channels; a single PC supports up to 2 SHN-32A-CT/PCI boards.
Each board can be configured to comply with either SIP or H.323 protocol, or both.
Supports the following functions during the call: voice recording and playing, volume adjustment, active/silence detection, voice QoS service assurance, dynamic CODEC change, etc.
All channels are allowed to play/record voices simultaneously; supports Automatic Gain Control (AGC) in recording operation.
Supports call transfer and call hold during IP calls.
Allows DTMF signal transmission and detection by any of the three methods: in-band, out-of-band (RFC2833), Signaling (SIP-INFO or H245 signaling).
Includes H.100 bus, compatible with MVIP, SC and ST bus, facilitating smoothconnectivity to third-party boards with H.100 bus for the transfer of voice data from/to other devices.
The flexible distributed conferencing system sets no limit on the number of simultaneous conferences and participants in each conference, allows monitoring and recording of the whole conference and each individual speaker.
Each board has a unique hardware serial number written in the firmware to distinguish itself from other boards and prevent piracy.
The on-board authorization code identification circuit is designed for software safety. Users can apply to our company for the authorization code.