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BTS-Dedalus  Dedalus 

ENERGIZE your network

with carrier-class, scalable and reliable softswitch, engineered for easy operation and maintenance, high performance and advanced troubleshooting.

DEDALUS provides a intuitive user interface to manage and control carrier interconnections for straightforward provisioning and configuration. Changes in provisioning and routing take place dynamically. Through the GUI the specific configuration for each interconnection is deployed, enabling the definition of technical requirements needed to identify each carrier in the network, trunk id, protocols, ip range, gateway list, ports, capacity, translations table, prefix, etc.

The consolidation of capabilities on a single platform eliminates the need for a separate overlay elements for each signaling protocol and transcoding, significantly reduces equipment expenditures while providing greater operational efficiencies.

MAIN FEATURES

DEDALUS manages conversion between Voip Protocols SIP, MGCP/SIGTRAN, H323.

•Support codecs g729r8, g723r53, g723r63, g711a-law, g711u-law, also HD codecs g722 and g722.2.

•An absolute transparent transcoding process in case we do not need to adjust codecs between endpoints. Every transcoding module can handle to 300 calls.

DEDALUS can manage up to 3000 simultaneous calls in a single chasis.

•Compatible with SIP/H323 standard developers. (Cisco, Quintum, Nextone, Linksys, Grandstream)

•Ability to bridge WebRTC endpoints to classic SIP phones without any dedicated SBC or media gateway.

•Endpoints ip registration and filtering calls from un-configured ips.

•It’s a scalable and reliable platform so there is no limit on the number of carriers interconnected.

•Platform based on a redundant cluster architecture with heartbeat and floating IP. All states are reported to OAM&P server in real time. If active application fails for any reason, hot standby process will take over as active.

•Redundant controllers provide backup for hardware, memory and databases.

•Bandwidth management via call counts.

•In-memory routing based on Redis and SQL DB for persistence storage.

•Combination of LCR and automatic intelligent routing scheme.

•Mapping release cause code.

 

 

•System monitoring with real time call status and ASR statistics.

•Customized log filter solution compatible with Wireshark and Ethereal for full graphical protocol traces.

•Managed through an intuitive Web User Interface.
•Real time configuration.

•User profile based on a role and permission security.

    • SIP-T

RFC 3372 Session Initiation Protocol for Telephones (SIP-T).

    • SIP-I

•ITU Q.1912 for ISUP/C7-to-SIP inter working.

    • SIP

RFC 3261 Session Initiate Protocol.
RFC 2976 SIP INFO Method.
RFC 3398 ISUP-SIP Mapping.
RFC 3515 Refer Method.
RFC 3578 Overlap.
RFC 2327 Session Description Protocol.
RFC 3581 An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing.
RFC 3665 Session Initiation Protocol (SIP) Basic Call Flow Examples.
RFC 3891 “Replaces” Header.
RFC 3892 Referred By Mechanism.
RFC 3311 Update Method.
RFC 4566 Session Description Protocol.
RFC 5806 Diversion Indication in SIP.
RFC 3326 The Reason Header Field.
RFC 2543 Session Initiation Protocol.
RFC 3262 Reliability of Provisional Responses.

 

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