고객센타 : 070-7752-2000
팩스 : 070-7752-2001
휴대폰 : 010-9513-0019
email : voipkorea@yahoo.co.kr


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JSS is a next-generation telecommunication service provider specializing in wholesale voice termination, aggregation and transit services. 

JSS specialized in direct routes to Africa, Middle East, Southern Asia, Europe, CIS, Central and Latin America. So far, and we had directly established business relations with more than 600 operators of which 150 mobiles from all over the world.

We take pride in our low cost structure, and pass on the benefit to our customers with the best service and prices in the industry. We are backed by a strong technology infrastructure using world-renowned platforms from Sonus Networks, Genband (formerly Nextone), ALOE Systems (formerly Mera Systems) and Cisco in our carrier-class facilities in New York, Los Angeles, Amsterdam, Johannesburg, Moscow and Singapore.

JSS services are available to both wholesale and retail carriers across the world. Regardless of you being an incumbent carrier, mobile operator, IGW operator, Calling-card operator, broadband telephony service provider, call-center, or even a small call-shop owner, we are equipped with the right systems and expertise to serve your wholesale voice needs efficiently and effectively.

JSS boast our portfolio of niche destinations (Asia,Europ,Nourth & South America, Africa, Algeria, Benin, Burkina Faso, Cape Verde, Cameroon, Cote d'Ivoire, Democratic Republic of the Congo, Ethiopia, Gambia, Ghana, Guinea, Kenya, Liberia, Mali, Mauritania, Nigeria, Senegal, South Africa, Sudan, Tanzania, Togo and Uganda) with strong business relationships that enable us to get the best quality route for you at competitive rates.

Network Infrastructure

JSS has a fully deployed and sophisticated digital switching network using Genband, the premier «Next-Generation» Soft-Switch technology, for emerging Voice-Over-IP (VoIP) for incoming/origination as well as outgoing/termination interconnections. OVETEL has a fully redundant IP network of private MPLS connectivity, and Public-IP via Tier-I carriers, and over a hundred traditional Time Division Multiplexing (TDM) Sonus trunk ports at DS-3 signaling levels offering robust quality and unmatched ASR, ACD, and PDD. Our wholesale voice carrier network offers you scalability in terms of capacity and destinations.

Wholesale services offered by JSS equally suits carriers seeking multiple DS3 capacity as well as smaller carriers seeking a reliable A - Z wholesale routing partner. We encourage smaller voice players to partner with us to meet all of their wholesale voice traffic needs.

Mission Statement

JSS’s mission is to provide reliable service and quality customer care to all of its customers. Our network is comprised of Tier1 telecommunication and bandwidth carriers to ensure quality and reliability.

Transparent Business Practices

We believe that in order to gain the confidence and trust of our vendors, customers and the industry, we must be transparent. This goal is accomplished in many ways.




The Leissner Softswitch is a low-cost but full-featured switching platform providing advanced class 4 transit services and extensive class 5 end-user services across any type of IP- or TDM-based infrastructure. Built on standard hardware and software components, with flexibility and cost-efficiency as guiding design principles, the Leissner system may be used in virtually any all-IP or hybrid VoIP deployment - at a fraction of the costs of a more recognized platform.

Complete VoIP switching system

The Leissner system comes with all the main building-blocks and functions of a true Softswitch; from advanced SS7 capabilities and C4 traffic management and routing, to an extensive range of subscriber services and features for both business and residential end-users. Multi-protocol support allows for MGCP, SIP and SIGTRAN/IUA to be used in parallel and fully interact with each other. End-users may even use MGCP and SIP in combined subscriptions; enabling for instance incoming ENUM-call to MGCP-devises, or bundling of SIP mobile phone clients with POTS-services on MGCP.

ISDN-emulation is a unique system feature providing fully transparent transport of Q.931 signalling and media over IP. SIP-Trunking with proven interoperability with a wide range of commercial and opensource based PBX-system, and different schemes for authentication and authorisation is available.

The system's Session Border Controller (SBC) module handles NAT-traversal, filtering and parsing of SIP- and MGCP-signalling, and setting of ToS/CoS-tags. DOS-protection, Topology Hiding, TLS-encryption and IP-address screening/filtering are other important contributions from an SBC. Interconnects arrangement based on PRI, SS7 and SIP may be used in parallel.

Comprehensive PSTN features

Call control, advanced routing schemes, white-list, carrier pre-select, number portability, subscriber services, centrex services, legal intercept, emergency calls (911/112); the Leissner switch handles all major functions of a legacy transit or local telephony switch. In fact, the system can easily turn into a full-featured TDM switch simply by adding a few TDM-interface modules to the system.

Swiss army-knife of telephony

Truly versatile and rich of features in relation to both PSTN and VoIP, Leissner offers a wide range of uses besides standard POTS-replacement type of VoIP. Whether you are looking for a SS7 gateway for an Asterisk or OpenSER system or require an ACQ implementation in your TDM-network, the Leissner switch will provide a smart and cost-efficient solution.

  • SS7/ISDN/SIP/MCCP signaling gateway.
  • SIP-based Class 4 transit services and SIP-Trunking with advanced security features.
  • IN/IN-light providing ACQ and intelligent routing to legacy or VoIP switching platforms.
  • Media-server providing IVR, Voice Mail, Fax-box.

With well structured program modules and standardized APIs for different types of system functionalities the system can be centralized to one single hardware platform, or distributed over multiple servers and locations. Adding new functions, support for third party protocol stacks or specific hardware is easily handled with adaptation modules to existing APIs. As the system use standard hardware and software components easily integrates to any open third-party VoIP-product or OSS-platform.

Robust and scalable

Field-proven hardware and software components working in a hot-standby redundancy configuration where system fail-over is managed in seconds without affecting traffic - the Leissner softswitch provides true carrier class availability and reliability.

With highly efficient core program modules system capacity is essentially hardware dependant. A minimal configuration based on one standard Intel server will handle up to 20,000 subscribers. By simply adding additional server capacity the system will scale to almost unlimited volumes.


  • Signaling and protocols
    • ISDN, PRI, BRI - network side
    • ISDN, PRI, BRI, User side
    • SIP
    • MGCP
    • SIGTRAN (MxUA Q3 2007)
    • RTP, RTCP
    • User-user signaling
  • SBC functionalities
    • MGCP to SIP/SIP to MGCP
    • Signalling control and filtering
    • TCP & UDP Transport and interworking
    • Transparent to and independant of codecs
    • Media flow-through
    • Legal Intercept
    • T.38 fax relay & T.38 digit relay
    • Call Admission Control (CAC)
    • Quality of Service (QoS) on signalling and Media
    • NAT Traversal
    • Network/Topology Hiding
    • PSTN Number Translation
    • ENUM
    • IPSec and TLS encryption
    • IP-address filtering
    • DOS-protection
    • CDR-generation
  • Network services
    • Access screening, white-list/black-list
    • All Call Query (ACQ)
    • Alternate routing
    • Authentication
    • Call Detail Record Generation
    • Call control
    • Call failure handling
    • Carrier Pre-Select support
    • Destination call routing
    • Dynamic routing
    • Emergency call routing
    • Free number service support
    • International routing
    • Least cost routing
    • Legal intercept
    • Load sharing
    • Multi conference control
    • Numbering plan E.164 extended
    • Number portability support
    • Overlap signaling in all protocols
    • Q&S monitoring
    • Routing based on origin/destination
    • Time based routing
    • Unlimited digit analysis
    • Voice-mail
  • Centrex services
    • Private dial-plans
    • Configurable procedure for reaching the public dial-plan (0, 00 or other)
    • Call pick-up
    • Follow me
    • Call queuing using various schemes
    • IP-Centrex extensions can be seamlessly integrated into the private dial-plan of existing PBX
  • Subscriber Services
    • Anonymous call rejection
    • Automatic Callback on busy
    • Business dial-plans
    • Business group line
    • Call barring (mobile, (international, primary rate)
    • Call barring on specific number series
    • Call forward (direct, busy, no answer)
    • Call transfer
    • Call waiting
    • Caller id presentation
    • Caller id restriction (permanent or per call)
    • Diversion warning (stutter dial-tone)
    • Explicit call transfer
    • Follow me
    • Hotline
    • Individual abbreviated dialing
    • Inquiry
    • Last Number Redial
    • Line hunting
    • Line block
    • Multiple numbers per endpoint
    • n-way calling
    • Refer back
  • System Management
    • Web-interface for configuration of routing, customer data and network layout
    • APIs based on: XML/SOAP
    • SNMP, command line interface.
    • Surveillance/monitoring using software of choice
    • Extensive logging
    • Externally controllable white-list.
  • Hardware
    • Standard Intel cPCI servers in redundant hot-standby configuration
    • Signaling GTW boards: Telesoft MPAC2400
  • Capacity
    • Standard configuration: up to 20,000 residential customers/250,000 BHCA.
    • Easily upgraded by adding server capacity
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■ 무료 : 유선 집전화 휴대폰 ( 한국 미국 중국 카나다) ↔ (국내 해외 여행자 상사 주재원 유학생) / 가입무 무제한무료■