What's GTalk2VoIP Service

GTalk2VoIP gateway는 아래 그림이 보여주는 것과 같이 Google Talk, MSN/Live Messenger and Yahoo! Messenger, Gizmo 사용자들간의 상호 연동 기능 및 SIP/XMPP Network으로의 연동 서비스를 제공한다.

Free voice services available for all major IM clients

  • Voice calls between Google Talk, MSN/Live Messenger and Yahoo!
  • Voice mail service, interoparable between major IM clients.
  • Voice conferencing: with any number of participants of any IM type.
  • Sending off-line messages and notifications.
  • Calling SIP phones and SIP services.
  • Receiving calls to your IM from any SIP phone.
  • Receiving calls to your IM from mobile or landline phones using SIP Broker.
  • Receiving calls to your IM from web users.
  • Free toll-free calling to 1-800, 1-866, 1-877 and 1-888 numbers via SIP Broker.

Paid voice services

  • Outgoing calls from your IM to any telephone numbers throgh a number of VoIP carriers.
  • Incoming calls to your IM from any phone number (PSTN DID assignment).
  • Sending SMS messages from your IM to any mobile phone number.

Information for VoIP service providers

We do offer a number of colaboration services for VoIP businesses, like:

  • PSTN termination to your equipment.
  • VoIM termination from your equipment to our users.

Subscribe to GTalk2VoIP

  • GoogleTalk이나 MSN에서 service@gtalk2voip.com을 친구로 등록함으로로써 가입절차는 끝난다. 이후의 서비스 이용은 메신져 창을 이용한 명령어를 통해 이루어진다. 아래 화면은 GoogleTalk에서 ‘HELP’ Command를 실행한 결과이다.


Call to SIP Phone

  • GoogleTalk에서 SIP UA(charleylim@iptel.org)로 전화를 거는 명령어는 ‘CALL charleylim@iptel.org’이며, 아래 그림이 그 실행 결과이다.

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SIP Server from Flextronics Software Sys

SIP Server from Flextronics Software Systems for Next Generation Networks

SIP Server Framework is a powerful framework to build carrier-class reliable, scalable and feature-rich SIP applications. The FSS SSF comprises two components:

  • SIP Core Infrastructure Server
  • SIP Application Server

SIP Server Framework serves as a base for Proxy as well as any other SIP-based server entity. SSF therefore, radically reduces time-to-market by providing a hassle-free infrastructure that:

  • Implements all the core SIP functionality including state information automatically.
  • Provides a carrier class framework that easily allows new components to be added in and out dynamically as well as statically.

The SIP-SF is a powerful yet flexible solution. It provides tested and standards compliant core SIP functionalities like Call State Control Function (CSCF), Proxy, Registrar, Redirect, B2BUA, and Location Server. More importantly the framework allows application developers to build upon these core functionalities and deliver niche applications. The framework isolates the application developer from all SIP specific protocol and behavioral details. The application developer can focus on what services to provide and the framework can take care of how. FSS' carrier class, proven, interoperable solution thus helps the customers in reducing time and development costs. The FSS solution, one of the most proven and mature products in the industry, handles live traffic for the largest Service Providers' network and for one of the largest Internet Telephony Service Provider (ITSP) Network in Japan.

FSS (formerly HSS)has also introduced Centrex functionalities in its SIP-SF. The Centrex feature enables Equipment Manufacturers to build a hosted PBX solution in minimal time. The FSS SIP Server Framework with Centrex functionality is a powerful framework with a sophisticated dial plan, a hunt group, privileged profiling and intranet/extranet calling.

SIP-SF Architecture

FSS' award winning SIP Server Framework (SIP-SF) ("Internet Telephony" Product of the Year 2002 and "Communications Solutions" Product of the Year 2003) is an ideal platform for hosting SIP services. SIP-SF is a ready-to-deploy SIP Proxy, Registrar, Redirect, Presence and Location Server. FSS’ SIP-SF today provides several advanced features, including SIP Centrex (Hosted PBX) and Per-User Call Processing Language, all of which render tremendous value in using it as an underlying platform to build new services while significantly reducing time, risk, and cost to market. The SIP-SF is a ready-to-deploy platform with Virtual IP redundancy, SIP Load Balancing, SNMP Manageability, Congestion Control, Emergency Calling, and other real-life network requirements already addressed.

Targeted at OEMs, SIP-SF has been developed keeping in mind the flexibility and extensibility that OEMs so much need to be able to offer differentiated products to the market.

FSS' SIP server is a dual play architecture, positioned as both Core Infrastructure and Application Development Framework.

Click to enlarge
Click to enlarge
SIP-SF Core Infrastructure Element
OSA Application Server

FSS’ SIP Server can be positioned both as the SIP core infrastructure element to build basic infrastructure needed in a SIP Network or using its B2BUA feature set, as well as an application infrastructure to build applications, such as Collaboration, CTI, and Gaming. The SIP-SF thus enables one to easily integrate multiple call models, be it BCSM-based or RFC 3261, and to use any one of them depending on the kind of service being executed. Further, the SIP-SF can be made to interact with an SCN using a regular BCSM-based model and at the same time, perform forking functionality of proxy using the regular 3261 UAC/UAS model.

SIP-SF Features

Salient Features

  • Ready-to-deploy SIP Proxy, Registrar, and Redirect Server
  • Value-added features including SIP Centrex, Call Processing Language, and Presence Server
  • Integrated load balancing and high availability platform
  • Multi operating system portability
  • Easy management - SNMP, API, XML
  • Multiple database support - ODBC
  • Supports both IPv4 and IPv6.
  • TLS security and digest authentication
  • Support for emergency calls and priority routing
  • Implemented in C++ to offer complete object-oriented interface
  • Configuration through Number Translation Mark up Language (NTML)

Architectural Superiority
The key architectural features of the SIP Server Framework are:

  • High scalability - performance increases with increase in number of CPUs
  • Can support multiple CSMs (service logic) simultaneously
  • Support for user-defined service logic
  • Component-based architecture- can chain components as desired for future enhancements
  • Supports distributed operation across various nodes
  • Centralized configuration of all components
  • Message-based interfaces
  • Threaded model-thread pool management
  • Easy integration with third party modules- all functional modules implemented over APIs
  • Compliant to ISO C++ and EC++ standards to ensure maximum portability

Key Standards Compliance

  • 'SIP: Session Initiation Protocol', RFC 3261
  • 'HTTP Authentication: Basic and Digest Access Authentication', RFC 2617
  • Session Initiation Protocol (SIP): Locating SIP Servers', RFC 3263
  • 'CPL: A Language for User Control of Internet Telephony Services', draft-ietf-iptel-cpl-06
  • 'URLs for Telephone Calls', RFC 2806
  • 'Management Information Base for Session Initiation Protocol', draft-ietf-sip-mib-04.txt

SIP Application Server Features

The SIP Application Server (B2BUA) provides a scalable, reliable, andfeature-rich environment for rapid service creation. FSS' B2BUA Serveris a framework that exposes a set of well-defined APIs to theapplication to develop various services, such as Prepaid, NatTraversal, Call Queuing, Click-to-Dial, and Call Pick-Up. These APIs aswell as their parameters are Open Service Architecture (OSA)-compliant.In addition, FSS' B2BUA Framework also supports additional APIs tocreate innovative applications.

Salient Features

  • RFC-compliant: RFC 3261-compliant B2BUA
  • Scalability: Highly scalable feature distribution
  • Availability: Reliable high availability support
  • Comprehensive support: Support for multi-party, multimedia call control
  • Flexible: Provides the applications complete access to SIP messages
  • Multiple deployment options: Single node or multi node deployment
  • SIP Interface to proxy/media servers
  • Multi operating system portability

Architectural Features

  • High availability APIs to allow the applications back up serialized critical data and reconstruct the state based on critical data
  • APIs to send SUBSCRIBE, NOTIFY, MESSAGE, and proprietary SIP messages in order to facilitate development of presence-based applications and chat applications
  • APIs to move call-legs from one call to another in order to facilitate development of applications, such as Call Pick-Up
  • In-built Load Monitor to protect against overload conditions
  • Event criteria filtering mechanisms

Key Standards Compliance

SIP Protocol Compliance

  • 'SIP: Session Initiation Protocol', RFC 3261, June 2002, Rosenberg et al.
  • Session Description Protocol, RFC 2327
  • An Offer/Answer Model with the Session Description Protocol (SDP), RFC 3264, June 2002
  • Reliability of Provisional Responses in the Session Initiation Protocol (SIP), RFC 3262, June 2002
  • SIP INFO Method, RFC 2976, October 2000

Open Service Access (OSA) Compliance

  • 3GPP TS 29.198-04-3-600 OSA API Part4-SubPart3-MPCC SCF
  • 3GPP TS 29.198-04-3-600 OSA API Part4-SubPart4-MMCC SCF
  • 3GPP TS 29.998-04-4-500 Part4-SubPart4-MPCC ISC

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