[采日] LifeLOG^1.0

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.

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

신명리 동백꽃 - 김종경 시 / 우덕상 작곡/ 김철호 음성

음악을 듣다가 동백에 대해 검색해 보니 아래와 같은 동백에 대한 전설이 있더군요.

FSS' Softswitch Framework

FSS' Softswitch conforms to TIPHON distributed gateway architecture. Softswitch solution from FSS supports a wide variety of signaling protocols, like SS7, R1/R2 and ISDN and has a potential to incorporate GR-303 and V5.2. FSS Softswitch can control media remotely through MEGACO and MGCP.

Softswitch Framework Context in System

Softswitch Configuration

The design of FSS' Softswitch enables easy deployment in the network. FSS Softswitch provides the flexibility to configure all external network elements like Gatekeeper, Media Gateway, Signaling Gateway, Application Server, etc. Policies can be configured by customizing Routing, Service Control and Address Translation. Further, product related configurations for initialization, provisioning and redundancy are available through open APIs.

Softswitch Service Architecture

• Services targeted – address translation, routing, IVR, Charging, Emergency, Network based, third party call handling, web based
• Service Interface provides support for – CLASS services/supplementary services, IN services, SIP based
• Architecture independent of signaling, but support for SIP, H.323, SS7, ISDN, R2, POTS ensured

FSS' Softswitch Framework : Functions

The FSS Softswitch is offered both as a product and a software framework with source license. FSS has already integrated the Softswitch into different product architectures and offers its services in integrating and porting the Softswitch solution onto the target.

Services

• Low-level call management APIs for INAP interfaces
• High-level call management APIs for Parlay adapter
• SIP interface to Application server
• CLASS services hosted natively
• Legacy and Enhanced services supported

Softswitch Framework Functional Block

Interworking Call Control

• IN CS2 – AIN 0.3 BCSM based call control
• Signaling Adaptations Supporting diverse SCN and packet protocols
• Enables easy integration of new signaling protocols

PSTN Interfaces

• SS7 termination (ANSI or ITU-T) either through external SG or direct
• Dual-seizure, reset, group blocking/unblocking supported
• Supports third-party pause and re-routing
• CAS R1/R2 supported on PSTN & H.323 v2 on packet network
• ISDN Q.931 interface supported

Packet Interfaces

• H.323 subscriber and trunk interface
• SIP subscriber and SIP-T trunk interface
• MGCP (RFC 2705) gateway interface
• MEGACO/H.248 gateway interface

Bearer Control

• H.248/MGCP based Gateway Control
• Ability to manage multiple transactions
• Enabling media control for packet-TDM inter-working
• Ability to manage various gateway types (Access, Trunking and Residential)

Resource Management

• Route Information maintained
• SCN and PDN terminations and associated data maintained
• Resource allocation, de-allocation and maintenance
• Local and External Address Translations supported
• Single point routing and distribution of messages to multiple MGs

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인터넷 기업들의 로고를 모아놓은 그림입니다. 그림을 바탕화면에 두고, 틈이 생기면 하나씩 둘러보고 있는 중입니다. 아래 그림은 http://flickr.com에 그림을 올리고, xmlrpc를 이용해서 태터로 보내본 것입니다. 아래와 같이 나오는군요. 태터의 MetaWebLogAPI의 주소는 http://bloghome/plugin/BlogAPI 로 주면 됩니다. 태터에서 Blogger API 플러그인이 설치되어 있어야 합니다.

Web2.0 Company, originally uploaded by Charley Lim.

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Bluetooth communication occurs between a master radio and a slave radio. Bluetooth radios are symmetric in that the same device may operate as a master and also the slave. Each radio has a 48-bit unique device address (BD_ADDR) that is fixed.

Two or more radio devices together form ad-hoc networks called piconets. All units within a piconet share the same channel. Each piconet has one master device and one or more slaves. There may be up to seven active slaves at a time within a piconet. Thus, each active device within a piconet is identifiable by a 3-bit active device address. Inactive slaves in unconnected modes may continue to reside within the piconet.

A master is the only one that may initiate a Bluetooth communication link. However, once a link is established, the slave may request a master/slave switch to become the master. Slaves are not allowed to talk to each other directly. All communication occurs within the slave and the master. Slaves within a piconet must also synchronize their internal clocks and frequency hops with that of the master. Each piconet uses a different frequency hopping sequence. Radio devices used Time Division Multiplexing (TDM). A master device in a piconet transmits on even numbered slots and the slaves may transmit on odd numbered slots.

Fig 1: Bluetooth Scatternets and Piconets

Multiple piconets with overlapping coverage areas form a scatternet. Each piconet may have only one master, but slaves may participate in different piconets on a time-division multiplex basis. A device may be a master in one piconet and a slave in another or a slave in more than one piconet.

The main principle in mind when developing the Bluetooth Protocol Architecture has been the maximization and the re-use of existing protocols for different purposes at the higher layers. The one main advantage is that existing (legacy) applications can be adapted to work with the Bluetooth Technology. The Bluetooth Protocol Architecture also allows for the use of commonly used application protocols on top of the Bluetooth-Specific protocols. In simpler terms, this permits new applications to take full advantage of the capabilities of the Bluetooth technology and for many applications that are already developed by vendors; they can take immediate advantage of hardware and software systems, which are also compliant with the Specification.

Fig 1:The Bluetooth Protocol Stack Model The protocols and layers in the Bluetooth protocol stack

Protocol Layer	Protocols in the stack
Bluetooth Core Protocols	Baseband, LMP, L2CAP, SDP
Cable Replacement Protocol	RFCOMM
Telephony Control Protocol	TCS Binary, AT-commands
Adopted Protocols	PPP, UDP/TCP/IP, OBEX, WAP, vCard, vCal, IrMC, WAE

In addition to the above protocol layers, the Specification also defines a Host Controller Interface (HCI). This provides a command interface to the baseband controller, link manager, and access to hardware status and control registers.

Three-in-One Phone

Telephone handsets built to this profile may connect to three different service providers. First, telephones may act as cordless phones connecting to the public switched telephone network (PSTN) at home or the office and incurring a fixed line charge. This scenario includes making calls via a voice base station, making direct calls between two terminals via the base station and accessing supplementary services provided by an external network. Second, telephones can connect directly to other telephones for the purpose of acting as a ~Swalkie-talkie~T or handset extension. Referred to as the intercom scenario , the connection incurs no additional charge. Third, the telephone may act as a cellular phone connecting to the cellular infrastructure and incurring cellular charges. The cordless and intercom scenarios use the same protocol stack, which is shown in Figure 5. The audio stream is directly connected to the Baseband protocol indicated by the L2CAP bypassing audio arrow.

Fig 3:Protocol Stack for Cordless Phone and Intercom Scenarios

Ultimate Headset

The headset can be wirelessly connected for the purpose of acting as a remote device~Rs audio input and output interface. The headset increases the user~Rs freedom of movement while maintaining call privacy. A common example is a scenario where a headset is used with either a cellular handset, cordless handset, or personal computer for audio input and output. The protocol stack for this usage model is depicted in Figure 6. The audio stream is directly connected to the Baseband protocol indicated by the L2CAP bypassing audio arrow. The headset must be able to send AT-commands (Attention commands) and receive result codes. This ability allows the headset to answer incoming calls and then terminate them without physically manipulating the telephone handset.

Fig 4: Ultimate Headset Protocol Stack

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블루투스 1.1 1.0B 규격에서 발견된 많은 오류들이 수정되었다. 비암호화된 채널들에 대한 지원이 더해졌다. Received Signal Strength Indicator(RSSI)

블루투스 1.2 이 버전은 1.1과 호환되며 다음과 같은 큰 발전이 있었다. Adaptive Frequency-Hopping spread spectrum(AFH), 호핑 시퀀스에서 붐비는 주파수를 피함으로써 라디오 주파수 간섭에 대한 저항력을 향상시켰다. 실사용에서의 향상된 전송속도 호스트 컨트롤러 인터페이스(HCI)가 3-wire UART를 지원한다. HCI가 어플리케이션을 위해 timing information에 액세스한다.

블루투스 2.0 이 버전은 1.x버전들과 호환된다. 가장 큰 개선사항은 2.1 Mbit/s의 Enhanced Data Rate(EDR)이다.

관련자료 및 최신동향기사

카테리니행 열차는 8시에 떠나가네/ 11월은 내게 영원히 기억 속에 남으리
내 기억 속에 남으리/ 카테리니행 기차는 영원히 내게 남으리
함께 나눈 시간들은 밀물처럼 멀어지고/ 이제는 밤이 되어도 당신은 오지 못하리
당신은 오지 못하리/ 비밀을 품은 당신은 오지 못하리
기차는 멀리 떠나고 당신 역에 홀로 남았네/ 가슴 속에 이 아픔을 남긴 채 앉아만 있네
남긴 채 앉아만 있네/ 가슴 속에 이 아픔을 남긴 채 앉아만 있네

<기차는 8시에 떠나네>가사 전문(신경숙 번안)