1 VoIP 개요

l 의미 : 현존하는 Data network을 통해 voice와 fax를 송수신할 수 있는 기술

l 장점

- 원거리 통신이나 국제전화시 가격 절감

- 기존 망 활용가능(Packet switching에 의한 대역폭 공유)

- 인터넷과의 결합으로 다양한 서비스 가능

- 기존 인터넷 서비스(www, mail등)와 결합 가능

l 발생배경

- 과거 : 통신을 위해 PSTN(Public Switched Telephone Network)을 사용했으나, communication의 필요성이 커짐에 따라, 많은 회사들이 고비용을 지불하고 독자적인 전용선을 설치하기 시작했다.

l 구현방식

- 음성데이타를 64Kbps급 디지털로 변환한 후 압축을 수행하고 이를 IP 데이타에 RTP헤더로 감싸 안은 후 기존 데이터망에 수용되는 방식

l 구현에 있어서 중요한 이슈

- Burst한 데이터 트래픽(예를 들어 FTP)에 영향받지 않는 음질 = QoS 필요

l 구현에 필요한 요소

q Digital Signal and Packet Processing

–디지털신호 생성 : Voice Coding, Trans-coding, Silence Removal, Echo canceling

–전송기술 : Packetization (Serialization), Buffer smoothing

q Addressing/Naming & Routing

–E.164, IPv4/IPv6, DNS and Directory Service

q Call Processing

–Signalling Channel Mode (CAS, CCS)

–Call Response Time

–Call Processing Capacity

q QoS/CoS

–End-to-End Delay/Jitter

–Packet Loss

- echo

q Management

–Security

–Accounting/Charging/Billing

q Interworking

–Bearer Interworking

–Service (IN) Interworking (080-xxx, 700-xxx, etc)

q Mobility

–Terminal and Service mobility

q Services

–Basic Services (PC-to-Phone, Phone-to-PC , Phone-to-Phone, PC-to-PC)

–Supplementary Services (UMS, SMS, Internet Call Waiting, Click-to-Dial etc)

q Interoperability

l Protocol Stack

l Packet voice processing

l 표준 Codec의 특성

l Delay

l Network Architecture

l signaling and switching

- 단말 또는 게이트웨이 간의 연결 설정

- 표준화 기구에서 표준 프로토콜 설정

2 SIP(Session Initiation Protocol)

- An application layer signaling protocol that defines initiation, modification and termination of interactive, multimedia communication sessions between users

l Architecture

- User Agent : An application that initiates, receives and terminates calls

UA Client/Server

- Proxy Server :

- Location Server : used by a SIP redirect or proxy server to obtain information about a called party’s possible location(s).

- Redirect Server : •A server that accepts a SIP request, maps the address into zero or more new addresses and returns these addresses to the client. Unlike a proxy server, the redirect server does not initiate its own SIP request. Unlike a user agent server, the redirect server does not accept or terminate calls

- Registrar Server : typically co-located with a proxy or redirect server and may offer location services

l SIP Addressing : user@host

l Process for Establishing Communication

1.Registering, initiating and locating the user.

2.Determine the media to use – involves delivering a description of the session that the user is invited to.

3.Determine the willingness of the called party to communicate – the called party must send a response message to indicate willingness to communicate – accept or reject.

4.Call setup.

5.Call modification or handling – example, call transfer (optional).

6.Call termination.

l Simplified SIP call setup and teardown

3 H.323

- Packet based multimedia communications systems

l Components

- End devices : 전화, PC, 네트워크의 끝단, Audio 반드시 지원(G.711 기본, G.723.1 low bit rate voice/video, G.729 low bit rate voice), video(H.261 기본)/data 지원은 옵션사항, control functions(H.245, H.225), media packet을 위한 RTP

- Gateways : 양 끝단 중 한 곳이상에서 전화를 사용한다면 필요함. 인터넷으로 전송하기 위한 포맷변경 역할 담당, 옵션항목

- Gatekeepers/proxies : centralized call management 담당(call admission control, bandwidth management, address translation, authentication, user location, 다른 모든 component 관리), 옵션항목

- Multipoint conference units : 3개 이상의 endpoints 사이의 conferencing 지원,

l 서로서로 signaling and voice transporting protocol로 통신함