Mengenal

Teknologi VOIP
Deris Stiawan

Data Networking

M 19 ainf 60 ram -19 e 70 s

M 197 ini 0-19 80s

PC/LA 1980- N 1990s

Internet
1990-21st Century

Voice Networking

Xs B g P 70s o l a 19 An 6019

Xs B P al 90s t i Dig 70-19 19

Xs B P 0s PC -200 0 199

Pendahuluan

Komunikasi Data
Modem Internet Modem

Workstation

Server

Komunikasi Suara
PSTN

New world applications

Headquarters (or SP)

Branch Office

IP

Application Servers

VOIP
Teknologi komunikasi suara dengan melewatkan voice pada jaringan internet protocol (IP) Membungkus (encapsulated) voice dengan encapsulations protocol jaringan IP Dilewatkan dengan paket data lainnya di jaringan Internet Menggunakan sebuah protocol

Dilewatkan pada jaringan Internet bukan pada jaringan PSTN PSTN = Public Switch Telephone Network. Jaringan infrastruktur telphone yang diselengarakan teleco (telkom

Mengapa voip

Tidak dikenakan biaya rekening telpon Satu link bisa digunakan untuk data n voice ekonomis Keutungan yang lebih dari PSTN

Voice Mail Audioconferencing Unifed Messaging Directory Services Presence Video Conferences Click to dial Dial Plans Soft clients

Komponen VOIP
USER AGENT PROXY PROTOCOL CODEC (Coder Decoder)

USER AGENT

Softphone SIP

SJphone X-Lite Idefisk IaxLite Netmeeting

Softphone IAX

Softphone H.323

IP Phone

Berbentuk seperti telepon biasa, terhubung langsung ke jaringan IP (tidak melalui perangkat lain) Berbentuk seperti telepon genggam, menggunakan kabel\dan terhubung ke PC melalui port USB Port FXS terhubung ke telepon biasa Port FXO terhubung ke PSTN langsung atau melalui PBX ITG dengan port FXS saja

USB Phone/ Wireless/ Bluetooth

Internet Telephony Gateway (ITG)

Analog Telephone Adaptor (ATA)

PROXY

Softswitch
Open

Source Asterisk (http://www.asterisk.org) OpenSER (http://www.openser.org) SER (http://www.iptel.org/ser/) Yate (http://yate.null.ro)

Non Open Source

Axon

(http://www.nch.com.au/pbx/) OnDO SIP Server (http://www.brekeke.com)

PROTOCOL

IETF
SIP

ITU-T
H.323

Asterisk
IAX

VoIP Protocol
H.323

Description
ITU standard protocol for interactive conferencing. Evolved from H.320 ISDN standard. Flexible, complex. Emerging standard for Public Switched Telephone Network (PSTN) gateway control, thin device control.

Media Gateway Control Protocol (MGCP)

Session Initiation Protocol (SIP)

IETF protocol for interactive and non-interactive conferencing. Simpler, but less mature, than H.323. IETF standard media streaming protocol.

Real-Time Transport Protocol (RTP)

Real-Time Transport Control Protocol (RTCP)

Protocol that provides out-of-band control information for an RTP flow.

Low Latency Queuing (LLQ)

Priority queuing technique that uses priority queuing-class-based weighted fair queuing (PQ-CBWFQ).

CODEC

Open Source dan Free

GSM Ilbc Speex G711

Licensed
G729 G723

Bandwidth
Coding Algorithm Voice Bandwidth (Kbps) Coded Frame Size (bytes) Voice Frames in VoIP Packet IP+UDP+R TP Header Size (bytes) Layer 2 Technology Used Layer 2 Header Size (bytes) Total Bandwidth Required

G.711 G.711 G.711 G.729 G.729 G.729

64 64 64 8 8 8

160 160 160 20 20 20

2 2 2 2 2 2

40 40 2 (cRTP) 40 40 2 (cRTP)

Ethernet II MLPPP MLPPP Ethernet II (Cisco Default) MLPPP MLPPP

26 7 7 26 7 7

90.4 82.8 67.6 34.4 26.8 11.6

Variant VoIP
PC ke PC (YM!, Skype, Merdeka,) Telphone ke Telphone (memerlukan geatekeeper, ITG) Telphone (HP) ke operator

Quality of Services (QoS)

Classification Queuing Policing Marking Shaping

Identify and Split Traffic into Different Classes

Discard Misbehaving Traffic to Maintain Network Integrity

Mark Traffic According to Behavior and Business Policies

Prioritize, Protect and Isolate Traffic Based on Markings

Control Bursts and Conform Traffic

Legal / Ilegal ?
Harus ada izin dari POSTEL sebagai penyedia layanan VoIP di indonesia Legal jika digunakan untuk internal perusahaan dan penggunaan pribadi (tidak mengkoneksikan ke operator) Monopoli operator pemerintah

Issues VoIP
Bandwidth Jaringan Keamanan Solusi Teknologi ITG Pelatihan Policy

Implementasi VOIP