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VoIP. Onno W. Purbo [email protected]. Arah Perkembangan. Philosophy. The old world Box reliability Centralized The new world Network availability Distributed. References. http://www.cisco.com http://www.lucent.com http://www.xl.com http://www.quicknet.net http://openh323.org - PowerPoint PPT Presentation
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VoIP Onno W. Purbo [email protected]
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Page 1: VoIP

VoIP

Onno W. [email protected]

Page 2: VoIP

Arah Perkembangan

Page 3: VoIP

Philosophy The old world

Box reliability Centralized

The new world Network availability Distributed

Page 4: VoIP

References http://www.cisco.com http://www.lucent.com http://www.xl.com http://www.quicknet.net http://openh323.org http://openphone.org http://www.pulver.com

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Outline End User Perspective Conceptual Framework Next Generation Telco ..

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End User - Internet Telephone

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Komponen VoIP / Menit Telkom Rp. 75 / menit ISP Rp. 75 / menit Hook-off Service

http://www.net2phone.com

Delay <250ms

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Biaya SLI / Menit (1US$=Rp8000)

Country Peak Normal

Disc VoIP

US 9900 8300 6230 1350

Singapore

6780 5650 4240 2550

Taiwan 9960 8300 6270 4070

Jepang 11286 9400 7050 2310

Hong Kong

9960 8300 6230 2710

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Solusi Kantor Kecil / Rumah Satu (1) Saluran VoIP

Internet PhoneJACK US$150 Internet LineJACK US$220 Internet PhoneCARD US$170

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Komunitas WARNET [email protected] Free subscribe:

[email protected]

Komunitas e-commerce [email protected] [email protected] [email protected]

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Kebutuhan WARNET Line telepon koin Leased line analog

Regional Network / DIVNET Akses ke box JAMUS V.35 / RS.449

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WARNET “tanpa” Telkom

WARNET ke Internet Satellite DirectPC – US$200 / card KabelVision – Rp. 300.000 / bulan

Jaringan WARNET 2-11 Mbps 2.4 GHz US$250 / card

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Teknologi VoIP sederhana

Page 14: VoIP

Internet Telephone

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Dialer

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Video Conference

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Internet PhoneJACK

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Connection PhoneJACK

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Internet PhoneCARD

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Voice Quality Comparison

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Solusi Gateway Sederhana

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VoiceTronix (8 port)

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VoIP untuk WARNET Investasi WARNET+ Invest Card DSP ~US$150 / kanal. Estimasi ke US biaya / menit

US$0.13 + Rp. 75 + Rp. 75 Estimasi ke US biaya SLI 001

Rp. 9900-8300 / menit

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Issue Desktop VoIP Solution Ease of Installation. Infrastructure. Flexibility. Compatibility. Ease-of-Use. Quality. Cost Growth Path. Base for Applications.

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Aplikasi Desktop VoIP IntraNet Communication Remote Call Forwarding Distributed Call Center Virtual PBX Line Doubling Personal Iphone Calling

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Personal Iphone Calling

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Intranet Communications

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Remote Call Forwarding

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Distributed Call Center

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Virtual PBX

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Line Doubling

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Mobile User

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Mobile User Internet Locator Service (ILS) Mapping Name Number

Dynamic IP

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Internet Locator Service (ILS) http://www.icq.com

ILS & chat system http://www.tzo.com

Dial your DNS name.

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Conceptual Framework H.323

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H.323 Benefit Codec Standards. Interoperability. Network Independence. Platform & Application Independence. Multipoint Support. Bandwidth Management. Multicast Support. Flexibility. Inter-Network Conferencing.

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H.323 mengatur .. Video Codec (H.261 dll). Audio Codec (G.711, G.723 dll). Data channel (T.120 dll) e-board dll. System Ctrl Unit (H.245, H.225.0 dll). H.225.0 layer – error detection, error

correction and frame sequencing.

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Open Standard

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Page 40: VoIP

Komponen H.323 Terminal Gateway Gatekeeper Multipoint Control Unit (MCU)

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Terminal H.323 Provide Real-Time. Provide Two-way communications. H.245 – nego channel usage &

capabilities. Q.931 – Registration/Admission/Status

(RAS). T.120 – Data Conferencing Protocols.

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Page 43: VoIP

Komponen H.323 H.323 Gateway

Interface H.323 terminal – WAN terminal. H.323 Gatekeeper

Address Translation (E.164 – IP) Admission Control. Authentication. Bandwidth Control. Zone Management. Call Control & Management

(e.g. VoIP – PSTN)

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H.323 Gateway Optional! Translation function between H.323

conferencing endpoints and other terminal types.

Translation between transmission formats (i.e. H.225.0 to H.221).

Translation between communications procedures (i.e. H.245 to H.242).

Translation between audio and video codecs. Performs call setup and clearing on both the

LAN side and the PSTN side.

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H.323/PSTN Gateway

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Gatekeeper The most important component

asct as a virtual switch of an H.323 enabled network.

It acts as the central point for all calls within its zone and provides call control services to registered endpoints.

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Gatekeeper Address translation from LAN

aliases for terminals and gateways to IP or IPX addresses (e.g E.164 – IP)

Bandwidth management. Route H.323 calls. Bill for calls.

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Gatekeeper Zone

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Isu Quality of Service (QoS) Kekurangan Bandwidth Delay terlalu lama. Jitter / Variasi delay.

Page 50: VoIP

QoS Resource Reservation Protocol (RSVP) Real Time Protocol (RTP) Compressed Real Time Protocol (CRTP)

Traffic Shaping Queuing: Custom Queuing, Weighted Fair

Queuing. Congestion Control: Random Early Detection. Effisiensi Jaringan: link fragmentation &

interleaving.

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Metoda KompresiKompresi Kbps MIPS ms MOSG.711PCM 64 0.34 0.12

54.1

G.726 ADPCM 32 14 0.125

3.85

G.728 LD-CELP 16 33 0.625

3.61

G.729 CS-ACELP 8 20 10 3.92G.729 x2 Encoding 8 20 10 3.27G.729 x3 Encoding 8 20 10 2.68G.729a CS-ACELP 8 10.5 10 3.7G.723.1 MPMLQ 6.3 16 30 3.9G.723.1 ACELP 5.3 16 30 3.65

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MOS vs. DSP Price

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Intermediate Solution VoIP

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Interkoneksi Suara-Data-Suara

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Interkoneksi PC-Telepon

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Gatekeeper Lucent PacketStar IP Gatekeeper

23.500 DSO 2.300.000 calls / day (one zone) 7.800.000 calls / day (multi zone)

Cisco CallManager Windows NT

Quicknet H.323 Internet SwitchBoard

Page 57: VoIP

Gateway Lucent PathStar

80.000 POTS 10.000 xDSL

Cisco DE-30+ 30 voice on E1

Cisco AT-2, 4, 8 Analog loop RJ-11, 2, 4, 8 POTS.

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Next Generation Telco

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Split Data & Voice Network

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Converged Enterprise Network

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Philosophy The old world

Box reliability

The new world Network availability

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Network Availability

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Network Availability Redudancy at many levels. Advanced Layer 3 Protocols:

Hot Standby Routing Protocol (HSRP). Open Shortest Path First (OSPF). Enhanced Interior Gateway Routing Protocol

(EIGRP). Multicast Routing:

Protocol Independent Multicast (PIM). Distance Vector Multicast Routing Protocol

(DVMRP).

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Campus Network

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Page 67: VoIP

How? – Detailed ..

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End-to-End Architecture

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Open Architecture H.323 Simple Gateway Control Protocol (SGCP) Media Gateway Control Protocol (MGCP) Session Initiation Protocol (SIP) Telephony Application Programmable

Interface (TAPI) Java Telephony Application

Programmable Interface (JTAPI)

Page 70: VoIP
Page 71: VoIP

Low Cost of Ownership

Page 72: VoIP
Page 73: VoIP

Hierarchical WAN Design the network must support and supply the

prerequisite QoS features. Hierarchical model to allow the most cost-

effective platforms. As the requirements for data traffic outstrip

those of voice, voice bandwidth decreases & lowering costs.

WAN links to support the minimum requirements for data + voice and video. When other applications are quiescent, the bandwidth is available for data.

Page 74: VoIP
Page 75: VoIP

WAN QoS-Enabling Tech.

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Page 77: VoIP

Classification Layer 3 (IP)

IP Precedence; VoIP 5. Differentiated Services Code Point

(DSCP) Layer 2 (MAC) schemes

Voice RTP Voice UDP Voice Port 16383 - 32767

Page 78: VoIP

Prioritizing Voice Traffic

Priority Queuing (PQ) Data Traffic

weighted fair queuing (WFQ) class-based weighted fair queuing

(CBWFQ)

Page 79: VoIP

Prioritizing

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Link-Efficiency Techniques Voice Compression (e.g. G.723). Compressed RTP

40 byte IP+UDP+RTP 2-4 bytes. Voice Activity Detection (VAD). Link Fragmentation & Interleaving.

Frame Relay – FRF.12 Serial Link – Multilink PPP (MLP).

Page 81: VoIP

Traffic Shaping Multiple access nonbroadcast

media such as ATM and Frame Relay where the physical access speed varies between two end points.

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Page 83: VoIP

Issue QoS - Campus Network Aggregation points where many

links aggregate into an uplink. aggregation points where speed

mismatches occur.

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Page 85: VoIP

QoS Campus Network The primary concern is buffer

management and the ability to provide the required queuing schemes to ensure voice gets the appropriate bounds for loss, delay, and jitter.

Page 86: VoIP

QoS – Strategies Classification is a key attribute. Queuing techniques. Buffer management is also used to

ensure that excess jitter is not added for voice traffic.

To combat this, the multilayer LAN switches provide either two queues or a single queue with definable drop thresholds to bound delay.

Page 87: VoIP

Application IP Telephony Unified Messaging IP Contact Center

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VoIP

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Unified Messaging

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Page 91: VoIP

IP Contact Center

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Page 93: VoIP
Page 94: VoIP

Migration Strategies

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Terima Kasih

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