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SIP-based VoIP Deployment in Taiwan

Aaron Solomon(a.k.a. Dr. Quincy Wu in Taiwan)

TWARENsolomon@ipv6.club.tw

2004.01.29

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Outline

• Introduction to TWAREN• NTP SIP-based VoIP Platform• Plans of VoIP Working Group• Prototypes of Some Utilities

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TWANREN• TWAREN - TaiWan Advanced Research &

Education Network– http://www.twaren.net/English/index.htm

• Dual physical circuits & Three network systems– Production Network

• Provide common academic usage• Provide usual utility

– Research Network• Provide advanced tech. (IPv6, MPLS, Multicast…)• Backup with Production Network

– Optical Network• Provide layer1 provisioning

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Backbone Network• TWAREN Backbone Network topology

（Provided by dual carriers CHTCHT,, EBTEBT）

Hsinchu10G*2

10G

10G

Tainan

Taipei

10G*2

10G*2

CHTCHTTainanTaiChung

CoreCore

TaiChungHsinchu

TainanHsinchu

TainanTaipei

HsinchuTaipei

EBTEBT

Bandwidth of each link is 10 G

Taichung

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•• Aggregated bandwidthAggregated bandwidth–– Backbone: 80GBackbone: 80G–– Regional: 145GRegional: 145G–– Dark fiber: 6Dark fiber: 6

POP (Point of Presence)

TaichungTaichung

2020GG

2020GG

2020GG

10G10G

10G10G

TaipeiTaipei

ASNCU

NCHU

NCNU

CCU

NCKU

NSYSU

NTU

NDHUNTHU

1010GG5G5Gfiberfiber1010G orG orfiberfiber

TainanTainan

HsinchuHsinchuNCTU

1010GG

1010GG

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VoIP on TWAREN• Why should TWAREN promote VoIP

– VoIP is convenient.– VoIP to Internet2 schools are free.– VoIP has hot research topics.– VoIP enables rich services.

• How should TWAREN promote VoIP– TWAREN has good QoS infrastructure.– TWAREN supports end-to-end performance

measurement.– TWAREN runs a conference bridge.– TWAREN provides a transition mechanism from H.323

to SIP.

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NTP VoIP Platform

NTU PBX

Phone31842

Phone31924

Phone59237

Phone59238

SIP Phone0944003005SIP Phone

0944003004

PSTN Gateway

SIP Phone0944002002

Phone3213

Phone4100

Phone4454

Phone6818

Phone02-87730600

StationInterface

StationInterface

StationInterface

StationInterface

Phone03-5912312

Admin Console

SIP Phone0944003003

SIP Phone0944002003

Hsinchu

Taipei

TrunkInterface

03-5712121

02-23630231Trunk

Interface

Call Server

TANet

NCTU

PSTN

NTUEdge Router

Edge Router

NCTU PBX

Softphone WLANAP

Call Server PSTN GatewayWGSN

•IPTel SER•ITRI Call Server

•Cisco 2621GW•ITRI PSTN GW

•Pingtel•Snom•Cisco

•Siemens•Microsoft•ITRI

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Academic ResearchesSupport academic

researches on NTP VoIP Platform

• NTU: SIP Signaling Performance Evaluation on SCTP

• NTHU: Secure RTP and Location Privacy on VoIP System

• NDHU: Voice over IP study on All IP networks

• NCKU: DNS/ENUM Automatic Updating Mechanism

• NCTU: NAT Traversal & WGSN Project for Integrated Wireless VoIP Services

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Numbering Plan

• GDS (Global Dialing Scheme)– 886-3-5712121-59238

• SIP URI– sip:solomon@ipv6.club.tw

• ENUM– 0944020678

• "A rose by any other name would smell as sweet."- William Shakespeare

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TWAREN VoIP Working Group

• TWAREN is chartering a VoIP WG.• Proposed projects in 2004 includes:

– SIP.edu– SIP/H323 Gateway + Conference Bridge– E2E Performance Measurement + Trouble-Ticket

System– NAT Traversal (STUN, TURN, UPnP, IPv6)– Instant Message & Presence Service– BoD for VoIP

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SIP.edu – Phase 1

SIPProxy

DNS SIP-PBXGateway PBX

INVITE (sip:dbaron@mit.edu)

INVITE(sip:21232@gw.mit.edu)

DNS SRV query sip.udp.mit.edu

telephoneNumberwhere mail=”dbaron@mit.edu”

PRI / CAS

CampusDirectory

SIP User Agent

Dennis’ Phone

Phase 1:Provide SIP connectivity to all users on a campus through the PBX

Source: SIP.edu Project of Internet2 VoIP Working Group

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SIP.edu – Phase 2

SIPProxy

DNS

DNS SRV query sip.udp.mit.edu

REGISTER(Contact: 18.142.2.4)

SIPRegistrar

INVITE (sip:dbaron@18.142.2.4)

INVITE (sip:dbaron@mit.edu)

SIP User Agent

Dennis' SIP Phone

Phase 2:Begin to supportUA registration so calls can be IPend-to-end

locationDB

If Dennis has registered, ring his SIP phone;Else, call his extension through the PBX.

Source: SIP.edu Project of Internet2 VoIP Working Group

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ApplicationsDeveloper

SystemAdministrator

LANAdministrator

CampusNetworking

Gigapop Gigapop

Backbone

CampusNetworking

LANAdministrator

SystemAdministrator

ApplicationsDeveloper

How do you solvea problem along a path?Everyone says it isworking fine!

Hey, this is not working right!

The computerIs working OK

Talk to the other guys

Everything isOK

No othercomplaints

The network is lightly loaded

All the lightsare green

We don’t seeanything wrong

Looks fine

Others are getting in ok

Not our problem

E2E Problems

Source: “End to End Performance Initiative”, Eric Boyd

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Software Under Development

1. SIP UA with NAT Traversal2. IPv6 SIP UA3. IPv6 SIP Packet Analyzer

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• NBEN UA runs on Windows 2000/XP/2003.

• Both signaling and media data are transported on UDP.– SIP: port 5060– RTP: port 9000

• Support audio codec:– G.711 (64Kbps)– G.729 (8Kbps)– G.723.1 (6.3Kbps)

• Support STUN (RFC 3489) for NAT traversal.

Project 1: SIP User Agent for NAT Traversal

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Project 2: IPv6 SIP UA

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Project 3: IPv6 SIP AnalyzerSIP Signaling Flow

Traffic Statistics RTP Monitor & Playback

SIP Packets Capturing

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SIP Message Contents

SIP Message Contents

Packet Analysis as Ethereal

SIP Session

SIP Request/Response

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SIP Signaling Flow (1)IPv6 address of caller/callee

Green arrow isSIP Response

Blue arrow isSIP Request

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SIP Signaling Flow (2)Dashed arrow represent a conjectured signal

(according to the Via/Route header field)

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RTP Monitor & Playback• Original purpose is to help assessing the packet loss rate of RTP

traffic.• It turns out to a tool to demonstrate the importance of encryption.

RTP Streams

RTP Stream Playback

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Statistics Data

Throughput (packet/s)

IPv6 Voice Stream

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Conclusion• By establishing a nation-wide VoIP testbed, TWAREN

wishes to promote the convergence of voice and data services and encourage advanced researches in Taiwan.

• SIP coverage in 2003 is approximately 50,000 users. NTP plans to double the coverage in 2004.

• There are prototypes of NAT traversal solutions and IPv6 clients. Larger deployment is needed to verify these techniques.

• VoIP WG needs to closely work with Measurement WG and Multimedia WG to leverage our efforts. It is also critical to consolidate our on-going projects in accordance with Internet2 VoIP Working Group.