Post on 03-Oct-2020
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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.