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SIP-based VoIP Deployment in
Taiwan
Aaron Solomon
(a.k.a. Dr. Quincy Wu in Taiwan)
TWAREN
[email protected] 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
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Backbone Network
•
TWAREN Backbone Network topology（Provided by dual carriers CHTCHT,, EBTEBT）
Hsinchu 10G*2 10G 10G Tainan Taipei 10G*2 10G*2 CHT CHT Tainan TaiChung Core Core TaiChung Hsinchu Tainan Hsinchu Tainan Taipei Hsinchu Taipei EBT EBT
Bandwidth of each link is 10 G
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•
• Aggregated bandwidthAggregated bandwidth
–
– Backbone: 80GBackbone: 80G
–
– Regional: 145GRegional: 145G
–
– Dark fiber: 6Dark fiber: 6
POP (Point of Presence)
Taichung Taichung 20 20GG 20 20GG 20 20GG 10G 10G 10G 10G Taipei Taipei AS NCU NCHU NCNU CCU NCKU NSYSU NTU NDHU NTHU 10 10GG 5G 5G fiber fiber 10 10G orG or fiber fiber Tainan Tainan Hsinchu Hsinchu NCTU 10 10GG 10 10GG
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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 Phone 31842 Phone 31924 Phone 59237 Phone 59238 SIP Phone 0944003005 SIP Phone 0944003004 PSTN Gateway SIP Phone 0944002002 Phone 3213 Phone 4100 Phone 4454 Phone 6818 Phone 02-87730600 Station Interface Station Interface Station Interface Station Interface Phone 03-5912312 Admin Console SIP Phone 0944003003 SIP Phone 0944002003 Hsinchu Taipei Trunk Interface 03-5712121 02-23630231 Trunk Interface Call Server TANet NCTU PSTN NTU Edge Router Edge Router NCTU PBX Softphone WLAN APCall Server PSTN Gateway WGSN
•IPTel SER
•ITRI Call Server
•Cisco 2621GW •ITRI PSTN GW •Pingtel •Snom •Cisco •Siemens •Microsoft •ITRI
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Academic Researches
Support 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:[email protected]
• 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
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SIP.edu – Phase 1
SIP Proxy DNS SIP-PBX Gateway PBX INVITE (sip:[email protected]) INVITE (sip:[email protected]) DNS SRV query sip.udp.mit.edu telephoneNumber where mail=”[email protected]” PRI / CAS Campus Directory SIP User AgentDennis’ Phone Phase 1: Provide SIP connectivity to all users on a campus through the PBX
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SIP.edu – Phase 2
SIP Proxy DNS DNS SRV query sip.udp.mit.edu REGISTER (Contact: 18.142.2.4) SIP Registrar INVITE (sip:[email protected]) INVITE (sip:[email protected]) SIP User AgentDennis' SIP Phone
Phase 2: Begin to support UA registration so calls can be IP end-to-end location DB
If Dennis has registered, ring his SIP phone; Else, call his extension through the PBX.
13 Applications Developer System Administrator LAN Administrator Campus Networking Gigapop Gigapop Backbone Campus Networking LAN Administrator System Administrator Applications Developer
How do you solve
a problem along a path?
Everyone says it is working fine!
Hey, this is not
working right!
The computer Is working OK
Talk to the other guys Everything is
OK
No other complaints
The network is lightly loaded All the lights
are green We don’t see anything wrong Looks fine Others are getting in ok Not our problem
E2E Problems
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Software Under Development
1. SIP UA with NAT Traversal
2. IPv6 SIP UA
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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 3: IPv6 SIP Analyzer
SIP Signaling Flow
Traffic Statistics RTP Monitor & Playback SIP Packets Capturing
18 SIP Message Contents SIP Message Contents
Packet Analysis as Ethereal
SIP Session
SIP
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SIP Signaling Flow (1)
IPv6 address of caller/callee
Green arrow is SIP Response Blue arrow is SIP 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
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Statistics Data
Throughput (packet/s) IPv6 Voice Stream23
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.
