SIP-based VoIP Deployment in Taiwan

Transcription

1 SIP-based VoIP Deployment in Taiwan Aaron Solomon (a.k.a. Dr. Quincy Wu in Taiwan) TWAREN

2 Outline Introduction to TWAREN NTP SIP-based VoIP Platform Plans of VoIP Working Group Prototypes of Some Utilities 2

3 TWANREN TWAREN - TaiWan Advanced Research & Education Network 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 3

4 Backbone Network TWAREN Backbone Network topology (Provided by dual carriers CHT, EBT) Hsinchu Bandwidth of each link is 10 G 10G*2 Taipei Core Core 10G 10G*2 Taipei Hsinchu 10G*2 Taipei Tainan Taichung Hsinchu TaiChung Hsinchu Tainan Tainan TaiChung 10G Tainan CHT EBT 4

5 POP (Point of Presence) Aggregated bandwidth Backbone: 80G Regional: 145G Dark fiber: 6 NCHU NCNU CCU NCTU NTHU NCKU NSYSU NCU 10G 10G 20G 20G AS Taichung Hsinchu 20G Tainan Taipei NTU NDHU 10G 10G 10G 5G fiber 10G G or fiber 5

6 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. 6

7 Siemens Microsoft ITRI Softphone WLAN AP Edge Router NTP VoIP Platform IPTel SER ITRI Call Server WGSN Call Server NCTU Cisco 2621GW ITRI PSTN GW PSTN Gateway Station Interface NCTU PBX Station Interface Trunk Interface Phone Hsinchu TANet Edge Router SIP Phone Call Server SIP Phone SIP Phone NTU PSTN Gateway Phone Station Interface Phone NTU PBX Station Interface Phone Phone Trunk Interface PSTN Taipei Pingtel Snom Cisco Admin Console SIP Phone SIP Phone Phone 3213 Phone 4100 Phone 4454 Phone 6818 Phone

8 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 8

9 Numbering Plan GDS (Global Dialing Scheme) SIP URI ENUM "A rose by any other name would smell as sweet." - William Shakespeare 9

10 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 10

11 SIP.edu Phase 1 DNS SRV query sip.udp.mit.edu SIP User Agent INVITE (sip:dbaron@mit.edu) Phase 1: Provide SIP connectivity to all users on a campus through the PBX DNS SIP Proxy INVITE (sip:21232@gw.mit.edu) SIP-PBX Gateway PRI / CAS PBX telephonenumber where mail= dbaron@mit.edu Campus Directory Dennis Phone 11 Source: SIP.edu Project of Internet2 VoIP Working Group

12 SIP.edu Phase 2 DNS SRV query sip.udp.mit.edu SIP User Agent INVITE (sip:dbaron@mit.edu) If Dennis has registered, ring his SIP phone; Else, call his extension through the PBX. Phase 2: Begin to support UA registration so calls can be IP end-to-end DNS SIP Proxy INVITE (sip:dbaron@ ) location DB SIP Registrar REGISTER (Contact: ) Dennis' SIP Phone 12 Source: SIP.edu Project of Internet2 VoIP Working Group

13 E2E Problems Applications Developer Hey, this is not working right! LAN Administrator Not our problem Talk to the other guys Others are getting in ok LAN Administrator Applications Developer System Administrator The computer Is working OK Campus Networking Gigapop How do you solve a problem along a path? Everyone says it is working fine! Everything is OK No other complaints Backbone Campus Networking Gigapop All the lights are green We don t see anything wrong The network is lightly loaded System Administrator Looks fine 13 Source: End to End Performance Initiative, Eric Boyd

14 Software Under Development 1. SIP UA with NAT Traversal 2. IPv6 SIP UA 3. IPv6 SIP Packet Analyzer 14

15 Project 1: SIP User Agent for NAT Traversal 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 (6.3Kbps) Support STUN (RFC 3489) for NAT traversal. 15

16 Project 2: IPv6 SIP UA 16

17 Project 3: IPv6 SIP Analyzer SIP Packets Capturing SIP Signaling Flow Traffic Statistics RTP Monitor & Playback 17

18 Packet Analysis as Ethereal SIP Session SIP Request/Response SIP Message Contents 18

19 SIP Signaling Flow (1) IPv6 address of caller/callee Blue arrow is SIP Request Green arrow is SIP Response 19

20 SIP Signaling Flow (2) Dashed arrow represent a conjectured signal (according to the Via/Route header field) 20

21 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 21

22 Statistics Data IPv6 Voice Stream Throughput (packet/s) 22

23 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 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. 23