1 Integrating 3G and WLAN Services in NTP SIP-based VoIP Platform Dr. Quincy Wu National Telecommunications Program Office

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Presentation transcript:

1 Integrating 3G and WLAN Services in NTP SIP-based VoIP Platform Dr. Quincy Wu National Telecommunications Program Office

2 Outline Introduction to NTP NTP SIP-based VoIP Platform NAT Traversal Integrating WLAN and GPRS network Future Work

3 NTP National Telecommunications Development Program –Established by the National Science Council (NSC) in May, –Phase I: from 1998 to –Total budget: 350 million US$ in 5 years. Program Focuses –Wireless Communication –Broadband Internet –Application Services Mission –To conduct R&D in advanced telecommunication technologies, for the promotion and development of telecommunication industry.

4 NTP VoIP Platform IPTel SER CCL Call Server Cisco 2621GW CCL PSTN GW Pingtel Snom Cisco Siemens Microsoft CCL

5 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

6 Simple Traversal of UDP Through Network Address Translators (STUN)

7 Common Scenario in VoIP UA1, UA3: User Agent behind NAT. UA2: SIP device outside NAT sip.ipv6.club.tw NAT UA1 UA2 IPv6 only Proxy Server UA3 IPv6 only

8 Network Address Translator Computer A IP: Port: 80 Computer B IP: Port: 80 NAT Public Internet IP: Port: IP: Port: Mapping Table : : DHCP Server DHCP Client PPPoE Client Private NIC Public NIC

9 The Problem (1/2) Because of its private address, the Via header and Contact address in SIP messages sent by UA1 are incorrect. –With incorrect Via header, responses of messages sent by UA1 cannot be routed back. –With incorrect Contact address in REGISTER messages, call server cannot inform UA1 the incoming calls. UA1 can only act as a calling party.

10 Incorrect REGISTER Message

11 The Problem (2/2) When UA1 initiate a call, the connection information for media establishment in SDP are also incorrect. –The IP address in SDP is the private IP address on UA1. –UA2 gets a private peer address, the RTP packets from UA2 cannot be routed to UA1. –Media can only be sent from UA1 to UA2.

12 Incorrect Fields in SDP of INVITE Message

13 STUN (RFC 3489) A mechanism for a socket behind NAT(s) to get its mapped (IP,port) on Internet. We build a STUN Server from Vovida’s open source code to answer the queries from STUN clients. –Vovida STUN server 0.8RC Our server is stun.ipv6.club.tw –We also implemented the “Binding Lifetime Discovery” algorithm in RFC 3489 so that a STUN client can determine how frequently it needs to refresh the binding. STUN mechanism in our SIP UA is developed from scratch, instead of the open source code. In addition to calling out, our SIP UA can also receives incoming calls.

14 STUN Server Allow clients to discover if it is behind a NAT, what type of NAT it is, and the public address & port NAT will use. A Very Simple Protocol. Easy to implement. Little computing load. Client IP: Port: 5060 IP: Port: STUN Server Port: 3478 NAT Client wants to receive packets at port 5060 Send a query to STUN server from port 5060 STUN Server receives a packet from port STUN Server sends a response to client, telling him that his public address is port 12345

15 Use STUN for SIP Registration Use port 5060 to send a packet to STUN Server Receive public address & port mapped to client:5060 from STUN Server Fill the SIP register message with client’s public address & port, send to proxy server IP: Port: STUN Server Port: 3478 NAT Proxy Server Port: 5060 REGISTER sip:proxy.club.tw SIP/2.0 Via: SIP/2.0/UDP :56540 From: Solomon To: Solomon … Contact: Solomon … Client IP: Port: 5060

16 Corrected SIP Message

17 Use STUN for RTP Send two STUN queries from RTP port (9000 & 9002) to STUN Server Use replied public address & port in SDP IP: Port: Port: STUN Server Port: 3478 NAT Proxy Server Port: 5060 INVITE … … Content-Type: application/sdp c=IN IP m=audio RTP/AVP m=video RTP/AVP Client IP: RTP Port: 9000 RTP Port: 9002 UA RTP Port: 9000 RTP Port: 9002

18 Corrected SDP

19 WLAN-based GPRS Support Node (WGSN)

20 WGSN (WLAN-based GPRS Support Node) Architecture

21 WGSN Features Service aspects –SIP-based VoIP Access control and security aspects –SIM-based authentication Roaming aspects –Utilizing the standard UMTS mobility management Terminal aspects –A Windows laptop equipped with a smart card reader Naming and address aspects –International Mobile Subscriber Identity (IMSI) Charging and billing aspects –Offline charging and online charging based on Call Detail Record (CDR)

22 WGSN Protocol Stack (a) WGSN Control Plane (b) WGSN User Plane IP EAP-GMM EAP EAP-OL EAP-OL L1 Ethernet IP UDP RADIUS L1 Ethernet UDP RADIUS EAP L1 MTP2 MTP3 SCCP TCAP MAP L1 MTP2 MTP3 SCCP TCAP MAP MS APWGSN HLR EAP-GMM

23 The MS Architecture

24 The WGSN Node Architecture

25 Future Work OSA – Open Service Architecture IMS – IP Multimedia Subsystem IPv6 – IP Version 6

26 Application Service Development Platform in Next Year — Open Service Architecture

27 CSCF OSA HLR (HSS) GPRS/3G IP Multimedia Subsystem (IMS) Phase I (now) Phase II Supported by other projects Roaming ENUM Campus WLAN WLAN + Cellular IMS Architecture WGSN

28 Conclusion By establishing a nation-wide VoIP testbed, NTP wishes to promote the convergence of voice and data services and encourage advanced researches in Taiwan. SIP coverage in this year is approximately 50,000 users. NTP plans to double the coverage in 2004, and hopefully extend the connectivity with SIP.edu project in Internet2. STUN proves to be a good solution for non- symmetric NAT. We are still surveying mechanisms for traversing symmetric NAT. WGSN can be plugged in any 3G core network as being considered as a separate GPRS network. With the adoption of WGSN, it can help mobile operators facilitating the provisioning of WLAN and VoIP services.