無線與頻寬整合之多媒體通訊技術

1. 無線與頻寬整合之多媒體通訊技術 周勝鄰 博士 工研院電通所 網路技術組 組長 2003. 11. 26

2. 課程目的 • 本課程將與您一起探討通訊產業中三個重要之發展趨勢 • IP 電信(VoIP)逐漸成為未來通訊技術之主流, 不僅將對服務產業將發生變化, 而且也將為通訊產品產業帶來新契機;本課程將提供學員對此一領域做初步之介紹 • ENUM為一項非常新的技術, 它將是推動 IP/ PSTN系統與服務整合的最大動力, 同時也將對網際網路之應用產生革命性之衝擊, 為人們使用日常生活之所有通訊服務帶來莫大之便利; 本課程將對此一最新技術及其發展趨勢做深入淺出之說明 • WLAN 之近年來成為無線通訊之熱門話題, 各企業甚至家庭開始積極建設WLAN, 此趨勢相信將持續幾年. 不過建設WLAN作何用途? Public WLAN 服務業者有無生存空間? WLAN 與 3G關係又是如何? 互補呢還是競爭?本課程最後一節將與您一起探討WLAN 與無線通訊系統之競合關係, 了解WLAN應用之發展趨勢

3. 課程大綱 • IP 電信技術的發展趨勢 • VoIP 的過去現在與未來 • VoIP 的技術發展---從H.323 到SIP • SIP 的應用與發展趨勢 • ENUM --- IP/PSTN整合之新趨勢 • 何謂ENUM? • ENUM 服務機制 • WLAN 應用服務的發展趨勢 • WLAN 之應用 • WLAN 與無線通訊系統之整合

4. 通訊產業幾個重要發展趨勢 • VoIP 產業復甦, 有蓬勃發展趨勢 • IP-Phone, Wi-Fi SIP Phone, Residential Gateway, IP-PBX, … • 日本 Yahoo BB, J-Phone, … • Wireless LAN 蓬勃發展, 但服務商業模式尚待開發 • 技術演進迅速, 公司企業, 家用 • Public WLAN, Cellular/WLAN 整合 ? • Wireless Data (Mobile Internet) 發展不如預期 • Mobile Data << Voice • SMS, MMS, … • 3G 通訊系統前景不明 • GPRS  3G ? • Cdma2000-1x vs. WCDMA ?

5. IP 電信技術的發展趨勢

6. Off-net call 7.5 Yen/ 3 min. Call Agent / Softswitch PSTN (NTT) Gateway ISP / ASP (Yahoo!BB) VoIP TA On-net call free DSLAM ASDL + VoIP as The First Mile S POTS phones + PSTN phone numbers VoIP TA (1) Calls from PSTN (2) PSTN specific services, e.g. emergency call Yahoo!BB VoIP Business Model Over 1M+ BB Phone users (Dec 5th, 2002 Press Release)

7. VoIP 的過去 • Born of VoIP Industry • VoIP Phone --- VocalTech 1995 • H.323 Standard for VoIP Industry • H.323 Was Proposed by ITU-T for Multimedia Communication over LAN in 1996 • Adopted Quickly and Widely by Industries • VoIP Gateway, Gatekeeper, Enterprise GW, IAD • Mainly Used in Enterprise, Toll-by-Pass Applications • Market Slow Down • Infrastructure Bottleneck --- Limitation of H.323 • New Technologies Developed • Users Hesitate to Buy H.323 Products • Vendors Invest Much in H.323

8. New Technologies Ready MGCP, MEGACO/H.248, SCTP/M3UA, SIP/SDP,… IETF-like Protocols New Products Developed Most Upgraded from H.323 Products Support New Technologies as well as H.323 VoIP Infrastructure Softswitch, Trunk GW, SS7 Signaling GW (SS7 over IP) Market Re-Booming IETF Technologies Dominate SIP Telephony Being Widely Accepted ENUM for IP/PSTN Convergence Being Developed Softswitch vs. SIP Softswitch: Mainly for Class 4/5 Switch Replacement SIP: Provide Telephony from IP’s Point of View VoIP 的現在

9. VoIP 的未來 • VoIP Infrastructure will Get Ready (How Soon ?) • ENUM for IP/PSTN Convergence • Product and Service Start to Boom • It’s Technologies Trend, Not Just for Cost Saving • Driving Force of NGN • Mandatory in Future All-IP Network • SIP Technologies Will Dominate • SIP Telephony/CO Signaling/Multimedia Service/3G • Key Technologies for IP/PSTN Convergence

10. IP-SCP • IP-SCE Complexity • Softswitch • Signaling Gateway • Media Server&Gateway(IAD, Trunk G/W) • Application Server • SIP-based Systems 3G All-IP Core Network System Applications Carrier-grade Internet Telecom. System VoIP Technologies for 3G Core Network • VoIP Gateway/Gatekeeper • IP-PBX • IP-Phone Internet Telecom Service Technologies Enterprise VoIP Systems Open Telephony Architecture • IP-based SCP • IP-SCE • Pretty Amazing New Services (PANS) Next Generation VoIP Protocols • 3GPP Open Service Arch. • ISC Softswitch • VoIP Call Modeling • API: Parlay, JAIN,... World • MGCP • MEGACO/H.248 • SIP/SDP • SS7 over IP (SCTP) Technology H.323 System Taiwan Year 97 98 99 00 01 02 03 04 05 06 Trends of VoIP Technologies

11. SIP Phone SIP Phone IP/PSTN Convergence ENUM Service System Application Server ENUM Servers Media Server DNS SIP SIP Softswitch SIP Telephony Operator 2 Operator 1 SIP Proxy Gateway PSTN H.323 IP Network

12. VoIP Services Applications • PSTN’s (Telecomm.’s) Point of View • Equip PSTN with VoIP Services • Original Voice Users • For Cost Saving, No Service Enhancement • Class 4/5 Switch Replacement with Softswitch • No Extra Revenue • ISP’s (IP’s) Point of View • Enhance Data Network with VoIP Capability • Additional Voice Service to Original Data Users • Even Voice/Data Integrated Services • SIP Telephony + Application/Media Servers

13. VoIP Technologies--- From H.323 to SIP

14. H.323 Terminal H.323 MCU Packet Based Network Packet Based Network H.323 Gateway H.323 Terminal H.323 Terminal H.323 Gatekeeper Guaranteed QoS LAN GSTN N-ISDN B-ISDN H.310 terminal operating in H.321 mode H.321 V.70 H.324 Speech Speech H.320 H.322 H.321 Terminal Terminal Terminal Terminal Terminal Terminal Terminal Terminal H.323 for VoIP • Has Been Adopted by Vendors for Years • Almost Current VoIP Products Support H.323 • Gateway • Gatekeeper • Terminal • MCU

15. Challenges of H.323 • H.323 Was Originally Proposed for LAN Environment • Limitation Being Applied for VoIP Infrastructure Applications • Architecture is Too Simple • Single Gateway to Interact with PSTN/GSM • Suitable for Peer to Peer Communication • Protocol is Too Complex • Complicated Call Set-Up Signaling • Hard to Extend • Signaling Is Not Enough • Only DTMF is Specified • No SS7 Signaling Capability • Advanced Services are Limited • Interoperability is Still an Issue

16. H.323 GW MGC: Call Control, Call Routing, Service Selection,… Media GW Controller (MGC) Signaling GW (SG) SG Interface with SS7 and Translate to SS7 over IP MGCP MG Dedicate for Media Translation and Transportation SS7 Signaling Media GW (MG) Traffic Gateway Decomposition

17. IPDC 0.12 IPDC 1.0 Level3,... MGCP 1.0 MEGACO/H.248 SGCP IETF/ITU-T Telcordia MDCP AT&T May ‘98 Aug ‘98 Sept ‘98 Dec ‘98 Aug ‘00 Media Gateway Control IPDC: IP Device Control Protocol SGCP: Simple Gateway Control Protocol MEGACO: Media GAteway Control (IETF Working Group) MGCP: Media Gateway Control Protocol MDCP: Media Device Control Protocol

18. MGC (CA) MGC (CA) SGW SGW MGCP/ MEGACO SS7 (ISUP) MGCP/ MEGACO SS7 (ISUP) CO RTP Streams Trunk GW Trunk GW Access GW/ IP-PBX/RGW MGCP IP phone IP phone MGCP/MEGACO/H.248 • Client-Server Architecture • Call Agent (Media Gateway Controller) • Call Control Server • Media Gateway • Dedicated for Media Transmission • Trunk/Access/Residential GWs • Signal Gateway • SS7 Interfacing

19. MGC MGC SG SG ISUP/MGCP/SIP Interworking

20. SIGTRAN Protocol Stack SS7 Protocol Stack OSI Layers MAP INAP INAP MAP Application Presentation Session TCAP TCAP ISUP ISUP SCCP SCCP Transport SCN Signaling Adaptation (M3UA, IUA, …) Network MTP Data Link Common Signaling Transport (SCTP) Physical IP SS7 over IP

21. IN/SCP PSTN Local Switch PSTN Local Switch STP SS7 Network SG SG IP Network SIGTRAN MGC MGC SIP-T 9000 Megaco Trunk Gateway Trunk Gateway RTP Streams Personalized VoIP Service System IP-PBX IP Phone IP Phone Billing/Internet Messaging Authentication/Directory/... PC Phone Softswitch System Architecture

22. Services, Applications & Features (Management, Provisioning and Back Office) PROPRIETARY Services & Applications Call Control & Switching Softswitch Call Control Open Protocols APIs Open Protocols APIs TransportHardware Transport Hardware What is A Softswitch ? Soft-Switched Circuit-Switched • Solutions are open standards-based products, and can come from multiple vendors. • Customers are free to choose best-in-class products to build their network. Open standards enable innovation and in the long run can reduce costs • Software, hardware and applications solutions typically come packaged in a single (proprietary) box. • Non-standard solutions are expensive to implement and leave little room for innovation.

23. GSM BSS HLR Gateway MSC/VLR Visited MSC/VLR PSTN PCU GPRS Gb Gn External Data Network Gi SGSN GGSN Wireless System --- From GSM to GPRS Always-On Wireless Packet Data Services

24. IP Multimedia in 3G All-IP System Legacy Mobile Signaling Network Applications & Services Alternative Access Network SCP R-SGW CSCF Ms Multimedia IP Networks Mw Mh CAP CSCF HSS Mm Cx Mg Mr Gr Gi Gc MRF EIR Gi MGCF Gf T-SGW Iu Gi BSS/ GERAN GGSN TE MT SGSN Gn R Um Mc Gb A Gi Iu Uu PSTN Legacy/External MGW TE MT UTRAN MGW Iu Nb R Mc Mc SIP Iu Nc GMSC Server MSC Server CAP C D CAP Applications & Services Signaling Interface HSS Signaling and Data Interface

25. SIP Request SIP Response Redirect Server RTP Media Stream Proxy Server Location Server Proxy Server Proxy Server User Agent Client(Caller) User Agent Server(Callee) SIP Telephony System

26. SIP Attributes • User Agents • End Systems Acting on Behalf of a User • Clients(UAC) to initiate a SIP request and servers(UAS) to receive the request and return responses • Servers • Proxy Server Relay Call Signaling • Redirect Server • Registrar Server: User Locations Tracking (Like HLR in GSM) • Naming & Addressing • SIP URL, e-mail like: user@host • Examples • sip:patrik@example.com • sip:Jackie@176.7.6.1

27. SIP Operation in Proxy Mode

28. SIP Operation in Redirect Mode

29. Protocol Format INVITE sip:pgn@example.se SIP/2.0 Via: SIP/2.0/UDP science.fiction.com From: Fingal <sip:ffl@fiction.com> To: Patric <sip:pgn@example.se> Call-ID: 1234567890@science.fiction.com Cseq: 1 INVITE Subject: lunch at La Empenada? Content-Type: applcation/sdp Content-Length: … SIP Header v=0 o=ffl 53655765 2353687637 IN IP4 123.4.5.6 s=Chorizo c=IN IP4 science.fiction.com m=audio 5004 RTP/AVP 0 3 5 a=rtpmap:0 PCMU/8000 Payload: MIME type (SDP)

30. Applications R S V P SDP/SDPng RTP/ RTCP HTTP SMTP SIP RTSP SAP TCP/SCTP TCP/UDP UDP IPv4/IPv6 with Mobility, DiffServ and Multicast SIP in IP Multimedia Architecture • Common Addressing, user@domain • (URI & URL) • Text-based and Encoding Format • Same Request-Response Model, • and Response Codes • MIME for Flexible Payload • DNS for Address Mapping Easy to Do Integrated Services by Combining more than One Protocols

31. SIP-Based Services • Internet Telephony • Well-Recognized IP Telephony System • Instant Messaging • Exchange of Content between Participants in Real-time • Presence • Subscription to and notification of changes in the comm. State of a users • Click-to-Call from an Electronic Document • ‘SIP as tag’ Contained in Document • Transport of MIME in SIP Signaling • Reference to Web/WAP Pages, mail, pictures • Interactive Games with SIP Signaling

32. Rich Call with SIP • Before the Call • Dynamic Phone Book • Click-to-Talk • During Call Establishment • Sending a Rich Clip, which Can be • A Picture with Call Subject, or • Special Ringing Tone • During the Call • Document Sharing • Information Push • During Session Tear-Down • (same as Call Establishment) • After the Call • Be Able to Send Instant Message

33. SIP for Instant Messaging • Integration of Messaging, Presence, and Session-oriented Communication • RFC 3428 --- SIP Extension for Instant Messaging • SIP MESSAGE Method --- Pager-mode IM • For small IM Exchange • Carry IM on Signaling Channel --- Big Overhead • Pager-mode vs. Session mode Instant Messaging • Pager-mode --- Each message is independent of any other message • Session-mode --- IMs are associated with a media session established by SIP INVITE method • Message Session Relay Protocol (MSRP) --- IETF Draft • Support Session-mode IM between Endpoints

34. SIP for IP Convergence • In 3G and NGN, the Focus Shift towards to All-IP Paradigm • IP Transport, IP Mobility and VoIP (e.g. SIP) Protocol • SIP Has Been Adopted by 3GPP for Session Signaling • SIP Telephony System Has Been Well Accepted • With SIP, Service Creation is Fast and Easy • IP Protocols vs. Telecom Protocol, like IN • Be able to Integrate Most IP-based Services • Service Mobility and Personal Reachability • By SIP and IPv6 Mobility

35. CCL VoIP Solutions

36. CCL VoIP Solutions • Protocols • H.323, SGCP, MGCP/H.248, SIP, RTP/RTCP, SCTP and M3UA, … • Related VoIP Products Technologies and Systems • CTI / IP-PBX • Softswitch / Media Gateway Controller • SS7 over IP Signaling Gateway • Trunk Gateway • Media Server, Application Server • SIP-based Presence/Messaging Server • SIP-based IP Telephony • SIP User Agent • SIP Proxy Server • SIP-based Application Server

37. IP-PBX • IP Trunking • Rich PBX features • Internet Call Center LAN/WLAN ACD: Automatic Call Distribution

38. CCL IP-PBX --- VONTEL (cont.) • Enterprise grade IP-PBX • Rich Call Features • Up to 5,000 devices • BHCA: 60,000 (Busy Hour Call Attempt) • Multimedia IP Call Center Solution Platform • Advanced Skill-based Routing ACD System • Rich 3-party Call Control API • Support Voice / E-mail / Fax / and other media • Provide both proprietary & TAPI 3.0 compatible APIs • Converged Switch System • PSTN Interworking (Megaco Interworking) • SIP Interworking • H.323 Interworking • Inter-Switch Call Setup

39. CCL IP-PBX --- VONTEL (cont.) • Media Server • Voice Play / Record • Fax Send / Receive • DTMF Generation / Detection • Advanced Multi-party Conference(up to 8 parties / wiretap / whisper) • Softphone • Full-Feature, Intelligent Softphone • PSTN Gateway • Support Analog FXO Interface • Support T1/E1 ISDN Primary Rate Interface • Web-based OA&M

40. Application/Media Server (IP Unity) SIP Proxy Server Softswitch Web/Audio/Video Conferencing SS7 PSTN Off-net Call IP Network E1 Trunk Trunk Gateway Unified Messaging Instant Messaging Presence On-net Call User Agent (Softphone) SIP Phone (Cisco, PingTel) SIP Telephony System

41. CCL SIP Protocol and Achievements • 7th SIPIT, March2001 March • March 2001, sponsored by ETSI at Cannes, France • CCL SIP Protocol Stack and User Agent • 10th SIPIT, 2002 April • April 2002 , sponsored by ETSI at Cannes, France • CCL SIP Proxy Server • 13th SIPIT, 2003 August • August 20023 , sponsored by ETSI in Canada • CCL SIMPLE Server • 15th SIPIT, 2004 August • ITRI will Host 15th SIPIT Sponsored by ETSI in Taiwan in August 2004

42. CCL SIP User Agent • CCL SIP Protocol Stack • Provide C language API • Support RFC 2543 headers • Support RFC 2327 SDP Parsing • Support TCP and UDP • Support multi-platform (Windows, Solaris, Linux) • CCL User Agent • RFC 2543 and RFC 2543bis compliant • Rich Call Feature • Call Forwarding, Transfer, Hold, Consultation, … • Support RFC 2916 ENUM (E.164 number and DNS) • Support Windows platform

43. CCL SIP Proxy Server • CCL SIP Proxy Server • RFC 2543 and RFC 2543bis compliant • Support Register • Support Redirect • Support location service • Support stateful and stateless mode operation • Support parallel and sequential search • Support RFC 2916 ENUM (E.164 number and DNS) • Support multi-platform (Windows, Solaris, Linux)

44. SIP Telephony Features • Basic telephony features • Basic on-net call • Basic off-net call (PSTN interworking) • Enhanced features • Calling number delivery (CND) • Call forwarding busy (CFB) • Call forwarding don’t answer (CFDA) • Call waiting (CW) • Selective call rejection (SCR) • Auto recall • Application/Media server functions • Announcement • Voice mail • Conferencing

45. IP Network • Media Server • Announcement • Voice Mail • Conferencing Element Management Server • Application Server • Web Conferencing • Unified Messaging IP Router SS7 PSTN 10/100 Ethernet Switch Signaling Gateway Billing Gateway Trunk Gateway Integrated Access Devices Database Server Media Gateway Controller CCL Softswitch Solution

46. ENUM --- IP/PSTN整合之新趨勢

47. Why ENUM ? • How do network elements (gateways, SIP servers etc) find services on the Internet if you only have a telephone (E.164) number? • How can subscribers define their preferences for incoming communications? • Many Number & Addressing Mechanism Used • With ENUM  Only One Number

48. ENUM 101 • What is ENUM (RFC 2916)? • Make Telephone Number Become a Domain Name in Internet • Any Telephone Number can Register As Domain Name in IP World • Importance of ENUM • New Addressing Mechanism in IP World • Mechanism for IP/PSTN Convergence • Global ENUM Activities • IETF, ITU-T, US ENUM Form, Europe, … • ENUM Trials: More Than 10 Countries • Different Kinds of ENUM • Public (Personalized) ENUM: Future Vision • Carrier ENUM: Enhance MMS/SMS Services • Enterprise ENUM: Enhance Enterprise Telecomm Service

49. What is ENUM ? Problem: How to Address (or Locate) a User in IP World from IP/PSTN Network ? ENUM is a Mechanism to Translate an E.164 Number into A List of URI (Service@Host) So that An IP User Can be Accessed by E.164 Number. ENUM Mechanism E.164 Number URI(Service@Host) Domain Name System (DNS)

50. DNS-Server Gateway Sip server Basic Operation of ENUM Query 1.3.1.9.5.8.6.8.6.4.e164.arpa.? Response sip:paf@cisco.com “Call setup” Dial +4686859131 Sip sip:paf@cisco.com Source: IETF