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Voice over IP 與 IP Telephony 簡介

Voice over IP 與 IP Telephony 簡介. 資策會 網路及通訊實驗室 Conference over IP Team 楊政遠 博士 yzy@netrd.iii.org.tw 2003/07/26. Review - PSTN. PSTN (Public Switch Telephone Network) Signaling : System Signal No: 7 ( SS7 ) Carrier : T1 主幹 and successors …. Signaling plane. STP. 局端 ( CO ).

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Voice over IP 與 IP Telephony 簡介

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  1. Voice over IP 與 IP Telephony 簡介 資策會 網路及通訊實驗室 Conference over IP Team 楊政遠 博士 yzy@netrd.iii.org.tw 2003/07/26

  2. Review - PSTN • PSTN(Public Switch Telephone Network) • Signaling: System Signal No: 7 (SS7) • Carrier: T1 主幹 and successors …... Signaling plane STP 局端 (CO) Bearer plane 客戶端(CPE) Local loop DTMF

  3. 1. ? 2. Conference setup 4. Conference terminate 3. Digital voice packets Review - Voice Conference • Basic issues of voice conference setup phonebook server AD/DA compress/decompress AD/DA compress/decompress

  4. Review - basic issues • Telephony Issues (PSTN v.s. VoIP) • Signaling • Addressing / Control • PSTN - SS7 (ITU E.164) • VoIP - H.323、SIP、MGCP、Megaco/H.248 • Capability exchange • PSTN - Analog voice / -law、A-law PCM • VoIP - Digital voice / G.711、G.723.1 、G.729

  5. Review - basic issues • Telephony Issues (PSTN v.s. VoIP) • Bearer • Transport • PSTN - TDM (Time Division Modulation) Trunk • VoIP - RTP over UDP/IP • Delay and Jitter • PSTN - circuitswitching / propagation delay • VoIP - packet switching / unbounded delay and jitter • Internetworking between the existent PSTN, GSM/GPRS and future 3G all IP network.

  6. Review - short conclusion • Signaling • Addressing: find call party • Call control: control the call progress • Capabilities exchange: negotiate the media types of this call • Media transport (bearer) • media processing • media transmission

  7. Media Transport Media ProcessingMedia Transmission

  8. Process of digital voice transmission Low-pass filter Sampling & A/D convert Silent detection RTP packet encapsulation Compression Internet Timing reconstruct RTP packet decapsulation D/A convert Decompression

  9. VoIP Endpoint Functionality PCM DSP coding Phone interface frames Buffering and packetization Jitter buffer Digital Signal Processing Packet handling AD/DA converter TCP/IP protocol stack Network interface copper wire IP

  10. Digitalization Speech • Digitalization Speech • Low Pass Filter (LPF) 300 Hz ~ 3000 Hz • Sampling and Quantization

  11. Digitalization Speech • PCM (Pulse Code Modulation) • digital quantization introduces distortions

  12. Digitalization Speech • main speech coding techniques • waveform codec, source codec and hybrid codec

  13. Color Transform RTP packet encapsulation Huffman encoder Discrete Cosine Transform Quantizer R,G,B bitmap Y,Cb,Cr matrix Huffman table frequency matrix Quantization table

  14. Media TransportMedia Processing Media Transmission

  15. Voice Quality of Service • Interactive Voice QoS factors • Packet lost • Delay • Jitter

  16. Voice QoS - Packet Lost Intranet Internet

  17. GPRS Backbone IP Network Voice QoS - Delay • Minimize one-way delay, keep it below 150ms ITU G.114 states one-way delay <= 150 msec ~200 msec is acceptable Fixed delay 1. Framing: 20~30 ms 2. Processing: 15 ms 3. Transmission: 10ms 4. Decompress/buffer: 25 ms Framing (algorithm): 20 ~ 30 ms Compress (H/W DSP): 5 ms Processing (packetize): 10 ms Variable delay 1. Buffer: 5~20 ms 2. Network: 20 ~ ? ms IP based network variable delay 20~300 or more ms Receiving buffer: 20 ms Decompress delay: 5 ms

  18. Voice QoS - Delay • Codec algorithm delay ( Ex. G.729 ) • serialize the frame ( 10 ms) • look ahead (5 ms) total algorithm delay = 15 ms next sample Sampling & A/D converter 8000 Hz Frame

  19. Voice QoS - Delay Internet Intranet

  20. Voice QoS - Jitter

  21. Jitter (Delay Variation) Internet Intranet

  22. Packet Handling Latency • Jitter • variability in the arrival rate of data is called jitter

  23. Voice QoS - Jitter • Jitter buffer

  24. Definitions • Voice over IP (VoIP) • Voice over Internet Protocol • voice packet over well controlledIP network ! • does not imply Voice over Internet • IP Telephony • Telephony system based on Internet Protocol • Inter-operabilities • standards • compatibility

  25. Voice packets transmission • TCP(reliable) or UDP(unreliable) ? • The characteristics of interactive voice/video • on-the fly (realtime) • retransmission is none-sense • human physiology • tolerate few information lost independently • isochronal • timing information snapshot and re-construct • media frame encapsulated in RTP/UDP/IP IP header (20 bytes) UDP header (8 bytes) RTP header (12 bytes) media payload

  26. RTP: A Transport Protocol for Real-Time Applications (RFC 1889) http://www.ietf.org/html.charters/avt-charter.html

  27. RTP (RFC1889) • The simplest RTP fixed header IP header UDP header RTP header RTP payload

  28. Fields of RTP Header • V (version): • RFC 1889 RTP version 2, V=2 • P (padding): • padding bytes ? • X (extension): • RTP header extension ? • CC (count of contributor): • number of media contributors (for mixer) • M (marker): • media specified • audio: the begin of talk spurt • video: begin of end of video frame • PT (payload type): • Defined by RFC 1990

  29. Fields of RTP Header • Sequence number: • increment by one • initial value is random • Timestamp: • reflect the sampling instant of the 1st data bytes • format depends on application • initial value is random, increments monotonically • Sync SRC: • synchronization source ID • random choice • RTP session global uniquely

  30. RTP Header profile (RFC1900)

  31. SignalingAddressing Call control Capabilities exchange

  32. Review • The milestone of Voice over IP • the 1st experiment of voice packet over IP • 1974 Network Voice Protocol (RFC741) • the 1st commercial Internet telephony AP, Windows 3.1 • Vocaltec, 1995 • the 1st version of H.323 • ITU, 1996 • the 1st widely deployed H.323 AP • Microsoft NetMeeting, May, 1996 • the 1st commerical Internet Telephony Service • Delta Three, 1996

  33. VoIP signaling protocol standard • ITU-T H.323 • http://www.itu.int/rec/recommendation.asp?type=folders&lang=e&parent=T-REC-H.323 • IETF MGCP • RFC2705 • IETF SIP • RFC3261 • http://www.ietf.org/html.charters/sip-charter.html • IETF/ITU-T Megaco/H.248 • RFC3015

  34. Session Initiation Protocol • SIP Architecture • RFC3261 SIP User Agent SIP User Agent SIP Server SIP User Agent Registrar Proxy Server Redirect Server

  35. INVITE SIP:abc@xyz.com.tw SIP/2.0 ……. 180, Ringing 200, OK ACK RTP (voice) BYE ACK VoIP protocol standard - SIP SIP BASIC Call flow Caller Callee Pickup & dial ringing ringback pick up on-hook

  36. Request Methods

  37. Response Status Line • SIP-Version SP Status-Code SP Reason-Phrase CRLF • Status-Code = • SIP/2.0 SP 180 SP Ringing CRLF

  38. SIP Request Example

  39. REGISTER sip:iptel.org SIP/2.0 From:sip:jiri@iptel.org To:sip:jiri@iptel.org Contact:<sip:195.37.78.173> Expires:3600 jiri@195.37.78.173 SIP/2.0 200 OK SIP Registration Location Server SIP Registrar (domain: iptel.org)

  40. jiri ? INVITE sip:jiri@195.37.78.173 From:sip:Caller@sip.com To:sip:jiri@iptel.org Call-ID:345678@sip.com INVITE sip:jiri@iptel.org From:sip:Caller@sip.com To:sip:jiri@iptel.org Call-ID:345678@sip.com jiri@195.37.78.173 SIP/2.0 200 OK SIP/2.0 200 OK ACK sip :jiri@195.37.78.173 SIP Operation in Proxy Mode Location Server SIP Proxy Server Caller@sip.com jiri@195.37.78.173

  41. Callee ? Callee@home.com INVITE sip:Callee@example.com 302 Moved Temporarily Contact: Callee@home.com ACK sip:Callee@example.com INVITE sip:Callee@home.com SIP/2.0 200 OK ACK sip:Callee@home.com SIP Operation in Redirect Mode Location Server Caller@sip.com Callee@home.com SIP Redirect Server

  42. SIP, H.323 and MGCP Call Control and Signaling Signaling and Gateway Control Media Audio/ Video H.323 H.225 H.245 Q.931 RAS SIP MGCP RTP RTCP RTSP TCP UDP IP H.323 Version 1 and 2 supports H.245 over TCP, Q.931 over TCP and RAS over UDP. H.323 Version 3 and 4 supports H.245 over UDP/TCP and Q.931 over UDP/TCP and RAS over UDP. SIP supports TCP and UDP.

  43. Protocol wars - Viewpoint from CISCO Projected Port (DS0) Protocol Transition Rates 100% 80% 60% Mixed H.323 & SIP % Port Unit Sales 40% MGCP / H.248 - DS0s SIP - DS0s 20% H.323 - DS0s Q1CY99 Q1CY00 Q1CY01 Q1CY02 Q1CY03 Q1CY04 Calendar Quarters

  44. Next Generation Converged Network andIP Telephony system

  45. 10 Total 9 Data 8 Telephony 7 6 5 Relative traffic 4 3 2 1 0 2003 Year 1997 1998 1999 2000 2001 2002 Siemens

  46. Next Generation Converged Network • Telecommunication deregulation • Investment reward : Data network > voice network • Cost - single network architecture • Cost - open standards / short time-to-market • Open VoIP and supplemental standards • H.323、MGCP 、 Megaco/H.248 、 SIP • Bandwidth is no more a critical issue • DWDM， xDSL / cable ， Fast/Giga Ethernet • Quality of Service guarantee

  47. Next Generation Converged Network

  48. Next Generation Converged Network • Residential Gateway / Integrated Access Device

  49. IP Telephony System must support IP Telephony System Feature and Application Creation Operation System Support Call Control and Switching

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