1. Voice over IP

2. 2 Agenda  Advantages of packet switching for voice communications  VoIP applications  VoIP technology overview  VoIP standards  Quality-of-Service in VoIP networks  Addressability in VoIP networks  VoIP regulatory considerations

3. 3 What is VoIP?  Technical answer: “the ability to make phone calls over IP-based data network”  Commercial answer: ”the Multi-Billion Revenue Opportunity for the 21st Century”  VoIP > IP Telephony  typically “IP Telephony” indicates using IP terminals  most VoIP is between normal telephones  VoIP < “Voice over Packet”  includes Voice over Frame Relay, ATM, xDSL, Ethernet, WiFi

4. 4 Circuit switching served voice well for 100 years!  Transmission circuits and switch path assigned during call setup for the duration of the call  Call blocks if not enough network resources available  Essentially one class of service: 3.5 kHz, 64 kb/s  Poorly matched for bursty data transmission User - AUser - B Loop Trunk Group Central Office - A Central Office - B Signal System 7 Data link Signal Transfer Point Transit Office Class 5 Switching SystemConnection Through Switching Fabric Class 4 Switching System

5. 5 Packet Switching Well-matched for data transmission  Great fit for bursty data transmission!  Packets sent at full rate of transmission facility  Supports variable information transfer rates  Resources not consumed when nothing to send  Potential to eliminate call setup phase  But …  Transmission capacity used for header  Buffering introduces varying delays, like speaking to man on moon Header Packet Payload Input Buffer Output Buffer Hdr. Trans Routing Fabric

6. 6 VoIP Network Architecture  Media gateways provide voice packetization  Gatekeepers provides call control logic and permissions  Gateway provides interworking with ISDN, SS7 and signaling of PSTN (POTS) IP network Media Gateway Gatekeeper Media Gateway PSTN network Gateway

7. 7 Advantages of VoIP  Lack of access charges, flat rate or volume based IP  Cheap setup costs competition with POTS  Cheaper switching systems  Per Gb/s, IP routers cheaper than TDM Class 5 switching systems  Ability to operate one network for voice and data  Cost savings through use of low-bit-rate voice  Ability to offer more complex services  E.g., Multimedia, conferencing calls  Intelligent terminals (e.g., PC)  Better (graphical) user interface  Clean slate design:  Separation of feature intelligence from switching fabric supplier  Self-provisioning networks

8. 8 PSTN Vs VoIP Network Costs  Network costs (transmission and switching costs) contribute only 10-15 % of overall cost of a voice call terminated by an ILEC or a PTT, and 20-30% of overall costs for calls not terminated by a ILEC or a PTT  Of the network costs, switching costs range between 50 % of network costs for domestic calls to 15 % of network costs for international calls, transmission costs contributing the rest  Negligible savings in transmission costs through the use of VoIP: lower bandwidth for VoIP offset by need for over- provisioning bandwidth to ensure quality  TDM Switch costs in traditional PSTN replaced by cost of Router plus cost of Gateway and new billing systems No network cost savings, and very likely a cost penalty, in the initial years, in going from PSTN voice to VoIP for public networks

9. 9 PSTN versus VoIP Today’s PSTNVoIP Underlying Technology TDM circuit switchingPacket switching QoS guaranteesYesNo Network resource reserved at call setup YesNo Network elements Class 4, Class 5 switching systems Gateways, gateway controllers, routers Call processing intelligence Mostly integrated in switching system In separate gateway controllers Bandwidth per call64 kb/sVariable 5.3 – 32 kb/s SignalingDTMF, SS7SIP, H.323, MGCP Transport TDM in access, edge, core ATM, FR, native IP in access; ATM native IP in core, WiFi How reliability achieved Redundancy within each network element Redundant routes through network

10. 10 VoIP versus Voice-over-the-Internet  Voice-over-the-Internet  No bandwidth guarantees  No prioritization of traffic within network  All traffic receives “best effort” service  Each Internet user is at the mercy of all other users  Voice quality ranges from acceptable to atrocious However  Internet technology continues to evolve (e.g., IPv6)  Development of Next Generation Internet

11. 11 What does “Carrier Grade” really mean?  “Five 9’s” reliability (down time of 5 minutes a year)  Full redundancy of electronics, power supplies, fans, etc.  No down time for upgrades or maintenance  Accounting and billing capabilities  Interoperability with legacy telecommunications equipment  Feature parity with equipment it replaces  Service quality measurements  Support for CALEA, unbundling, and other governmental mandates  NEBS compliance for operation in central offices  Both safety and performance requirements  Scalability to millions of subscribers  Integration into the myriad of Operations Support Systems

12. 12 VoIP market Voice over Internet Protocol (VoIP) gateway sales will increase 280 percent during the next five years, reaching $3.8 billion in 2003, according to research by Cahners In-Stat Group. IP TELEPHONY OVER LAN MARKET FORECASTED TO GROW 138% AVERAGE ANNUALLY OVER NEXT 5 YEARS September 22, 1999 - IP Telephony [IP PABXes], according to a study from The Phillips Group-InfoTech, will spawn a $1.9 billion industry by the year 2004 with an average annual industry growth of 138 percent over the next 5 years. IDC Forecasts IP Telephony Market Will Soar to 2.7 Billion Minutes of Use and $480 Million in Revenues by Year end 1999 Business Use Will Accelerate in 2001 September 1, 1999 - The worldwide Internet protocol (IP) telephony will explode from 310 million minutes of use in 1998 to 2.7 billion by year end 1999. By 2004, IP telephony minutes will reach 135 billion. Revenues for this service will skyrocket from $480 million in 1999 to $19 billion by 2004. IP Telephony Services: Market Review and Forecast, 1998-2004. IDC Forecasts IP Telephony Market Will Soar to 2.7 Billion Minutes of Use and $480 Million in Revenues by Year end 1999 Business Use Will Accelerate in 2001 September 1, 1999 - The worldwide Internet protocol (IP) telephony will explode from 310 million minutes of use in 1998 to 2.7 billion by year end 1999. By 2004, IP telephony minutes will reach 135 billion. Revenues for this service will skyrocket from $480 million in 1999 to $19 billion by 2004. IP Telephony Services: Market Review and Forecast, 1998-2004.

13. 13 Growth in VoIP  Early growth from expense savings  Later growth from revenue generation from new services  Early deployment by enterprises and CLECs  Later deployment by incumbent carriers (source: Frost & Sullivan)

14. 14 Class 5DLCClass 5DLC VoIP Applications  Some trends can be discerned:  First wave: Bypassing the PSTN  Second wave:Replacing the PSTN  Third wave:Value-added services PSTN

15. 15 PSTN bypass – IP Telephony (PC to PC)  Microsoft NetMeeting or similar  through dial-up/adsl/cable connection to ISP  All VoIP processing in the PC  no special infrastructure required  Issues:  software compatibility  QoS / latency over public Internet  Strange dialing Internet Class 5DLCClass 5DLC RAS modem RADIUS server RADIUS server

16. 16 PSTN bypass – IP Telephony (PC to PHONE)  From Multimedia PC to any PHONE  First applications 1993  Required:  VoIP gateway on the phone side  gateway manager  billing system (unless free)  Issues:  software compatibility  QoS / latency over public Internet Internet Class 5DLCClass 5DLC RAS RADIUS server VoIP Gateway Gate Keeper modem

17. 17 PSTN bypass – IP Telephony (phone to phone)  From any PHONE to any PHONE  First VoIP application – 1995  Caused by high international tariffs  Required:  VoIP gateway on both sides  gateway manager  billing system (unless free)  Issues:  QoS / latency over public Internet  sometimes it takes 24 digits to reach a subscriber… Class 5DLCClass 5DLC VoIP Gateway Gate Keeper VoIP Gateway IP network

18. 18 PSTN bypass – IP Telephony (phone to pc)  From any PHONE to any PC  First VoIP application – 2004  Try to replace PSTN  Required:  VoIP gateway on PSTN side  MSN numbers  gateway manager  billing system (unless free)  Issues:  QoS / latency over public Internet Class 5DLC Gate Keeper VoIP Gateway IP network Class 5DLC RAS modem RADIUS server

19. 19 PSTN replacement – Softswitch  Replace complete Class 4 / Class 5 switch  very ambitious undertaking!  different introduction strategies  Required  Softswitch- contains Call Control & Mgmt software  Trunking Gateway – interfaces to “legacy” PSTN  Access Gateway – interfaces to DLCs  Issues:  immaturity of standards (MGCP vs Megaco debate) DLCClass 5DLC Access Gateway Trunking Gateway Soft switch IP network

20. 20 PSTN replacement – Integrated access network  Integrating Access Gateway into DLC  Required:  “Next Gen” DLC, with integrated IP gateway  Issues:  immaturity of standards NexGen DLC NexGen DLC Soft switch IP network

21. 21 PSTN bypass – IP PABX  Two steps:  A. PABX with integrated IP gateway  B. Fully integrated enterprise LAN  Required:  IP PABX  IP phones (step 2)  Issues:  dial plan configuration not easy!  how to quarantee QoS on LAN? (step 2) IP network IP-PABX IP-phone PSTN AB

22. 22 Class 5 PSTN Gate Keeper VoIP Gateway Integrated Access Device PSTN replacement – Integrated Access Devices  Target: single voice/data access network  for example wireless access network  Home networks  companies  Required:  Integrated Access Device (IAD)  gateway to PSTN somewhere  Issues:  immaturity of standards Integrated Access Device IP network Soft switch

23. 23 Value Added Services  Converged services  Internet Call Waiting  Click to Call  Unified messaging  …  Video telephony (3rd time right?)

24. Standards for VoIP

25. 25 The H.323 Protocol Stack H.225 RAS channel H.225 RAS channel Q.931 call setup Q.931 call setup H.245 control H.245 control Audio And Video Control RTCP Audio And Video Control RTCP T.120 Audio codec G.711 G.723 G.729 Audio codec G.711 G.723 G.729 Video Codec H.261 H.263 Video Codec H.261 H.263 RTP Transport Layer (TCP or UTP) IP System control user interfaceMicCamera Data applications

26. 26 H.225 RAS Control  Gatekeeper  Optional network entity  Offers bandwidth control services  Offers address translation to enable use of aliases  H.225  Operates between a Gatekeeper and the endpoints it controls  Provides functions of discovery, registration, admission, bandwidth change, disengage Gatekeeper Endpoint Gateway Multiport Control Unit H.225

27. 27 Call Signaling in H.232  Q.931  Establishes and tears down calls between endpoints  (Q.931 is the signaling protocol for the ISDN user-network interface)  H.245  Negotiates and establishes media streams between call participants  Takes care of multiplexing multiple media streams for functions such as lip synchronization between audio and video Q.931 H.245

28. 28 Session Initiation Protocol (SIP)  User to user protocol  Developed by IETF (RFC 2543)  Establishes and maintains session level information  Creating and tearing down of sessions, session parameters, and media type  Supports personal mobility  Heavily influenced by http protocol  A light weight protocol compared to H.323  Fewer messages required on a typical call  Allows for faster call setup  Flexible in enabling other information to be included messages  Allows user devices to exchange specialized information to enable new services  E.g., indicate when a busy terminal will become free  Example SIP addressing; sip:9729965000@gateway

29. 29 Internet call processing  Decentralized (independent, self-reliant, user to user):  ITU H.323  IETF Session Initiation Protocol (SIP)  Centralized (intelligence in Softswitch):  IETF MEGACO  ITU H.248

30. 30 Softswitch Architecture  Softswitch separates function of Gateway from the media gateway Access Gateway Trunk Gateway Softswitch IP Network PSTN Network MGCP Or Megaco SIP-T To other Softswitches

31. 31 ATM QoS Parameters  Peak-to-peak cell delay variation  Maximum cell transfer delay  Cell loss ratio  Cell error ratio  Severely errored cell block ratio  Cell misinsertion rate Negotiated at start of call Controlled via Network design

32. 32 Real-Time Multimedia over ATM (RMOA)  Developed by ATM Forum  More efficient and scalable than H.323 VoIP over ATM  New type of gateway: the H.323 to H.323 gateway  Placed at the edges of an ATM network  Intercepts H.323 signaling messages to set up virtual circuits in the ATM network  Efficient: IP and UDP headers not carried on the ATM network  Takes advantage of QoS capabilities of the ATM network ATM network PSTN Switch PSTN Switch IP Network VoIP Gateway VoIP Gateway H.323 Gateway H.323 Gateway

33. 33 Resource Reservation Protocol (RSVP)  Specified in RFC 2215  Reserves resources along path from received back to sender  Implements various services  Guaranteed service – no packet loss and minimal delay  Controlled load service – service like a lightly loaded network  Number of parameters associated with each service  Comprehensive, close to circuit emulation, but at significant cost Application RSVP Process Policy Control Admission Control Packet Scheduler Packet Classifier Control Routing Process RSVP Process Policy Control Admission Control Packet Scheduler Packet Classifier Control HostRouter

34. 34 Adding QoS to IP Networks: Diffserv  Relatively simple means for prioritization traffic (RFC 2475)  Makes use of the IPv4 Type of Service (TOS) field  Defines two types of packet forwarding:  Expedited Forwarding – assigns a minimum departure rates greater than the per-agreed maximum arrival rate  Assured Forwarding – packets are forwarded with high probability if arrive no faster that per-agreed maximum  Keeps core relatively simple  Pushes processing to the edge Meter ClassifierMarker Shaper / Dropper

35. VoIP access via DSL and Cable Modems

36. Cable Telephony  Where to put the RJ-11 telephone jack?  On cable modem  On set-top box  On separate telephony modem  On interface on side of house  Local powering or network powering options Head end Head end Video Content Fiber Node Internet Service Gateway PSTN

37. What is DOCSIS? (Data Over Cable System Interface Specifications)  Started 12/95 by MCNS consortium (Multimedia Cable Network System)  Goal: Interoperable cable modems and Cable Modem Termination Systems (CMTS)  Steamed rolled slower (ATM-based) IEEE 802.14 standardization process  Gaining momentum in Europe as EuroDOCSIS (8 MHz channelization)  Testing and certification by Cable Labs

38. Who are the DOCSIS Cable Modem Suppliers?  3Com  Ambit  Arris Interactive  Askey Computer Corp.  Best Data  Castlenet  Cisco Systems  Com21  Dassault  DeltaKable  DX Antenna  ELSA  E-Tech  Future Networks  GadLine  Toshiba  Turbocom  General Instrument  GVC  Joohong  Motorola  Net N Sys  Nortel  Philips  Powercom  Samsung  Sohoware  Sony  Tarayon  Thomson  Zoom  ZyXel

39. tCable projected to capture 15 % telephony market share by 2005 tShift from proprietary TDM solutions towards VoIP DOCSIS tResidential VoIP happening first in the Cable Access Market North America Cable Telephony Market Size

40. 40 Class 5 Switch ATM Switch Voice Gate Way Integrated Access Device DSLAM LAN 1 VC for Voice 1 VC for Data ADSL DS3 / OC-3 GR303 HOME/BUSINESS CO / CEV CO 4-16 Voice over DSL  Integrated Access Device (IAD) provides LAN interface and provides multiple telephone interfaces  IAD could be integrated into NID at side of the home  Voice Gateway provides same switch interface as though lines were concentrated on a Digital Loop Carrier system  GR303 allows for number portability, billing and additional voice features PSTN Data Network

41. 41 IP ATM DMT Analog Spectrum Voice over IP Voice over ATM Voice over TDM Voice in separate spectrum (e.g., ADSL over DAML) Voice over ADSL Alternatives  Choice of Voice over ATM in initial implementations – AAL-2 – Low-delay, clear 64 kb/s PCM and 32 kb/s ADPCM – QoS support within ATM – Full PSTN quality – V.90 modem support  Support for Voice over IP gaining momentum  Maturing of QoS capabilities  Potential of IAD becoming a SIP terminal Layer 3 Layer 1 Layer 2 Alternatives for VoDSL

42. Quality issues for the transport of voice over packet-based networks

43. 43 The three essential stages of packet-based voice transport one-way Mouth-to-Ear (M2E) delay overall distortion (codec & packet loss) Encoding and packetization stage Packet transport stage Echo control performed close to destination (Concatenation of) Packet-based Packet-basedNetwork(s) Dejittering and decoding stage

44. 44 Packetizationdelay Totalqueuingdelay Dejitteringdelay Totalminimaldelay M2E delay Components of the M2E delay the source terminal or ingress GW  Packetization delay is chosen by the source terminal or ingress GW QoS provided by traversed network(s)  Minimal delay and queuing delay depend on QoS provided by traversed network(s)  Each network component has its specific contribution the destination terminal or egress GW  Dejittering delay is chosen by the destination terminal or egress GW

45. 45 Trade-off M2E delay vs. packet loss in destination or egress GW Packet loss Dejitteringdelay Dejittering delay Delay of first packet Minimal delay M2E delay Pdf(delay)  Static dejittering mechanism = delay first packet over dejittering delay and then read dejittering buffer periodically  Choose dejittering delay on save side: for the case when first packet is the fastest possible  Adaptive dejittering

46. 46 Contributions to distortion  Voice compression  encoding/decoding  voice activity detection  transcoding  Packet loss  in network  in dejittering buffer  Remarks  packet loss concealment techniques  trade-off packet loss vs. delay when choosing the dejittering delay

47. 47 Trade-offs Packetsize Network (transport) parameters  minimal delay  delay jitter  packet loss Codec Efficiency of transport Voice quality DejitteringdelayEchocontrolHeadercompression

48. 48 Speech Coding Techniques  Waveform coding – Tries to preserve the time-domain picture of the signal  Sampling – 2 X highest frequency preserved  Quantizing – the accuracy of each sample  Linear – simple digital / analog conversion  Logarithmic – more accuracy for weak signals  Adaptive – match measurement to size of signal  Sounds great at high bit rates but degrades quickly at lower bit rates  Vocoding – Tries to represent the characteristics of the human voice  Prametric Vocoders  Dozen coefficients to define vocal tract  Indication of voiced or unvoiced  Excitation energy  Pitch  Synthetic sounding at all bit rates but works OK at low bit rates  Vector Quanitization – Matches information signal with entries in a code book.  Uses lots of processing power but provides the best quality at lower bit rates

49. 49 Major Parameters of Standard Codecs OriginStandardType Codec Bit rate Voice Frame (ms) Look ahead (ms) Algor. delay (ms) le Intrinsic quality ITU-T G.711PCM64 0.1250 094.3 G.726 G.727 ADPCM 165044.3 242569.3 32787.3 400.1250 292.3 G.728LD-CELP 12.8 0.6250 2074.3 16787.3 G.729(a)CS-ACELP8105151084.3 G.723.1 ACELP5.3 307.537.5 1975.3 MP-MLQ6.31579.3 ETSI GSM-FRRPE-LTP13200 74.3 GSM-SRVSEPL5.6200 2371.3 GSM-ESRACEPL12.2200 589.3

50. 50 Influence of packet loss on distortion

51. 51 Transcoding matrix  Transcoding is the translation of one codec format into another (via the linearly quantized 8 kHz sampled voice format)

52. 52 M2E delay and packet loss bounds  If there is no packet loss, the M2E delay can exceed 150 ms  If the M2E delay is below 150 ms some packet loss can be tolerated Bounds under perfect echo control

53. 53 Quality of a telephone conversation (using the E-model of ITU-T Rec. G.107 and G.109) (Very) Bad Poor Medium High Best Perfect echo control

54. 54 Conclusions Quality of a telephone call  (Perfect) echo control is strongly recommended  Under perfect echo control the intrinsic quality remains constant if M2E delay < 150 ms  Choose codec to have an intrinsic quality that is good enough  e.g. G.711, G.729,...  Avoid transcoding from one low bit rate codec into another  Keep M2E delay and packet loss under control  bounds are codec-dependent  There is a trade-off between M2E delay and distortion

55. 55 Conclusions Setting the parameters  The quality with which the voice flows are transported influence the overall quality, but …  … the choice of the codec, packet size and dejittering delay is also primordial  In the choice of the codec there is a trade-off between efficiency and quality  In the choice of the packet size there is a trade-off between efficiency and quality  Tuning the dejittering mechanism correctly is important to attain high quality

56. Addressability in VoIP Networks

57. 57 Addressibility in VoIP  Question: How do you dial a VoIP user if all you have is their telephone number? alcatel.comge.comfcc.govibm.com  Users resistant to change services if they have to change phone numbers

58. 58 What is ENUM?  Telephone number mapping (RFC 2916, RFC 2915)  Allows a phone number to enable a caller to reach all kinds of devices (fax, IP telephone, email, etc.) by knowing a single contact number  Originally proposed by Patrik Falstrom of Cisco  Uses DNS structure to map an E.164 phone number into a series of Internet addresses:  SIP, H323, SMTP, VPIM, IPP, etc.  Enables Local Number Portability, 800 services

59. 59 DNS-B (0.5.8.9.1.9.1.e164.arpa) DNS-A (9.1.9.1.e164.arpa) How does ENUM work? proxy.com INVITE Answer = sip:niel@proxy.com “(919) 850-5500" Query 0.0.5.5.0.5.8.9.1.9.1.e164.arpa Authority = DNS-B Query 0.0.5.5.0.5.8.9.1.9.1.e164.arpa

60. Regulatory Considerations

61. 61 Context  The third ITU-T World Telecommunication Policy Forum (Geneva, March 7-9 2001) discussed issues related to “Internet Protocol (IP) Telephony”.  The WTPF discussed the impact of IP telephony on regulation and policies of ITU member states and ways for offering technical assistance to developing countries.  A report of the secretary-general and draft opinions for the forum are finalized and available on the ITU website (http://www.itu.int/wtpf).

62. 62 What is at stake ?  Beyond the technological hype surrounding IP telephony, the real issue is the structure of the 21st century world-wide telecom network and the nature - and mere existence ! - of the settlement system governing the interconnection between operators.  Many developing countries are fearing that widespread deployment of unregulated IP telephony traffic will dramatically lower the revenue stream drawn from the settlement system and, by way of consequence, the eventual insolvency of their local PTO(s).  The secretary-general’s report on IP telephony is quite objective and factual but the WTPF draft opinion recommendations reflect conflicting interests.

63. 63 The “Netheads” view  Driven by CISCO, VON coalition, global operators (Worldcom, AT&T).  Objective: convince reluctant (mainly developing) countries to allow free competition of IP telephony with their local PTO.  Mantra:  IP is “the new” technology for telecommunications;  IP is much more efficient (cost) than legacy TDM;  IP networks open the way for new services and help reduce the “digital divide”;  IP telephony should not fall under the telecom regulation regime (or this regime should evolve) because it uses a new technology.

64. 64 The EU view  Advocates the principle of technological neutrality.  EU has a strict definition of voice telephony in terms of the following four principles:  it is offered commercially as such;  it is provided to the public;  it is provided to and from PSTN termination points;  it involves speech transport and switching of voice in real-time with the same level of reliability and quality as existing PSTN networks.

65. 65 Other Regulatory Implications  Regulatory parity (regulating services vs. technologies)  Should a telephone call be regulated differently if it is TDM, VoIP, FTTH, DOCSIS?  Protocol conversion  Is gateway functionality protocol conversion in a CI-II / CI-III context?  Unbundling  What are the UNE’s of a VoIP network?  How should competitive access provided in a VoDSL and FTTH environment?  CPE Deregulation  With gateway functionality moving to the end user

66. 66 Further Reading …  David J Write, Voice over Packet Networks, J. Wiley.  Jonathan Davidson and James Peters, Voice over IP Fundamentals, Cisco Press.  Daniel Minoli and Emma Minoli, Delivering Voice of IP Networks, Wiley Computer Publishing.  David Collins, Carrier Grade Voice over IP, McGraw-Hill.