Voice over IP 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 2 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 3 Circuit switching served voice well for 100 years! Signal System 7 Data link Signal Transfer Point Trunk Group Loop User - A Class 5 Switching System User - B Central Office - A Transit Office Connection Through Switching Fabric Central Office - B Class 4 Switching System 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 4 Packet Switching Well-matched for data transmission Packet Payload Header Input Buffer Hdr. Trans Routing Fabric Output Buffer Hdr. Trans 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 5 VoIP Network Architecture Gateway Gatekeeper PSTN network IP network Media Gateway Media Gateway Media gateways provide voice packetization Gatekeepers provides call control logic and permissions Gateway provides interworking with ISDN, SS7 and signaling of PSTN (POTS) 6 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 7 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 overprovisioning 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 8 PSTN versus VoIP Today’s PSTN VoIP TDM circuit switching Packet switching QoS guarantees Yes No Network resource reserved at call setup Yes No Class 4, Class 5 switching systems Mostly integrated in switching system Gateways, gateway controllers, routers In separate gateway controllers 64 kb/s Variable 5.3 – 32 kb/s DTMF, SS7 SIP, H.323, MGCP Underlying Technology Network elements Call processing intelligence Bandwidth per call Signaling Transport How reliability achieved 9 ATM, FR, native IP in TDM in access, edge, access; ATM native IP core in core, WiFi Redundancy within Redundant routes each network through network element 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 10 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 11 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 $19 billion by 2004. IP Telephony Services: Market Review and Forecast, 1998-2004. 12 Growth in VoIP 25.0 Early growth from expense savings Later growth from revenue 20.0 Revenues ($ billion) generation from new services 15.0 and CLECs 10.0 Later deployment by incumbent carriers 5.0 0.0 2000 13 Early deployment by enterprises 2001 2002 2003 (source: Frost & Sullivan) 2004 2005 2006 VoIP Applications Some trends can be discerned: First wave: Bypassing the PSTN Second wave: Replacing the PSTN Third wave: Value-added services PSTN DLC 14 Class 5 Class 5 DLC 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 RADIUS server RADIUS server Internet modem DLC 15 Class 5 RAS RAS modem Class 5 DLC 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 RADIUS server Gate Keeper Internet modem DLC 16 Class 5 RAS VoIP Gateway Class 5 DLC 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… Gate Keeper IP network DLC 17 Class 5 VoIP Gateway VoIP Gateway Class 5 DLC 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 Gate Keeper RADIUS server IP network DLC 18 Class 5 VoIP Gateway RAS modem Class 5 DLC 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) Soft switch IP network DLC 19 Class 5 Trunking Gateway Access Gateway DLC PSTN replacement – Integrated access network Integrating Access Gateway into DLC Required: “Next Gen” DLC, with integrated IP gateway Issues: immaturity of standards Soft switch IP network NexGen DLC 20 NexGen DLC 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) A B IP network IP-PABX IP-PABX PSTN 21 IP-phone 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 Gate Soft Keeper switch VoIP Gateway 22 IP PSTN network Class 5 Value Added Services Converged services Internet Call Waiting Click to Call Unified messaging … Video telephony (3rd time right?) 23 Standards for VoIP The H.323 Protocol Stack System control user interface H.225 RAS channel Q.931 call setup H.245 control Mic Audio And Video Control Camera Data applications Video Codec H.261 H.263 T.120 Audio codec G.711 G.723 G.729 RTCP RTP Transport Layer (TCP or UTP) IP 25 H.225 RAS Control Gatekeeper Endpoint H.225 Multiport Control Unit Gateway 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 26 Call Signaling in H.232 Q.931 H.245 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 27 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 28 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 29 Softswitch Architecture To other Softswitches SIP-T MGCP Or Megaco Softswitch IP Network PSTN Network Trunk Gateway Access Gateway Softswitch separates function of Gateway from the media gateway 30 ATM QoS Parameters Peak-to-peak cell delay variation Maximum cell transfer delay Negotiated at start of call Cell loss ratio Cell error ratio Severely errored cell block ratio Cell misinsertion rate 31 Controlled via Network design Real-Time Multimedia over ATM (RMOA) PSTN Switch H.323 Gateway VoIP Gateway IP Network ATM network H.323 Gateway PSTN Switch VoIP Gateway 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 32 Resource Reservation Protocol (RSVP) Host Router RSVP Process Application Control Policy Control RSVP Process Routing Process Control Admission Control Packet Classifier Packet Scheduler Policy Control Admission Control Packet Classifier Packet Scheduler 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 33 Adding QoS to IP Networks: Diffserv Meter Classifier Marker Shaper / Dropper 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 34 VoIP access via DSL and Cable Modems Cable Telephony Video Content Internet Service Head end Fiber Node PSTN Gateway 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 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 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 North America Cable Telephony Market Size 1 6 ,0 0 0 1 4 ,0 0 0 1 2 ,0 0 0 1 0 ,0 0 0 8 ,0 0 0 6 ,0 0 0 4 ,0 0 0 2 ,0 0 0 0 C irc uit S w itc he d V o IP 05 20 04 20 03 20 02 20 01 20 20 19 00 T o ta l 99 Million H ouseholds N o rth A m e ri c a C a b l e T e l e p h o n y Cable projected to capture 15 % telephony market share by 2005 Shift from proprietary TDM solutions towards VoIP DOCSIS Residential VoIP happening first in the Cable Access Market Voice over DSL CO PSTN CO / CEV GR303 Voice Gate Way Data Network HOME/BUSINESS ADSL DS3 / OC-3 Class 5 Switch 4-16 ATM Switch 1 VC for Voice 1 VC for Data DSLAM LAN Integrated Access Device 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 40 Alternatives for VoDSL Voice over ADSL Alternatives • Voice over IP IP Layer 3 • Voice over ATM ATM Layer 2 • Voice over TDM DMT Layer 1 • Voice in separate spectrum (e.g., ADSL over DAML) Analog Spectrum 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 41 Quality issues for the transport of voice over packet-based networks The three essential stages of packet-based voice transport one-way Mouth-to-Ear (M2E) delay overall distortion (codec & packet loss) (Concatenation of) Packet-based Network(s) Encoding and packetization stage Packet transport stage Dejittering and decoding stage Echo control performed close to destination 43 Components of the M2E delay Packetization delay Total minimal delay Total queuing delay Dejittering delay M2E delay Packetization delay is chosen by the source terminal or ingress GW Minimal delay and queuing delay depend on QoS provided by traversed network(s) Each network component has its specific contribution Dejittering delay is chosen by the destination terminal or egress GW 44 Trade-off M2E delay vs. packet loss in destination or egress GW Pdf(delay) Dejittering delay Packet loss Minimal Delay of delay first packet M2E 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 45 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 46 Trade-offs Network (transport) parameters minimal delay delay jitter packet loss Echo control Dejittering delay Voice quality 47 Packet size Codec Header compression Efficiency of transport 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 48 Major Parameters of Standard Codecs Origin Standard Type Codec Bit rate G.711 PCM 64 Voice Frame (ms) Look ahead (ms) Algor. delay (ms) le Intrinsic quality 0 94.3 50 44.3 24 25 69.3 32 7 87.3 2 92.3 20 74.3 7 87.3 10 84.3 19 75.3 15 79.3 16 0.125 G.726 G.727 0 0.125 ADPCM 40 0.125 0 0.125 ITU-T 12.8 G.728 LD-CELP 0.625 0 0.625 16 G.729(a) CS-ACELP 8 ACELP 5.3 G.723.1 ETSI 49 10 5 15 30 7.5 37.5 MP-MLQ 6.3 GSM-FR RPE-LTP 13 20 0 20 20 74.3 GSM-SR VSEPL 5.6 20 0 20 23 71.3 GSM-ESR ACEPL 12.2 20 0 20 5 89.3 Influence of packet loss on distortion 100 90 Intrinsic rating R int 80 70 60 50 40 G.729(A) + VAD G.723.1@6.3 kb/ s + VAD GSM-EFR G.711 with PLC G.711 without PLC 30 20 10 0 0 2 4 6 8 10 packet loss ratio (%) 50 12 14 16 Transcoding matrix Transcoding is the translation of one codec format into another (via the linearly quantized 8 kHz sampled voice format) CODEC G.711 (64kb/s) G.726 (40kb/s) G.726 (32kb/s) G.726 (24kb/s) G.726 (16kb/s) G.728 (16kb/s) GSM-FR (13kb/s) G.728 (12.8kb/s) GSM-EFR (12.2kb/s) G.729 (8kb/s) G.723.1 (6.3kb/s) GSM-HR (5.6kb/s) G.723.1 (5.3kb/s) 51 G.711 G.726 G.726 G.726 G.726 G.728 GSM-FR G.728 GSM-EFR G.729 G.723.1 GSM-HR G.723.1 (64kb/s) (40kb/s) (32kb/s) (24kb/s) (16kb/s) (16kb/s) (13kb/s) (12.8kb/s) (12.2kb/s) (8kb/s) (6.3kb/s) (5.6kb/s) (5.3kb/s) 94.3 92.3 87.3 69.3 44.3 87.3 74.3 74.3 89.3 84.3 79.3 71.3 75.3 92.3 90.3 85.3 67.3 42.3 85.3 72.3 72.3 87.3 82.3 77.3 69.3 71.3 87.3 85.3 80.3 62.3 37.3 80.3 67.3 67.3 82.3 77.3 72.3 64.3 68.3 69.3 67.3 62.3 44.3 19.3 62.3 49.3 49.3 64.3 59.3 54.3 46.3 50.3 44.3 42.3 37.3 19.3 0 37.3 24.3 24.3 39.3 34.3 29.3 21.3 25.3 87.3 85.3 80.3 62.3 37.3 80.3 67.3 67.3 82.3 77.3 72.3 64.3 68.3 74.3 72.3 67.3 49.3 24.3 67.3 54.3 54.3 69.3 64.3 59.3 51.3 55.3 74.3 72.3 67.3 49.3 24.3 67.3 54.3 54.3 69.3 64.3 59.3 51.3 55.3 89.3 87.3 82.3 64.3 39.3 82.3 69.3 69.3 84.3 79.3 74.3 66.3 70.3 84.3 82.3 77.3 59.3 34.3 77.3 64.3 64.3 79.3 74.3 69.3 61.3 65.3 79.3 77.3 72.3 54.3 29.3 72.3 59.3 59.3 74.3 69.3 64.3 56.3 60.3 71.3 69.3 64.3 46.3 21.3 64.3 51.3 51.3 66.3 61.3 56.3 48.3 52.3 75.3 73.3 68.3 50.3 25.3 68.3 55.3 55.3 70.3 65.3 60.3 52.3 56.3 M2E delay and packet loss bounds Bounds under perfect echo control If there is no packet loss, the M2E delay can exceed 150 ms Origin codec bit M2E delay Standard rate (kb/s) bound (ms) G.711 G.726 G.727 ITU-T G.728 G.729(A) G.723.1 ETSI 52 GSM-FR GSM-HR GSM-EFR 64 16 24 32 40 12.8 16 8 5.3 6.3 13 5.6 12.2 400 NA NA 324 379 212 324 296 221 253 212 180 345 If the M2E delay is below 150 ms some packet loss can be tolerated Origin ITU-T ETSI codec bit PL bound rate (kb/s) (%) G.711 without PLC 64 1 G.711 with PLC 64 10 G.729(A) + VAD 8 3.4 6.3 2.1 G.723.1@6.3 kb/s + VAD GSM-EFR 12.2 2.7 Standard Quality of a telephone conversation (using the E-model of ITU-T Rec. G.107 and G.109) 45 Perfect echo control 40 (Very) Bad 35 Poor distortion 30 25 Medium 20 15 High 10 5 Best 0 0 50 100 150 200 M2E delay (ms) 53 250 300 350 400 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 54 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 55 Addressability in VoIP Networks Addressibility in VoIP Question: How do you dial a VoIP user if all you have is their telephone number? alcatel.com ibm.com ge.com fcc.gov Users resistant to change services if they have to change phone numbers 57 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 58 How does ENUM work? DNS-A (9.1.9.1.e164.arpa) DNS-B (0.5.8.9.1.9.1.e164.arpa) proxy.com INVITE “(919) 850-5500" 59 INVITE Regulatory Considerations 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). 61 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. 62 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. 63 The EU view Advocates the principle of technological neutrality. EU has a strict definition of voice telephony in terms of the following four principles: 64 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. 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 65 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. 66