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VoIP Conception and Implementation

VoIP Conception and Implementation. LANtel Telecommunication Corp. Senior Product Manager Jeremy Chan. Agenda. What’s VoIP and IP Telephony (IPT) VoIP Applications VoIP QoS Issue VoIP Architecture VoIP Signaling Fax over IP (FoIP) . What’s VoIP and IP Telephony (IPT). VoIP. VoIP :

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VoIP Conception and Implementation

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  1. VoIP ConceptionandImplementation LANtel Telecommunication Corp. Senior Product Manager Jeremy Chan

  2. Agenda • What’s VoIP and IP Telephony (IPT) • VoIP Applications • VoIP QoS Issue • VoIP Architecture • VoIP Signaling • Fax over IP (FoIP)

  3. What’s VoIP and IP Telephony (IPT)

  4. VoIP VoIP : • VoIP, Voice over Internet Protocol, is the technology that uses the Internet Protocolto transmit voice conversation over a data network. • The primary advantages of moving voice over a data network are increased efficiency and decrease cost.

  5. IPT (IP Telephony) IPT (IP Telephony) : • An IP Communications System that provides a high availability and scalability telephony system. • Provide support industry standards such as H.323, MGCP, SIP, JTAPI, TAPI, … etc. VoIP signal protocol.

  6. Enterprise Voice Solution PBX PBX PSTN PBX PBX IP WAN Router/GW Router/GW Soft-Switch Soft-Switch IP WAN Router/GW Router/GW Application Servers

  7. Packet Voice Technology

  8. VoIP Applications

  9. Packet Network PSTN Branch Office Application HQ Branch 1 IWF Server Farm Telephone IWF Branch N PBX Telephone IWF Telephone *IWF: Interworking function

  10. Packet Network Interoffice Trunking Application PBX PBX Telephone Telephone

  11. Interoffice Trunking Application

  12. VoIP QoS Issue

  13. VoIP QoS Issue • Delay • Algorithmic Delay • Processing Delay • Network Delay • Jitter • Lost-Packet Compensation • Echo Compensation

  14. Delay • Algorithmic Delay • Collect a frame of voice samples to be processed by the voice coder. • G.726 adaptive differential pulse-code modulation (ADPCM) (16, 24, 32, 40 kbps)—0.125 microseconds • G.728 LD–code excited linear prediction (CELP)(16 kbps)—2.5 milliseconds • G.729 CS–ACELP (8 kbps)—10 milliseconds • G.723.1 Multirate Coder (5.3, 6.3 kbps)—30 milliseconds

  15. Delay • Processing Delay • Actual process of encoding and collecting the encoded samples into a packet for transmission over the packet network. • The encoding delay is a function of both the processor execution time and the type of algorithm used.

  16. Delay • Network Delay • Physical medium and protocols used to transmit the voice data and by the buffers used to remove packet jitter on the receive side. • Network delay is a function of the capacity of the links in the network and the processing.

  17. Delay Causes Problems • Echo • Signal reflections of the speaker's voice from the far-end telephone equipment back into the speaker's ear. • Round-trip delay becomes greater than 50 milliseconds. (G.131) • Talker overlap • one talker stepping on the other talker's speech • the one-way delay becomes greater than 250 milliseconds. (G.114)

  18. Jitter Variable interpacket timing caused by the network a packet traverses. • Removing jitter: collecting packets and holding them long enough to allow the slowest packets to arrive in time to be played in the correct sequence. • Causes additional delay

  19. Lost-Packet Compensation Lost packets can be an even more severe problem, depending on the type of packet network that is being used. • Interpolate for lost speech packets by replaying the last packet received during the interval. • Send redundant information. • Use a hybrid approach with a much lower bandwidth voice coder to provide redundant information. • Avoiding and Managing network congestion

  20. Echo Normal Telephony Call Normal Telephony Call with an Echo

  21. Echo Compensation • Signal reflections generated by the hybrid circuit that converts between a four-wire circuit (a separate transmit and receive pair) and a two-wire circuit (a single transmit and receive pair). • The round-trip delay through the network is almost always greater than 50 milliseconds. • ITU standard G.165 defines performance requirements that are currently required for echo cancellers.

  22. VoIP Architecture

  23. VoIP–Embedded Software Architecture

  24. Voice Packet Software Module • digital-signal processor (DSP) • Telephony-Signaling Gateway Software Module • Translating signaling into state changes used by the packet protocol module to set up connections. • Packet Protocol Module • processes signaling information and converts it. • Network-Management Module • Voice-management interface to configure and maintain the other modules

  25. VoIP Signaling

  26. Signaling – H.323 • H.323 • Umbrella standard covering multimedia communications over LANs that do not provide a guaranteed Quality of Service. (H.323 v1) • Entities • Terminals • Gateways • Gatekeepers • MCUs • Protocols • Parts of H.225.0 - RAS, H.225 (Q.931) • H.245 • RTP/RTCP • Audio/video codecs

  27. G.711 G.728 G.722 G.729 H.261 H.263 G.723.1 H.450.3 H.450.2 H.450.1 H.323 Protocol Stack Data Audio Signal T.127 Video Signal T.126 Presentation Session Transport T.124 RTCP RAS RTP Supplementary Services T.125/T.122 H.235 X.224.0 Control UDP TCP H.245 H.225 Network IP Data Link Physical

  28. H.323 protocols • H.225 Covers narrow-band visual telephone services • H.225 Annex G • H.235 Security and authentication • H.245 Negotiates channel usage and capabilities • H.450.1 Series defines Supplementary Services for H.323 • H.450.2 Call Transfer supplementary service for H.323 • H.450.3 Call diversion supplementary service for H.323 • H.450.4 Call Hold supplementary service • H.450.5 Call Park supplementary service • H.450.6 Call Waiting supplementary service • H.450.7 Message Waiting Indication supplementary service • H.450.8 Calling Party Name Presentation supplementary service • H.450.9 Completion of Calls to Busy Subscribers supplementary service • H.450.10 Call Offer supplementary service • H.450.11 Call Intrusion supplementary service • H.450.12 ANF-CMN supplementary service • H.261 Video stream for transport using the real-time transport • H.263 Bitstream in the RTP Q.931manages call setup and termination • RAS Manages registration, admission, status • RTCP RTP Control protocol • RTP Real-Time Transport • T.38 IP-based fax service maps • T.125 Multipoint Communication Service Protocol (MCS).

  29. H.323 Architecture

  30. Typical H.323 Deployment

  31. Signaling – MGCP, MAGACO • Media Gateway Control Protocol • Using packages model and providing an centralized architecture where call control and services. • Controlling Telephony Gateways from external call control elements called media gateway controllers or call agents. • Entities • MGC (Media Gateway controller / Call agent) • MG (Media Gateway) • Protocols • MGCP v1 – RFC 2705 • H.248 (H.248 / MAGACO) – RFC 3525 • SDP (Session Definition Protocol) - RFC 3407

  32. PSTN MGCP Architecture MGCP Voice Gateway T1/E1 FXO/FXS E&M PBX IP Phone ( MGCP Client ) IP Phone ( MGCP Client ) MGCP Call Agent RTP

  33. Signaling – SIP • Session Initiation Protocol • Multimedia protocol that could take advantage of the Internet model for building VoIP networks and applications. Using distributed architecture. • Entities • User Agent • Gateways • Proxy Server • Redirect Server • Registrar Server • Protocols (RFC 2543 v1, RFC 3261 v2) • SDP ( Session Definition Protocol ) • URLs • DNSs • TRIP ( Telephony Routing Over IP

  34. SIP Architecture

  35. ENUM • “ENUM protocol is defined by RFC 2916, aiming at translating the numbers stemming form the ITU-T E.164 Recommendation into Internet Domain Names; ENUM is an opportunity for developing the information society.” • “As a matter of fact, ENUM allows to use a traditional telephone number in the context of different communications media, in particular those rising from the development of IP networks (e-mail, VoIP, …) and therefore, could facilitate the penetration of new applications into the mass market easily ( this market is accustomed to E.164 numbers).”

  36. ENUM (Cont.) • ENUM is part of Convergence • ENUM is part of series of technical initiatives underway in both the IETF and ITU to develop Internet Telephony Standards. • Call Setup – H.323 – SIP • Quality of Service – DIFFSERV – INTSERV – MPLS • PSTN – IP Interworking H.248/MEGACO • FAX – T.37, T.38 – RFC 2503 • Mobile – 3GPP related • ENUM is about new service creation • It must address naming and numbering issues

  37. VoIP Signaling Comparison

  38. VoIP Signaling Comparison

  39. Fax over IP

  40. FAX over IP • ITU and Internet Engineering Task Force (IETF) are working together to continue to evolve both the real-time FoIP network standard (T.38) as well as the store-and-forward FoIP network standard (T.37). • T.38 is the fax transmission protocol selected for H.323.

  41. FoIP QoS • Timing • network delay • processing delay • IWF must compensate for the loss of a fixed timing of messages over the packet network. • Jitter • collect packets and hold them long enough so that the slowest packets to arrive are still in time to be played in the correct sequence. • Lost-Packet Compensation • repeating information in subsequent frames • using an error-correcting protocol

  42. Reference • Cisco • Introduce H.323 • SIP Presentation • REDCOM • H.323 Tutorial • IEC • Voice and Fax over Internet Protocol (V/FoIP) • ENUM.ORG • Study Group A Presentation on ENUM • IETF • ftp://ftp.isi.edu/in-notes/rfc2916.txt -- ENUM Core Protocol • ftp://ftp.isi.edu/in-notes/rfc3261.txt -- SIP • ftp://ftp.isi.edu/in-notes/rfc2705.txt -- MGCP

  43. Thank You

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