1 / 102

Convergence Technologies

Convergence Technologies. Lesson 1: Convergent Network Traffic Protocols. Objectives. Compare and contrast circuit-switched and packet-switched technologies, including ways that packets traverse multiple WAN links, and call and call flow descriptions

rumer
Download Presentation

Convergence Technologies

An Image/Link below is provided (as is) to download presentation Download Policy: Content on the Website is provided to you AS IS for your information and personal use and may not be sold / licensed / shared on other websites without getting consent from its author. Content is provided to you AS IS for your information and personal use only. Download presentation by click this link. While downloading, if for some reason you are not able to download a presentation, the publisher may have deleted the file from their server. During download, if you can't get a presentation, the file might be deleted by the publisher.

E N D

Presentation Transcript


  1. Convergence Technologies

  2. Lesson 1:Convergent Network Traffic Protocols

  3. Objectives • Compare and contrast circuit-switched and packet-switched technologies, including ways that packets traverse multiple WAN links, and call and call flow descriptions • Define the Realtime Transport Protocol (RTP) and the Realtime Transport Control Protocol (RTCP) • Identify the components of Session Initiation Protocol (SIP) and describe the format of an SIP Uniform Resource Identifier (URI) • Identify the functions of signaling protocols for converged networks (e.g., Session Initiation Protocol [SIP], H.323, H.225, H.320, H.450, Media Gateway Control Protocol [MGCP], Media Gateway Control [Megaco]) • Compare and contrast the functions of gatekeepers, gateways and proxies in relation to SIP and H.323 devices • Compare and contrast SIP, H.323 and Megaco/MGCP

  4. Defining Convergence • Convergence– The integration of telephony and data technologies • Integration includes: • Placing the voice network (telephony), the video network (television, satellite) and the Internet (rich media) onto common platforms

  5. Smart Network and Dumb Network

  6. Circuit-Based vs.Convergence Calling • Circuit-switched network – uses a dedicated physical path to send and receive information • Circuit-based calls: • Provide very good voice quality • May fail if the destination is busy or the network fails at any point in the connection • Packet-switched network – places addressing information into data packets • Convergence-based calls: • Dynamically reroute packets to other network nodes if a network node fails • Result in increased latency because packetization and compression add processing time to the signal

  7. Transport Through a Packet-Switched Network • Packets are encapsulated in Ethernet frames • At Layer 4, source and destination port numbers are added • At Layer 3, source and destination IP addresses are added • At Layer 2, source and destination MAC addresses are added

  8. User Datagram Protocol • UDP header is very simple, consisting of source and destination port numbers, a length field, and a checksum field

  9. Realtime Transport Protocol (RTP) • Used to transport voice and video payloads for real-time applications • Provides end-to-end delivery services • Runs over both UDP and TCP • Uses even port numbers that are generally assigned dynamically • Default port is 5004 • RTP profiles define a set of codes for each type of payload

  10. RTP Packets • RTP packets are encapsulated in UDP packets

  11. Realtime Transport Control Protocol (RTCP) • Does not transport any data itself • Partners with Realtime Transport Protocol (RTP) • Monitors the media stream • Provides feedback on the Quality of Service (QoS) being provided by RTP • While RTP uses an even port number, RTCP always uses the next odd port number • Default port is 5005

  12. Session Initiation Protocol (SIP) • Signaling protocol only — does not deliver media streams, nor does it control the delivery of media streams • Initiates and manages sessions (or connections) between 2 or more participants • Primary function is to set up, modify and tear down a connection • Developed by the IETF, SIP is modeled after Hypertext Transfer Protocol (HTTP)

  13. SIP Related Protocols • Session Description Protocol (SDP) • Describes the characteristics of end points in a session • Multiprotocol Label Switching (MPLS) • Can provide QoS for SIP connections • Resource Reservation Protocol (RSVP) • Can provide QoS for SIP connections • Differentiated Services (DiffServ) • Can provide QoS for SIP connections

  14. SIP ports and URIs • SIP uses both UDP and TCP ports 5060 by default • SIP URI takes the following format: sip:user@host • SIP URI examples: sip:555-1110@ctp-certified.com sip:charles.chaplin@64.128.206.2 sip:charles.chaplin@sip.ctp-certified.com

  15. SIP Components • User agents • User agent client (UAC): initiates an SIP request • User agent server (UAS): responds to SIP request • Servers: • Proxy: perform routing, authentication and accounting functions • Redirect: relays information to a user agent, such as the IP address of the party to be called • Registrar: enables a client to let a proxy or redirect server know how the client can be reached

  16. SIP Messages • Requests • INVITE • ACK • BYE • Cancel • Options • Register • Each request (except for an ACK request) requires a response

  17. SIP Messages (cont'd) • Responses are composed of a 3-digit Status Code and an associated Reason Phrase

  18. SIP Calls Session Invitation • Consists of one INVITE request, usually sent to an SIP proxy • A 200 OK response is generated when the called party answers the phone • Media streams are sent directly between end points

  19. H.323 • Defines the following: • How an audiographic call is set up across a network • How to negotiate capabilities • How to transmit data and control conferencing • Which default audio and video codecs to use

  20. H.323 Architecture • Terminals • H.323 end points • Can be a stand-alone device (IP phone) or a logical device within a PC • Includes audio and video codecs • Must support H.245 for capabilities negotiation • Uses Q.931 for call signaling and setup • Uses H.225 RAS for communicating with gatekeepers • Must support RTP and RTCP

  21. H.323 Architecture (cont'd) • Gateways • Connect and translate protocols between dissimilar networks • Provide protocol translation, media format conversion and data transfer between H.323 and non-H.323 networks • Optional element; not required for connections within one LAN • Required to establish connections between terminals in H.323 networks and terminals in networks with different protocols

  22. H.323 Architecture (cont'd) • Gatekeeper functionality: • Admission control • Address translation • Bandwidth control • Zone management • Call control for point-to-point conferences • Codec translation • Call authorization • Bandwidth and call management • Accounting and billing • Call routing • Multipoint Control Unit (MCU) – required whenever three or more H.323 terminals are connected

  23. H.323 Protocol Stack

  24. H.225 RAS • RAS messages (requests and responses) are sent between end points and gatekeepers via UDP • Gatekeeper messages are sent for gatekeeper discovery (GRQ, GCF, GRJ) • Registration messages are sent for negotiating a registration with a gatekeeper (RRQ, RCF, RRJ) • Admission messages are requests and replies for address translation (ARQ, ACF, ARJ) • Status messages are used to monitor end point status during calls that are routed through a gatekeeper (IRQ, IRR) • Disengage messages signal the end of a call (DRQ, DCF)

  25. H.323 Calls • In a typical call: • A client contacts a gatekeeper and requests an address using H.225 RAS admission request (ARQ) • Gatekeeper forwards address to the client • Client establishes session using H.225 • Session is negotiated using H.245

  26. H.323 Calls (cont'd) H.225 call signaling is used between terminals to set up and tear down a connection

  27. H.323 Calls (cont'd) H.245 call control signaling is used for negotiating capabilities and master/slave determination

  28. Media Gateway Control Protocol (MGCP) • Media Gateway Control Protocol (MGCP) – a signaling protocol used in IP telephony systems • MGCP controls media gateways by sending signals from a media gateway controller • MGCP is a master/slave protocol • MGCP assumes that call logic and call state are maintained by intelligent end points

  29. Network Call Signaling (NCS) • Network Call Signaling (NCS) – a protocol that creates embedded agents to use MGCP in a network

  30. Megaco/H.248 • Enhanced version of MGCP • Result of a joint effort between IETF and ITU • Megaco enables the separation of call control from media conversion • Megaco instructs an MG to connect streams coming from outside a packet or cell data network onto a packet or cell stream such as Realtime Transport Protocol (RTP) streams

  31. SIP vs. H.323 vs. Megaco

  32. Summary • Compare and contrast circuit-switched and packet-switched technologies, including ways that packets traverse multiple WAN links, and call and call flow descriptions • Define the Realtime Transport Protocol (RTP) and the Realtime Transport Control Protocol (RTCP) • Identify the components of Session Initiation Protocol (SIP) and describe the format of an SIP Uniform Resource Identifier (URI) • Identify the functions of signaling protocols for converged networks (e.g., Session Initiation Protocol [SIP], H.323, H.225, H.320, H.450, Media Gateway Control Protocol [MGCP], Media Gateway Control [Megaco]) • Compare and contrast the functions of gatekeepers, gateways and proxies in relation to SIP and H.323 devices

  33. Lesson 2:Implementing VoIP

  34. Objectives • List essential steps for qualifying a network's ability to support convergence (e.g., cable inspection, existing and maximum device capacity, replacing hubs with switches, Power over Ethernet [PoE] requirements, VLAN creation, conducting network reconnaissance) • Describe the features of Telephony Application Programming Interface (TAPI) and Messaging Application Programming Interface (MAPI) in a converged solution • Implement Telephone Number Mapping (ENUM), elements of global and private numbering plans, Local Number Portability (LNP)/Wireless LNP, end-point addressing, path selection, calling classes, digit manipulation, overlapping number ranges • Identify common G.7xx codecs and their bandwidth requirements in a converged environment (e.g., G.711, G.729, G.729a, G.726 and others)

  35. Objectives (cont'd) • Describe the impact of compression on voice quality, and identify issues involved when converting voice to analogue and digital formats • Identify benefits and drawbacks of various codecs in relation to bandwidth and voice quality • Calculate and estimate bandwidth usage for various codecs, including considerations of overhead, connection quality, and other factors that affect theoretical calculations (e.g., capacity planning, choosing connection speeds) • Recommend codecs for use with local/in-network/within-LAN calls, and for across WAN connections • Explain wireless convergence technologies, including Digital Enhanced Cordless Telecommunications (DECT) and DECT layers, Personal Wireless Telephone (PWT), Generic Access Profile (GAP), expected ranges for interference-free communication, and the MHz ranges for each standard

  36. Objectives (cont'd) • Identify the elements of the IP Multimedia Subsystem (IMS) • Explain real-time faxing, according to standards such as ITU T.38 • Explain store-and-forward faxing, according to standards such as ITU T.37 • Identify the features, benefits, problems and management of presencing, including single sign-on, features available in various devices • List unified message methods and benefits (e.g., fax, voice, text, video) • Identify common and essential videoconferencing codecs, standards and practises (e.g., Moving Picture Experts Group [MPEG], Quarter Common Intermediate Format [QCIF], etc.), and choose the appropriate codecs for various bandwidths

  37. Objectives (cont'd) • Summarize television/video-calling standards and practises • Identify multimedia conferencing standards, including all subsets of T.120 (e.g., T.123, T.124, T.135) • Explain fundamentals of Internet Protocol television (IPTV), including set-top box, Video on Demand (VoD), accepted codecs (e.g., Video Codec [VC-1]) • Identify the purpose and function of voice and videoconferencing hardware (e.g., Multipoint Control Unit [MCU], set-top box, Session Border Controller [SBC]) • Compare and contrast traditional and IP-based private branch exchange (PBX) systems • Identify convergent terminal equipment and software, including analogue telephone adapter (ATA), single line adapter, soft phones (WiFi, PDA, PC-based), analogue phones, time division multiplexer (TDM), protocol-specific handsets (e.g., SIP, Megaco)

  38. Objectives (cont'd) • Explain power issues, including redundancy planning, Power over Ethernet (PoE)/802.3af, PoE classes, expected voltage, wattage, power sourcing equipment (PSE), powered devices (PDs)

  39. Planning aConvergent Network • Major phases of an implementation plan include these steps: • Identifying expectations • Determining bandwidth requirements • Performing a network health check • Creating a phased deployment plan

  40. Identifying Expectations • Identify how network(s) will be used • Identify specific protocols that will be used • Identify and explain potential challenges

  41. Determining Bandwidth Requirements • Identify current digital connection • Determine bandwidth required by existing network • Monitor current network performance • Evaluate current network performance • Calculate additional requirements for VoIP • Take wide area network (WAN) links into account • Take growth into account

  42. Performing a Network Health Check • Check network cabling • Replace hubs with Layer 2 switches • Implement VLANs • Prioritize VLAN traffic • Check routers • Identify the entity that manages Internet router • Examine current IP addressing scheme • Examine Domain Name System (DNS) • Examine firewall • Identify whether NAT will be implemented • Identify whether VPNs must be supported • Identify whether any part of the LAN will be wireless

  43. Creating a Phased Deployment Plan • Create a detailed, approved implementation plan • Use a test network • Deploy incrementally • Do not begin with the sales department

  44. TAPI and MAPI • Telephony Application Programming Interface (TAPI) is an API used for connecting a Windows PC to telephone services • Messaging Application Programming Interface (MAPI) is a Windows API that allows different e-mail applications to work together to distribute mail

  45. Private numbering plans allow a company to create its own numbering system Extensions can be created based on an organisation’s needs Number plan defines the format of telephone numbers Implementing VoIP involves designing a numbering plan and a dial plan. Dial plan must include rules for dealing with: End point addressing Path selection Calling classes Digit manipulation Overlapping number ranges Numbering Plans

  46. Telephone Number Mapping (ENUM) • Maps E.164 telephone numbers into the Domain Name System (DNS) • Creates a dynamic mapping of E.164 addresses to IP addresses • ENUM domain names are hosted in the e164.arpa domain • A telephone number such as +1 (602) 555-1212 is converted into the ENUM domain name 2.1.2.1.5.5.5.2.0.6.1.e164.arpa • ENUM domain name resolves to one or more DNS NAPTR records

  47. G.7xx Codecs • Various codecs provide different amounts of compression • Compression allows more voice traffic, but can also: • Introduce delay • Adversely affect voice quality • Put a significant strain on CPU resources, depending on the complexity of the algorithm and the amount of compression

  48. Comparison of G.7xx Codecs

  49. Calculating VoIP Bandwidth Requirements • Calculations for bandwidth requirements must factor in: • Codec, sample period and frame size • Frames per packet • IP overhead • Ethernet overhead • Number of simultaneous calls • Silence suppression • Compressed headers

  50. Wireless Convergence Technologies • Components • Radio exchange • Base stations (transceivers) • Portable phones • Digital Enhanced Cordless Telecommunications (DECT) is an ETSI standard for digital portable phones • Generic Access Profile (GAP) guarantees interoperability between any handset and any base station, regardless of make or model • Operates in the 1880 MHz to 1900 MHz band in Europe, Africa, Australia and Asia (except China) • Operates in the following bands in North America: 902 MHz to 928 MHz, 2400 MHz to 2483.5 MHz, 5725 MHz to 5850 MHz

More Related