Convergence Technologies

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 analog 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 practices (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 practices • 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 analog telephone adapter (ATA), single line adapter, soft phones (WiFi, PDA, PC-based), analog 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 organization’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