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H.323 Chapter 4. Internet Telephony 4-2 Introduction We have learned IP, UDP, RTP The set-up and tear-down of the sessions Signaling In traditional telephony.

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Presentation on theme: "H.323 Chapter 4. Internet Telephony 4-2 Introduction We have learned IP, UDP, RTP The set-up and tear-down of the sessions Signaling In traditional telephony."— Presentation transcript:

1 H.323 Chapter 4

2 Internet Telephony 4-2 Introduction We have learned IP, UDP, RTP The set-up and tear-down of the sessions Signaling In traditional telephony networks ISUP, Integrated Services Digital Network User Part A component of the Signaling System 7 (SS7)

3 Internet Telephony 4-3 H.323, ITU-T recommendation The 1 st version, 1996 Visual Telephone Systems and Equipment for Local Area Network which Provide A Non-Guaranteed Quality of Service Version 2, 1998 Packet-based Multimedia Communications Systems Widely implemented in VoIP solutions Version 4

4 Internet Telephony 4-4 The H.323 Architecture Entities Terminals Gateways Gatekeepers MCUs Protocols Parts of H.225.0 - RAS, Q.931 H.245 RTP/RTCP Audio/video codecs

5 Internet Telephony 4-5 H.323 Architecture H.323 Network Architecture and Components

6 Internet Telephony 4-6

7 Internet Telephony 4-7 Terminals Endpoint on a LAN Supports real-time, 2-way communications with another H.323 entity Must support: Voice - audio codecs Signaling and setup - Q.931, H.245, RAS Optional support: Video Data

8 Internet Telephony 4-8

9 Internet Telephony 4-9 Gateways Interface between the LAN and the switched circuit network Mandatory Functions Transmission Format Translation Communication Procedure Translation Call Setup and Clearing On Both Sides Optional Function Media Format Translation Example: IP/PSTN gateway

10 Internet Telephony 4-10

11 Internet Telephony 4-11 Gatekeepers Optional, but must perform certain functions if present e.g., Netmeeting does not use gatekeepers? Manage a zone (a collection of H.323 devices) Usually one gatekeeper per zone alternate gatekeeper might exist for backup and load balancing Typically a software application implemented on a PC, but can be integrated in a gateway or terminal

12 Internet Telephony 4-12 Mandatory functions: Address translation (routing) Admission control Minimal bandwidth control - request processing Zone management Optional functions: Call control signaling - direct handling of Q.931 signaling between endpoints Call authorization, bandwidth management, and call management using some policy Gatekeeper management information (MIB) Directory services

13 Internet Telephony 4-13 MCUs MCU - Multipoint Control Unit Endpoint that supports conferences between 3 or more endpoints Can be stand-alone device or integrated into a gateway, gatekeeper or terminal Typically consists of multi-point controller (MC) and multi-point processor (MP) MC - handles control and signaling for conference support MP - receives streams from endpoints, processes them, and returns them to the endpoints in the conference

14 Internet Telephony 4-14 MC and MP

15 Internet Telephony 4-15 MCU (MC+MP) MCU (MC+MP) Terminal media stream (unicast) control message Centralized Conference MCU handles both signaling (MC) and stream processing (MP)

16 Internet Telephony 4-16 De-centralized Conference MCU handles only signaling streams go directly between endpoints In this case MCU functions without MP MCU (MC) MCU (MC) Terminal media stream (multicast) control message media stream (multicast) control message media stream (multicast)

17 Internet Telephony 4-17 Hybrid Multipoint Conference MCU (MC+MP) MCU (MC+MP) Terminal video stream (multicast) video stream (multicast) video stream (multicast) Terminal control message audio stream (unicast) control message audio stream (unicast) control message audio stream (unicast)

18 Internet Telephony 4-18 Mixed Multipoint Conference Terminal MCU (MC+MP) MCU (MC+MP) multicast audio and videounicast audio and video Decentralized sideCentralized side

19 Internet Telephony 4-19 Overview of H.323 Singaling Audio codecs (G.711, G.723.1, G.728, etc.) and video codecs (H.261, H.263) Media streams transported on RTP/RTCP RTP carries actual media RTCP carries status and control information RTP/RTCP carried unreliably on UDP Signaling is transported reliably over TCP RAS - registration, admission, status (over UDP) Q.931 - call setup and termination H.245 - capabilities exchange

20 Internet Telephony 4-20 H.323 Protocol Stack

21 Internet Telephony 4-21 Overview of H.323 Protocols H.225.0, a two-part protocol A variant of ITU-T recommendation Q.931, the ISDN layer 3 spec. The set-up and tear-down of connections Call signaling or Q.931 signaling RAS signaling Registration, Admissions, and Status Between endpoints and gatekeepers

22 Internet Telephony 4-22 H.245, control protocols Used between two or more endpoints Manage the media streams of a session Capability exchange RAS, Q.931 and H.245 RAS to obtain permission from a gatekeeper RAS channel Q.931 to establish communication and set up the call Call-signaling channel H.245 to negotiate media parameters H.245 control channel Media streams over logical channels

23 Internet Telephony 4-23 H.323 Addressing An entity in the H.323 network A network address, an IP address URL, Uniform Resource Locator E.g., ras://GK1@somedomain A port number appended RAS, a default port number = 1719 The TSAP, Transport Service Access Point An id for a particular logical channel at a given entity GK UDP Discovery Port = 1718 GK UDP Reg and Status Port = 1719 Call-signaling TCP Port =1720

24 Internet Telephony 4-24 Terminal and gateways Have one or more aliases Can take any number of forms Must be unique within a zone E.164 number Codecs Video codec is optional H.261 QCIF G.711 (A-law and  -law) is mandatory

25 Internet Telephony 4-25 RAS Signaling Used between a GK and endpoints in its zone Since a GK is optional Defined in H.225.0 GK Discovery Registration Unregistration Admission Bandwidth Change Endpoint Location: GK (address translation) Disengage Status Resource Availability: GW → GK Non-standard

26 Internet Telephony 4-26 Gatekeeper Discovery Find a suitably accommodating GK Several GKs, backup GK GRQ – GK-request Known addresses, multicast GK id: if empty, soliciting GKs GCF – GK-Confirm Optionally, indicating one or more GKs to try I cannot help you, but try the GK next door Only one GK can be choosed GRJ – GK-Reject With a reason (e.g., a lack of resource)

27 Internet Telephony 4-27 GK Discovery

28 Internet Telephony 4-28 Endpoint Reg and Reg Cancellation To become controlled by a GK RegistrationRequest (RRQ) RAS signaling port is 1719 Includes An address for RAS messages An address for call-signaling messages An alias One or more redundant addresses and aliases Optional TTL, keepAlive parameters RegistrationReject (RRJ) RegistrationConfirm (RCF) May assign an alias

29 Internet Telephony 4-29 UnregistrationRequest (URQ) Cancel registration By endpoints By GKs UnregistrationConfirm (UCF) UnregistraionReject (URJ) For an unregistered endpoint

30 Internet Telephony 4-30 Endpoint Location Request a real address of an alias LocationRequest (LRQ) To a GK or the GK discovery multicast address A GK can also send an LRQ LocationConfirm (LCF) A call-signaling address An RAS signaling address LocationReject (LRJ) The endpoint is not registered

31 Internet Telephony 4-31 Admission Request permission from a GK to participate in a call AdmissionRequest (ARQ) The type of the call (e.g., two-party or multi-party) The endpoint ’ s own id A call identifier (a unique string) A call-reference value (an integer) Information of the other party Aliases Signaling address Bandwidth (mandatory) TransportQOS: endpoint or GK to reserve the resource

32 Internet Telephony 4-32 AdmissionConfirm (ACF) Many of the same parameters as ARQ A firm order from the GK callModel Optional in ARQ; mandtory in ACF The endpoint sends call signaling directly or via the GK AdmissionReject (ARJ) With a reason Bandwidth, address translation, unregistered endpoint

33 Internet Telephony 4-33 Direct call signaling

34 Internet Telephony 4-34 GK-routed call signaling

35 Internet Telephony 4-35 Pregranted Admission to minimize call setup delay A GK provide an endpoint with admission in advance the RCF includes the parameter preGrantedARQ

36 Internet Telephony 4-36 Bandwidth Change Request an increase or decrease in allocated bandwidth Can change without request if within the limit in ACF BandwidthRequest (BRQ) The new bandwidth, the call id BCF BRJ The GK can also request an endpoint to change the bandwidth The endpoint must comply Closely tied to H.245 signaling (for logical channels) A reduction in bandwidth Require an existing logical channel to be closed and reopened

37 Internet Telephony 4-37 An example bandwidth change

38 Internet Telephony 4-38 Status A GK be informed of the status of an endpoint InformationRequestResponse (IRR) Endpoint information The active call information Call id, call reference value, call type, the bandwidth RTP session information (CNAME, RTP/RTCP address, etc.) IRQ or ACF with an irrFrequency parameter willRespondToIRR parameter IACK INACK An IRR message in error

39 Internet Telephony 4-39 Disengage The end of the call DisengageRequest (DRQ) Call id, call reference value, a disengage reason (e.g., normalDrop) DCF DRJ For an unregistered endpoint not a call party The GK might issue DRQ to an endpoint The endpoint must Close the session Respond to the GK with a DCF message

40 Internet Telephony 4-40 Resource Availability ResourceAvailableIndicate (RAI) An gateway sends to a GK The available call capacity and bandwidth for each protocol almostOutofResource parameter RseourceAvailableConfirm (RAC)

41 Internet Telephony 4-41 Service Control SCI (ServiceControlIndication) SCR (ServiceControlResponse) to enable advanced features Request in Progress RIP A given request takes longer the the timeout period The expected delay and the reason After timeout, the RAS message can be retransmitted

42 Internet Telephony 4-42 Call Signaling For the establishment and tear-down of calls Q.931 modified by rec. H.225.0 Reuse some messages with few modifications Specify a number of rules regarding the usage of information elements defined in Q.931 E.g., no Transit Network Selection, … Certain Q.931 mandatory information forbidden or optional A clever use of User-to-User information element Convey all of the extra information needed in H.323 E.g., H.245 addresses to be used for logical channel

43 Internet Telephony 4-43 Setup The first call-signaling message Q.931 Protocol Discriminator A call reference Bearer Capability RTP information, such as payload type A gateway needs to perform the mapping User-to-User information element Mandatory: Call id, call type, conference id, the caller information Optional: source alias, destination alias, H.245 address, destination call-signaling address

44 Internet Telephony 4-44 Call Proceeding Optional call-establishment procedures are underway Mandatory Protocol discriminator, call reference, and message type, user-to-user User-to-user: similar to the setup message Alerting The called user is being alerted The same parameters as Call Proceeding

45 Internet Telephony 4-45 Progress sent by a called gateway the Cause information – in-band tones the User-to-User info. – same as Call-Proceeding Connect The called party has accepted the call Must be sent if the call is to be completed Call Proceeding and Alerting are optional User-to-User information The same as Call Proceeding Plus Conference Identifier The same as Setup

46 Internet Telephony 4-46 Release Complete Terminate a call No Release message In ISDN, Release and Release Complete Cause information element, optional Otherwise, a Release reason in User-to-User Facility (Q.932) A call should be redirected Also be used for supplementary services User-to-User contains reason parameter E.g., routeCallToGatekeeper

47 Internet Telephony 4-47 Interaction between Call Signaling and H.245 Call signaling: call establishment and tear-down H.245: the negotiation and establishment of media streams When to begin the exchange of H.245 messages? Between the Setup and Connect messages Equipment dependent

48 Internet Telephony 4-48 Call Scenarios Basic Call without GKs Reliable transport

49 Internet Telephony 4-49 Call with GKs and Direct-Endpoint Call Signaling

50 Internet Telephony 4-50 Gatekeeper routed/direct-endpoint call signaling

51 Internet Telephony 4-51 Gatekeeper-routed call signaling EP GK GK EP ARJ with a cause code of routeCallToGatekeeper A Facility with a reason indicating the call be rerouted

52 Internet Telephony 4-52

53 Internet Telephony 4-53 Optional called- endpoint signaling

54 Internet Telephony 4-54 H.245 Control Signaling To establish and control media streams Agree on the media formats and bandwidth Multiplexing multiple media streams No actual media A generic protocol for the control of media streams How it works in an H.323?

55 Internet Telephony 4-55 H.245 Message Groupings Requests Require the recipient to perform some action and response Response In reply to Requests Commands Require the recipient to perform some action, but no response is necessary Indications Of an informational nature only

56 Internet Telephony 4-56 The Concept of Logical Channels A Logical channel A unidirectional media path An IP address and port number Has a number A two-party conversation Two logical channels exist Potentially in different formats Or a bidirectional channel consists of two logical channels An endpoint issues Open Logical Channel Logical channel number and media information (RTP payload type) Far endpoint responds with Open Logical Channel Ack An RTP port Messages over H.245 Control Channel (channel number 0) Permanently open

57 Internet Telephony 4-57 H.245 Procedures Capabilities Exchange Share information regarding receive and transmit capabilities Indicate a preference TerminalCabilitySet message A request message A sequence number plus the types of audio and video formats TerminalCapabilitySetAck Empty with a sequence number TerminalCapabilitySetReject With a reason for rejection

58 Internet Telephony 4-58 TerminalCapabilitySetRelease If no response within a timeout period SendTerminalCapabilitySet Request terminal Capability information Request to indicate all of its capabilities Or request confirmation about specific capabilities A command message that is replied

59 Internet Telephony 4-59 Master-Slave Determination One of the endpoints needs to be the master Of particular importance for the setup of a multi-party conference Compare two pieces of information at each entity A terminal type value A terminal without an MC: 50 A gateway without an MC: 60 An MCU for audio, vedio: 190 An MCU managing a conference: 240 (the highest) A random number (1..16,777,215) Master-Slave Determination message Master-Slave Determination Ack A “ master ” or “ slave ” indication

60 Internet Telephony 4-60 Establishing and Releasing Media Streams For media exchange

61 Internet Telephony 4-61 Open Unidirectional Logical Channel

62 Internet Telephony 4-62 Open Bidirectional Logicl Channels

63 Internet Telephony 4-63 Closing Logic Channels and Ending a session CloseLogicalChannel, CloseLogicalChannelAck Only the initiator can issue Or the receiving end can humbly request A bidirectional channel can be closed by either end Once all logical channels are closed EndSession, EndSession commands

64 Internet Telephony 4-64 Close a Logical Channel

65 Internet Telephony 4-65 A Slow Start Plus Capability exchange Master-slave determination

66 Internet Telephony 4-66 Fast-connect Procedure The Setup A faststart element OpenLogicalChannel requests No H.245 control channel

67 Internet Telephony 4-67 H.245 Message Encapsulation is encapsulated with Q.931 messages Set the element h245Tunneling true in the first Q.931 message The h245Control element contains the encapsulated data Conflict with the faststart element Use Facility when no Q.931 message can be used to start a separate H.245 control channel

68 Internet Telephony 4-68 Conference Calls MC manage multi-point conference A Pre-Arranged Conference Establish a call with the MCU The MCU specifies the conference mode Communication Mode command (H.245) Specify all the sessions in the conference The transmit requirements of each endpoint, not the receive requirements

69 Internet Telephony 4-69 An Ad-Hoc Conference Expand an existing two-party call One the the endpoints must contain an MC

70 Internet Telephony 4-70 New Features in H.323 Version 2 H.235 - security and authentication, i.e. passwords for registration with gatekeeper H.450.x - supplementary services such as call transfer and forwarding Fast call setup: Bypasses some setup messages Triggered by Q.931 Fast Start message that contains basic capabilities

71 Internet Telephony 4-71 New Features in H.323 Version 2 (cont.) Mechanism to specify alternative gatekeepers to endpoints Gatekeeper can request forwarding of Q.931 information on direct routed calls; only RADCOM can play back H.323 streams off a network: a true differentiation Smoother integration of T.120 (optional standard for data) T.120 channel opened like any H.323 channel

72 Internet Telephony 4-72 T.120 Data Channel Defined in T.120 A Channel Between Endpoints A Reliable Channel Functions Transport Data Multipoint Delivery Internetworking PSTN, ISDN LAN (TCP/IP, IPX)

73 Internet Telephony 4-73 The Future of H.323 Inter-Gatekeeper Communication: Current H.323 standards do not provide an inter-zone model that scales well for large networks Inter-gatekeeper protocols being discussed to enable gatekeepers to efficiently locate one another to route calls to non-local address Hierarchical arrangements with “ clearing house ” gatekeepers have been proposed This is critical for widespread interoperability between VoIP service providers

74 Internet Telephony 4-74 H.323 Version 3 Modest improvements to H.323 version 2 Maintaining and Reusing Connections Conference out of Consultation Caller ID Language Preference Usage of Annex E/H.323 Remote Device Control Generic Capabilities Annex G/H.225.0 - Communication between Administrative Domains

75 Internet Telephony 4-75 Annex E/H.323 - Protocol for Multiplexed Call Signaling Transport Annex F/H.323 - Simple Endpoint Type H.341 - H.323 Series MIB Supplementary Services

76 Internet Telephony 4-76 H.323 Version 4 Many new enhancements Gateway Decomposition Multiplexed Stream Transmission Supplementary Services Annex K/H.323 Annex L/H.323 H.450.8 - Name Identification Service H.450.9 - Call Completion H.450.10 - Call Offer H.450.11 - Call Intrusion

77 Internet Telephony 4-77 Additive Registrations Alternate Gatekeepers Usage Information Reporting Endpoint Capacity Caller Identification Service Tones and Announcements Mapping Aliases Indicating Desired Protocols Bandwidth Management Reporting Call Status

78 Internet Telephony 4-78 Enhancements to Annex D (Real-Time Fax) Call Linkage Tunneling QoS H.245 in Parallel with Fast Connect Generic Extensibility Framework H.323 URL Call Credit-Related Capabilities DTMF Relay via RTP

79 Internet Telephony 4-79 The Decomposed Gateway Media gateway Media gateway controller

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