Comparative analysis of the H.323 and SIP protocols. Basic architecture of the H.323 standard

Useful tips

Recommendation H.323 Specifies multimedia communications systems that are designed to operate over packet-switched networks that do not provide guaranteed quality of service. Defines the protocols, methods, and network elements necessary to enable multimedia communications between two or more users.

An H.323 terminal is a user's terminal device for an IP telephony network, which provides two-way voice (multimedia communication) with another H.323 terminal, gateway or control device. An IP telephony gateway implements the transmission of voice traffic over networks with routing of IP packets using the H. 323. The main purpose of the gateway is to convert voice information coming from the PSTN into a form suitable for transmission over networks with IP packet routing. Basic network devices based on H.323 recommendation

Gatekeeper – performs the function of managing an IP telephony network zone, which includes terminals, gateways and conference control devices registered with this gatekeeper. Conference control unit (MCU) - used to organize and maintain conferences of any kind Basic network devices based on the H.323 recommendation

Gatekeeper functions Conversion of alias address (subscriber name, telephone number, address Email etc.) to the transport address of networks with IP packet routing (IP address and TCP port number). Controlling access of system users to IP telephony services using RAS signaling. Control, management and redundancy bandwidth networks. Routing signaling messages between terminals located in the same zone Gatekeeper - network administrator

Conference Control Unit (MCU) Multipoint Controller (MC) – required element Processor for processing user information during multipoint connections - Multipoint Processor (MP) - there can be several

Multipoint Controller Used to organize a conference of any kind. Organizes information between conference participants about the functionality of their terminals, indicates in what mode conference participants can transmit information, and this mode can change during the conference, and can also be common for all participants or separate for Each of them There may be several MCs in the network, therefore, for each newly created conference, a master / slave determination procedure is performed to determine which MC will control the conference

Protocol family H.323 protocol for interaction between terminal equipment and the gatekeeper - RAS, works via the UDP protocol connection control protocol - H.225, works via the TCP protocol logical channel control protocol - H.245, works via the TCP protocol

RAS protocol Determining the location of equipment Changing the bandwidth during a call Polling and indicating the current state of the equipment Notifying the gatekeeper about the release of bandwidth previously occupied by the equipment Gatekeeper detection Registration of terminal equipment with the gatekeeper Access control network resources

Gatekeeper discovery manual method of gatekeeper discovery automatic method of gatekeeper discovery Automatic method of gatekeeper discovery GRQ - Gatekeeper Request GCF - Gatekeeper Confirmation GRJ - Gatekeeper Reject UDP port 1719(1718) (Gatekeeper UDP Discovery MulticastAddress) rasAdderess

Registration and unregistration process RCF - Registration Confirmation RRJ - Registration Reject URQ - Unregister Request UCF - Unregister Confirmation URJ - Unregister Reject Gatekeeper UDP Registration and Status Port 1719 RRQ - Registration Request timeToLive keepAlive

H.225 Discriminator message format—distinguishes call control messages from other messages Length of call reference bits—length of the call reference parameter Call reference value—value of the call reference parameter Message type—message type Information elements—user information

H.245 control channel Definition of master and slave devices Exchange of functionality data Open and close unidirectional logical channels Open and close bidirectional logical channels Determine the delay that occurs when transmitting information from source to destination and in the opposite direction Select information processing mode Loopback signaling created for the purposes of equipment maintenance. The transfer of H.245 control information is carried out by the TCP protocol over a zero logical channel, which must be constantly open from the moment the H.245 channel is established until its elimination

Determination of master and slave equipment First option Second option terminalType statusDeterminationNumber

Logical channels Information transmitted by a source to one or more receivers in networks based on the H.323 recommendation? It is carried over logical channels, which are identified by a channel number unique for each direction of transmission. 2 types of logical channels: Unidirectional - opening in the direction from the source to the receiver Bidirectional - from the source of information to the receiver and back

Control Message Tunneling H.245 control messages are transmitted over a signaling channel rather than over a separate control channel. To apply H.245 message encapsulation, the calling equipment must set the h245Tunneling element carried in the Setup message and subsequent Q.931 messages to true. A similar procedure must be performed in the opposite direction. If the equipment does not support H.245 tunneling, then a separate channel is opened for the transmission of control messages.

Establishing a connection between terminals H T1 sends an ARQ message to the zone controller over the RAS channel and requests permission to use a forward signaling channel with T. The zone controller satisfies T1's request with an ACF message. 3. T1 sends a Q.931 “setup” message to terminal T2. 4. T2 responds with a Q.931 “call proceeding” message. 5. T2 registers with the zone controller by sending it an ARQ message over the RAS channel. 6. The zone controller confirms registration with an ACF RAS message. 7. T2 notifies T1 of its registration (and therefore permission to establish a connection) with a Q.931 “alerting” message. 8. After establishing a connection, T2 informs T1 about the completion of the Q.931 procedure with the “connect” message.

Establishing a connection between H.323 terminals (2) 1. T1 sends a “TerminalCapabilitySet” message to T2. 2. T2 confirms the start of the capability negotiation session with a “TerminalCapabilitySetAck” message. 3. T2 informs terminal T1 about its parameters with the message “TerminalCapabilitySet”. 4. T1 completes the capability negotiation process with a “TerminalCapabilitySetAck” message. 5. T1 opens a channel for transmitting multimedia information in the direction of T2 with the “openLogicalChannel” message (it includes the transport address of the RTCP channel). 6. T2 acknowledges the opening of a logical bearer channel from T1 with an “openLogicalChannelAck” message (this also includes the RTP address of the T2 terminal and the RTCP address received from T1). 7. T2 opens a multimedia channel in the direction of T1, informing about this with the “openLogicalChannel” message (it contains an RTCP address). 8. T1 acknowledges the establishment of a logical bearer channel from T2 with an “openLogicalChannelAck” message (this includes the RTP address of the T1 terminal and the RTCP address received from T2). This completes the process of establishing a bidirectional connection.

Establishing a connection between H.323 terminals (3) 1. T2 initiates a release by sending an H.245 “EndSessionCommand” message. 2. T1 ends the communication and confirms the disconnect with an “EndSessionCommand” message. 3. T2 closes the connection after sending the Q931 “release complete” message. 4. T1 and T2 initiate their disconnection from the zone controller with RAS DRQ messages. 5. The zone controller disables T1 and T2, having previously notified them of this with DCF messages. Fast Connect procedure The calling equipment sends a Setup message with a fastStart element fastStart includes one or more OpenLogicalChannel structures One of the OpenLogicalChannel structures must necessarily contain a forwardLogicalChannelParameters element (one unidirectional channel) and may contain reversLogicalChannelParameters (a channel in the reverse direction) fastStart may contain several alternative OpenLogicalChannel structures that differ in coding algorithms transmitted information or decoding received information

Fast Connect procedure The called equipment may reject the Fast Connect procedure if: It does not support it There is a need to use H.245 procedures by opening a separate H.245 channel or tunneling control messages The called equipment may begin transmitting messages immediately following any Q message. 931 with fastStart element The calling equipment that initiated the Fast Connect procedure may begin transmitting voice information immediately upon receiving any of the enabled Q.931 messages containing the fastStart element

Ministry of Education Russian Federation

MOSCOW STATE INSTITUTE

ELECTRONICS AND MATHEMATICS (TECHNICAL UNIVERSITY)

Abstract on the subject

Computer network management

“Internet telephony. H.323 protocol"

Checked by Kharlamov A.G.

Performer Group S-94

Murchie A.E.

Moscow 2010

Introduction

In just a few years, IP telephony technologies have evolved significantly, and the solutions common today are significantly different from previous ones. On the one hand, this is due to the development of hardware solutions, in particular the emergence of powerful backbone and transit routers and high-speed telecommunication channels. On the other hand, one cannot fail to note the emergence of such qualitatively new technologies as dynamic routing taking into account the quality of service in multiservice IP networks and resource reservation to control the quality of service of transit routers.

Modern equipment for voice transmission over the Internet Protocol (VoIP) makes it possible to ensure the priority of voice traffic transmission over the transmission of regular data, obtain acceptable quality of the audio signal with strong compression, and effectively suppress various noises.

Today, telecommunications operators specializing in the provision of IP telephony services use dedicated channels with priority for voice traffic over data traffic, which guarantees high quality voice transmission. In this case, several options for routing voice traffic are used for each of thousands of directions, and if any problems arise, the traffic is automatically redirected to other channels.

As it develops, IP telephony undergoes important qualitative changes: from additional service it is gradually turning into a kind of basic service, which may soon become one of the components of multi-service technology.

The protocol for transmitting voice traffic plays an important role. Firstly, H.323, which originates from traditional telephone protocols, and, secondly, protocols created on the basis of IP technologies, such as SIP, MGCP, MEGACO, are actively developing.

Russian IP telephony operators most often use protocols of the H.323 group. This is because this protocol was the first generally accepted standard for the industrial implementation of IP telephony. Currently, more and more attention is being paid to SIP. The SIP protocol in this group is the simplest type of protocol, more accessible to perception and understanding by the average IT specialist. SIP is especially good for use in intranets. At the same time, the external protocol in the network of a telecommunications operator for an enterprise, as a rule, will still remain either H.323 or MGCP/MEGACO.

As noted, IP telephony is becoming one of the components of the solution for transmitting heterogeneous multimedia traffic using the TCP/IP protocol. And it is quite natural that the development of individual multimedia traffic management tools affects the entire system of packet data transmission technologies.

It should also be kept in mind that IP telephony is not just an alternative to regular telephony. The relevance of the development of IP telephony solutions is due not only to the possibility of reducing costs for telephone conversations And Maintenance infrastructure (although this certainly matters). In strategic terms, IP telephony can become unified technical platform, which will combine solutions for data and voice transmission, as well as for processing and subsequent use of this information in all business processes. Thus, the development of IP telephony in a certain sense is a means of increasing labor productivity and business development.

Protocol H .323

In 1990, the first international standard in the field of video conferencing, the H.320 specification, was approved to support video conferencing over ISDN. Then the ITU-T approved a whole series of recommendations related to video conferencing. This series of recommendations, often referred to as H.32x, in addition to H.320, includes the H.321-H.324 standards, which are intended to various types networks. In the second half of the 90s, IP networks and the Internet received intensive development. They have evolved into a cost-effective data transmission medium and have become almost ubiquitous. However, unlike ISDN, IP networks are poorly suited for transmitting audio and video data. The desire to use the established structure of IP networks led to the appearance in 1996 of the H.323 standard, which contains descriptions of terminal devices, equipment and network services designed to implement multimedia communications in packet-switched networks (for example, Intranet or Internet). H.323 terminal devices and network equipment can transmit data, voice and video information in real time. The H.323 recommendation does not define: network interface, the physical medium for transmitting information and the transport protocol used in the network. The network over which communication between H.323 terminals occurs can be a segment or multiple segments with a complex topology. H.323 terminals can be integrated into personal computers or implemented as standalone devices. But support for voice exchange is a mandatory feature for any H.323 device.

· bandwidth management;

· Possibility of network interaction;

· platform independence;

· support for multipoint conferences;

· support for multicast transmission;

· standards for codecs;

· support for multicast addressing.

Bandwidth Management

The transmission of audio and video information very intensively loads communication channels, and if this load growth is not monitored, the performance of critical network services may be disrupted. Therefore, the H.323 recommendations provide for bandwidth management. You can limit both the number of simultaneous connections and the total bandwidth for all H.323 applications. These restrictions help preserve the necessary resources for running other network applications. Each H.323 terminal can manage its own bandwidth in a particular conference session.

Internet conferences

Platform independence

H.323 is not tied to any hardware or software technology solutions. Applications that interact with each other can be created based on different platforms, with different operating systems.

Multipoint conference support

The H.323 recommendations allow for a conference with three or more participants. Multipoint conferences can be held either with or without a central controller - MCU (multipoint conference unit).

Multicast support

H.323 supports multicast in a multipoint conference if the network supports the multicast control protocol. With multicast transmission, one packet of information is sent to all necessary recipients without unnecessary duplication. Multicast uses bandwidth much more efficiently because exactly one stream is sent to all mailing list recipients.

Codec Standards

H.323 sets standards for encoding and decoding audio and video streams to ensure equipment compatibility different manufacturers. At the same time, the standard is quite flexible. Requirements are formulated, the fulfillment of which is mandatory, and there are optional features, if used, it is also necessary to strictly follow the standard. In addition, the manufacturer may include additional features in multimedia products and applications if they do not contradict the mandatory and optional requirements of the standard.

Compatibility

There may be cases where conference participants want to communicate with each other without worrying about compatibility issues among themselves. H.323 recommendations support elucidation general opportunities end user equipment and establishes the best possible encoding, calling and control protocols common to conference participants.

Flexibility

An H.323 conference can include participants whose end equipment has different capabilities. For example, one of the participants may use a terminal with only audio capabilities, while the rest of the conference participants may also have the ability to transmit/receive video and data.

H.323 architecture

· terminal;

· zone controller;

· gateway;

· Multipoint Conference Control Unit (MCU).

Rice. 1. Structural scheme IP telephony networks according to the H.323 standard

Terminal ( Terminal) - a terminal multimedia (voice, video, data) device intended for participation in a conference. By terminal, the standard refers to network endpoint equipment that allows users to communicate with each other in real time. The H.323 terminal must support the following protocols:

1. H.245 to establish terminal capabilities and create an audio information exchange channel.

2. H.225 for call signaling and communication parameters setting.

3. RAS for registering the user terminal and setting additional parameters for managing the zone controller.

4. RTP/RTCP for ordering audio and video packets.

The H.323 terminal must also support an audio codec in accordance with G.711.

The H.225 and RAS protocols are used between H.323 endpoints (terminals and gateways) and the area controller to provide:

· zone controller detection (GRQ);

endpoint registration;

Determining the location of the end point;

· authentication management;

· setting an access token.

RAS messages are sent over unreliable RAS channels, so message exchange is subject to loss, delay, and retransmissions.

H.323 protocol stack

The H.323 standard defines broad requirements for many different protocols that make up the complete H.323 protocol stack.

The H.323 stack consists of 7 protocol groups:

1. control and alarm;

2. processing of audio signals;

3. video signal processing;

4. conference call;

5. transmission of multimedia information;

6. provision information security;

7. additional services;

1. Connection control and signaling:

· 1.a. H.225.0: Media stream signaling and packetization protocols (uses a subset of the Q.931 signaling protocol).

· 1.b. H.225.0/RAS: Registration, admission and status procedures.

· 1.v. H.245: Control protocol for multimedia.

2. Processing of audio signals:

· 2.a. G.711: pulse code modulation of voice frequencies.

· 2.b. G.722: 7 kHz audio encoding at 64 kbps.

· 2.v. G.723.1: dual-rate speech encoders for multimedia communications at 5.3 and 6.3 kbit/s.

· 2.g. G.728: Linear prediction coding of 16 kbit/s speech signals with low-latency excitation signal coding.

· 2.d. G.729: Linear prediction coding of 8 kbit/s speech signals with algebraic coding of the conjugate structure excitation signal.

3. Video signal processing:

· 3.a. H.261: Video codecs for audiovisual services at 64 kbps.

· 3.b. H.263: Video encoding for low bit rate transmission.

4. Data conference call:

· 4.a. T.120: This is a protocol stack (which includes T.123, T.124, T.125) for transferring data between endpoints. It can be used for various Collaboration Work applications such as collaborative raster image editing, sharing applications and collaborative document organization. T.120 uses a layered architecture similar to the OSI model.

5. Multimedia transmission:

· 5.a. RTP: Real Time Transport Protocol.

· 5 B. RTCP: Real Time Transmission Control Protocol.

6. Security:

· 6.a. H.235: Security and encryption for multimedia terminals on the H.323 network.

7. Additional services:

· 7.a. H.450.1: Generic functions for managing supplementary services in H.323.

· 7.b. H.450.2: transfer of the connection to the telephone number of the third subscriber.

· 7.c. H.450.3: Call forwarding.

· 7.g. H.450.4: Call hold.

· 7.d. H.450.5: Call Park ( park) and answer the call ( pick up).

· 7.e. H.450.6: Incoming call notification in conversation state.

· 7.g. H.450.7: message waiting indication.

· 7.z. H.450.8: Name Identification Service.

· 7.i. H.450.9: Call termination service for H.323 networks.

Establishing a connection via H.323

Zone Controller Discovery (GRQ)

The zone controller discovery process is used by H.323 endpoints with which the endpoint must register. Zone controller discovery can be done statically or dynamically. In static mode, the endpoint knows the controller's transport address a priori. In dynamic controller discovery mode, the endpoint sends a controller lookup multicast message (GRQ) to the controller lookup multicast address containing the question "Who is my controller?" One or more controllers may respond with a GCF message: "I can be your controller."

Endpoint registration

Registration is the process used by endpoints to connect a zone and tell the controller the parameters of the zone's bearer network that provides the transport, and one of its address aliases. All endpoints register with the zone controller.

Determining the position of the endpoint

Determining the location of an endpoint is the process of associating its network address (address on the transport network) with its H.323 alias or E.164 address (telephone number).

Other control functions

The RAS channel is also used for other types of control mechanisms, such as authentication control, limiting endpoint entry into a zone, bandwidth control, and control of disconnection (disconnection) processes when an endpoint disconnects from the current zone controller and exits the zone.

Standards H.225 - call signaling and H.245 - control signaling

H.225 - call signaling

H.225 - call signaling - is used to establish a connection between H.323 endpoints (terminals and gateways) through which real-time data will be transported. Call signaling involves the exchange of H.225 protocol messages over a reliable channel enabled for this purpose (call signaling channel).

If the H.323 network does not have a zone controller, the endpoints exchange call signals directly with each other. If there is a zone controller, then two calling methods can be used: signaling directly between endpoints (the so-called "direct calling method") and signaling between endpoints only after contacting the zone controller and routing the call ("method with call routing in the zone controller" "). The method used is selected when the endpoint registers with the zone controller.

Method with routing "calls in the zone controller"

Call signaling between endpoints and the zone controller is carried over RAS channels. The zone controller receives the call message through the signaling channel from one endpoint and forwards it to the other endpoint through the signaling channel of the other endpoint.

H.245 - control signaling

H.245 control signaling consists of the end-to-end exchange of H.245 messages between H.323 endpoints. H.245 control messages are transmitted over H.245 control channels. H.245 - control channel is a logical channel that is constantly open, unlike multimedia stream exchange channels. Control signaling messages can be divided into two groups: H.323 terminals exchanging their parameters and control messages.

· Parameter exchange messages

Parameter exchange allows terminals to select the communication modes and encoding formats that they can use when working with each other. The capabilities of the terminals, both for reception and transmission, are being clarified.

Process control messages are logical channels between endpoints

A logical channel carries information from one endpoint to another endpoint (in the case of a point-to-point conference) or multiple endpoints (in the case of a multipoint conference). The H.245 protocol provides a set of messages that enable the opening and closing of these channels. The logical channel is always unidirectional.

Multimedia gateway (Gateway)

It is a device designed to convert multimedia and control information when connecting heterogeneous networks (Fig. 2).

Rice. 2. H.323/PSTN Gateway

The gateway is not a required component of an H.323 network. It is only necessary when it is necessary to establish a connection with a terminal of a different standard. This communication is ensured by the translation of protocols for establishing and terminating connections, as well as data transfer formats. According to H.323, a multimedia gateway is an optional element in an H.323 conference. He can do a lot various functions. Its typical function, for example, is the task of converting transmission protocol formats (for example, H.225.0 and H.221). H.323 gateways are widely used in IP telephony to interconnect IP networks and digital or analog switched telephone networks (ISDN or PSTN). If there is no gateway in the network, one of its functions must be implemented - converting the PSTN number into the transport address of the IP network using other means. From the side of networks with IP packet routing, as well as from the PSTN side, the gateway can participate in connections as a terminal or conference control device.

Multipoint Control Unit (MCU) designed for organizing conferences with three or more participants. This device must contain a Multipoint Controller (MC) and possibly Multipoint Processors (MP). The MC controller supports the H.245 protocol and is designed to coordinate the processing parameters of audio and video streams between terminals. Processors are responsible for switching, mixing and processing these streams.

Multipoint conference configurations can be centralized, decentralized, hybrid, or mixed.

Rice. 3. Schemes of centralized and decentralized conference organizations in H.323

Centralized multipoint conference requires an MCU device. Each terminal exchanges audio, video, data and control commands with the MCU in a point-to-point manner. The MCU controller, using the H.245 protocol, determines the capabilities of each terminal. The MP processor generates the multimedia streams necessary for each terminal and sends them out. In addition, the processor can provide conversions of streams from various codecs with different speeds data.

Decentralized multipoint conference uses multicast technology. H.323 terminals participating in the conference multicast the multimedia stream to other participants without sending it to the MCU. The transmission of control and control information is carried out in a point-to-point manner between the terminals and the MCU. In this case, multipoint control is performed by the MCU controller.

The hybrid conferencing scheme is a combination of the previous two. The H.323 terminals participating in the conference multicast the audio-only or video-only stream to the remaining participants without sending it to the MCU. The transmission of other streams is carried out using a point-to-point scheme between the terminals and the MCU. In this case, both the controller and the MCU processor are involved.

Rice. 4. Decentralized and mixed conference organization schemes in H.323

In a mixed conferencing scheme, one group of terminals can work according to a centralized scheme, and another group - according to a decentralized one.

Zone Controller (or Gatekeeper)- recommended, but not mandatory device, providing network management and acting as a virtual telephone exchange.

The zone controller provides call control services for H.323 endpoints such as address translation and bandwidth management according to the RAS protocol. The zone controller in an H.323 network is not a required component. However, if it is present in the network, then terminals and gateways must use its services. The H.323 standard defines both mandatory zone controller services and additional (optional) ones. functionality which it can provide.

An optional feature of the zone controller is call routing. Endpoints send ringing messages to the zone controller, which routes them to destination endpoints. Alternately, endpoints can send call signaling messages directly to each other. This capability is valuable for ongoing case monitoring and case management on the network. Routing calls through a zone controller provides better network efficiency because the controller can make routing decisions based on a number of factors, such as load balancing among gateways.

The services offered by the zone controller are defined in the RAS and include address translation, receive control, bandwidth control and zone control. H.323 networks that do not have a gateway controller do not have these capabilities. H.323 networks containing IP phones and gateways must contain a zone controller to translate incoming E.164 phone addresses into transport addresses. The zone controller is a logical component of H.323, but it can also be implemented as part of a gateway.

Required Features zone controller

· Address broadcast

A call originated within an H.323 network can be used to address the desired terminal using its alias (short name). A call originating outside the H.323 network and received through a gateway to be addressed to the recipient terminal may use a telephone number in accordance with the E.164 recommendation (for example, 310-442-9222). This recommendation is used to address ISDN subscribers. The zone controller converts the received E.164 phone number or alias into network address(For example, 204.252.32.156 for an IP network) of the recipient terminal. The destination endpoint can be reached using this network address.

· Registration management

The zone controller can manage the registration of endpoints in the H.323 network. In this case, RAS messages are used: registration request ( ARQ), confirmation ( ACF) and deviation ( ARJ). Registration control can be a dummy function that admits all endpoints to the H.323 network.

· Bandwidth management

The controller provides bandwidth management using RAS messages: Bandwidth Request ( BRQ), confirmation (BCF) and rejection ( BRJ). For example, if network manager has defined a threshold for the number of simultaneous connections for an H.323 network, the zone controller may refuse to establish new connections unless this threshold is reached. As a result, it is possible to limit the total allocated bandwidth to some portion of the total bandwidth of the data network, leaving the remaining bandwidth for data applications. Bandwidth management can also be a dummy function that simply receives requests without processing them.

Optional zone controller features

· Call management

The zone controller can route calls between H.323 endpoints. In a point-to-point conference, the area controller can handle H.225 ringing messages. Alternatively, the zone controller may allow endpoints to independently exchange H.225 call signaling messages directly with each other.

When the endpoint sends call messages to the area controller, it can accept or reject the call, in accordance with the H.225 standard. Reasons for rejection may be access or time restrictions set for specific terminals or gateways.

Call control

The zone controller can monitor data for all active H.323 connections, allowing zone management to provide bandwidth control and network load balancing by routing calls between terminals and gateways.

H.323 connection procedure

Let's take a step-by-step look at the scenario of establishing a connection between two H.323 terminals without using a zone controller (Fig. 5).

1. Final destination A(caller) connects to endpoint B(called party) and sends a message Setup(setting as defined in H.225.0) including the call type (e.g. audio only), called number and calling party and address.

Rice. 5. H.323 connection setup scenario

2. Final destination B responds with a notification message ( Alerting). Final destination A must receive this message before the installation time expires.

3. When the user is at the endpoint B answers a call (picks up the phone), message Connect(connection) is sent to the endpoint A .

4. Both terminals transmit information about their capabilities (media types, codec selection, and multiplexing information) in the message TerminalCapabilitySet(setting terminal capabilities).

5. Each terminal responds with a message TerminalCapabilitySetAck(confirmation of installation of terminal capabilities). If the remote endpoint does not have any capabilities, a message will be sent TerminalCapabilitySetReject(terminal capability setting rejection), and the terminals will continue to send these messages until they determine that the capabilities being set are supported by both endpoints.

6. Each terminal transmits an H.245 message OpenLogicalChannel(open logical channel) allows you to open a logical channel with a remote endpoint to configure the voice channels over which media streams will be exchanged.

7. If ready to receive data, each terminal transmits OpenLogicalChannelAck(logical channel open acknowledgment) to the remote endpoint, specifying the port number on which the remote endpoint should send RTP data, and the port number on which RTCP data should be sent to the remote endpoint.

8. Endpoints exchange information in RTP packets. During this exchange, RTCP packets are transmitted to monitor the quality of data transmission.

9. When the final destination A hangs up (hangs up), he must send an H.245 message CloseLogicalChannel(close logical channel) for each channel open with the endpoint B .

10. Final destination B replies with a message CloseLogicalChannelAck(confirmation of closing a logical channel).

11. Final destination A sends an H.245 command EndSessionCommand(session end command) and closes the channel after receiving the same message from the endpoint B .

12. Both terminals send H.225.0 message ReleaseComplete(release complete) on the call signaling channel, which closes the channel and ends the call.

Characteristics of IP telephony gateways

In general, IP telephony relies on two main operations: converting bidirectional analog speech into digital form within an encoder/decoder (codec) and packaging it into packets for transmission over an IP network. These functions are most often performed by autonomous gateways, which come in several varieties. These can be dedicated devices or combined routers/switches with built-in gateway hardware and software. Another type is when the gateway is combined with remote access equipment and a pool of modems.

Regardless of the hardware implementation method, IP telephony gateways must have a number of necessary properties.

· Compatible with H.323 standard.

Basic protocol for the operation of IP equipment, the vast majority of manufacturers have adopted the protocol described by the ITU-T in recommendation H.323v2, which standardizes multimedia communications in packet-switched networks

Users of multimedia personal computers with H.323 software can connect to such a gateway system. Calls can be routed to gateways from other manufacturers that support H.323. As a result, the system will provide real-time integration of speech, video and data (such as Microsoft NetMeeting).

Rice. 6. Position of the gateway in the IP telephony network

· Availability of resource reservation mechanisms.

Support for any prioritization scheme (RSVP reservation protocol or service differentiation byte - DS byte) to enable the choice of priority between transmitted voice or data is an important characteristic of the gateway. At the same time, the RSVP protocol allows routers to reserve part of the bandwidth for organizing voice traffic.

· Support for basic telephone interfaces and alarm types.

An important criterion when assessing the characteristics of gateways is the greatest possible variety of telephone interfaces supported by the IP gateway (E1, PRI, BRI), and analog in particular, as well as support for the main types of telephone signaling: CAS, DTMF, PRI and SS No. 7. Essential role The equipment supports security mechanisms in accordance with the mentioned recommendation H.235.

· Transport architectures.

The range of transport architectures with which modern gateways work is quite wide: leased lines, ISDN, Frame Relay, ATM, Ethernet.

· Scalability.

An important characteristic of the gateway is its scalability, which is ensured modular construction equipment. At the first stage of deployment of an IP telephony network, it is possible to use an incomplete resource of available ports with a gradual further increase in the number of involved voice ports. In this case, the number of ports corresponds to the number of simultaneous calls that the gateway can make, since each of its ports is equipped with its own digital signal processor (DSP - Digital Signal Processor) for digitizing voice signals.

· Providing fax communications.

The vast majority of gateways manufactured have the ability to provide IP-based fax communications. It is based on two main standards proposed by ITU-T. The T.37 standard reduces fax transmission to store-and-forward delivery, since fax images are transmitted as email attachments. With T.37, fax machines and fax servers can communicate with each other as seamlessly as traditional fax machines. Another T.38 standard describes the transmission of faxes in real time, either by simulating a connection to a fax machine or using a modulation method called FaxRelay. T.38 can be used to provide functionality more similar to traditional fax, such as immediate confirmation.

· Gateway management.

Gateways may differ in the controls they provide. These controls have the function of routing calls between gateways and transcoding telephone numbers to IP addresses. They can be structurally integrated with the gateway or represent a separate “multimedia conference manager” or “multi-voice access manager”. One solution is to use a single package that includes billing, call routing and network administration tools.

Possibility of installation various algorithms speech coding.

The quality of voice transmitted over an IP network is significantly affected by the coding scheme used in the VoIP gateway when compressing voice information. The most common is the scheme that provides the highest degree of information compression and complies with the G.723.1 specification (up to 5.3 kbit/s). Other schemes are also used - G.729a, G.711, G.726, G.728. In this case, it is extremely important to equip the gateway with an additional installation of the voice compression scheme used.

Classification of IP telephony gateways

Based on the scale of application, they can be divided into two main types: gateways aimed at corporate use, and gateways designed for operators and communication service providers. The latter type of products are different large capacity and scalability, the presence of authentication and monitoring tools, as well as additional billing capabilities.

According to their design, gateways can be:

· Autonomous.

Most gateway manufacturers offer standalone IP gateways, which typically consist of PC-based servers with a set of voice cards. Voice cards are not designed for audio compression/decompression, so this operation must be executed by the PC's main processor.

· Gateway routers.

In the world of telecommunications equipment manufacturers, there is a tendency for large companies to equip traditional network equipment with nodes responsible for IP telephony. These products - routers and devices for accessing distributed networks with built-in IP telephony gateways - occupy a separate important niche in the network equipment market.

· RAS gateways.

VoIP gateways, which consist of cards installed in remote access servers (RAS), occupy their part of the IP telephony equipment market. Device installation of this type when building IP networks, it is justified when working with applications with many voice ports.

· Gateway modules for UPBX.

Currently, IP telephony gateways have become widespread, which are structurally modules for classic office PBXs. Such a system checks the quality of the connection before establishing a connection via an IP network. If its quality is sufficient (the standard is set by the system administrator), the connection is established. If this is not the case, the call is routed over traditional lines. Thus, there is a desire among manufacturing companies to gradually replace the transport environment without affecting the telephone service provided to end users.

· Gateways with business application integration.

As IP telephony systems develop, service functions take on leading roles. At the same time, the equipment should focus not only on traffic integration, but also on the integration of business applications, which makes it possible to increase the productivity of enterprises. It allows you to implement a click-to-talk service, for example to establish telephone communication between visitors to a company's Web site and its employees.

· Institutional PBX based on gateways.

Another direction in the development of IP telephony equipment is the construction of institutional telephone systems based on LAN infrastructures. In cases where it is not practical to install a separate server to convert telephone signals into IP packets, network devices are used that are connected directly to the 10BaseT network (type Ethernet hubs). Moreover, each hub is, in fact, a small PBX with voicemail and an automatic secretary, connected via an RJ-14 connector to external and internal telephone lines and via RJ-45 connectors to the local Ethernet networks. With ease of management and built-in computer-telephone integration, these systems are able to compete with conventional private telephone exchanges.

· Network cards with telephony functions.

One of the IP telephony solutions is multi-purpose network cards with telephony functions. Such devices are equipped with RJ-11 ports for connecting a regular telephone.

· Standalone IP phones.

They provide an all-in-one solution for one line. In appearance and basic service capabilities, hardware implementations of IP phones are no different from regular phones, but their electronic “stuffing” can significantly reduce the load on personnel responsible for telephone communications.

In addition to hardware, there are also software implementations of IP phones. In this case, a personal computer (PC), equipped with a telephone headset or microphone and speaker systems, turns into a multifunctional communication center. PC user, in addition to accessing the usual telephone service, receives a set of additional capabilities: obtaining information about the calling client (thanks to the presence of a standard TAPI interface to other programs), monitoring telephone calls and working with voice mail. The disadvantages of such systems are incomplete compatibility with H.323 version 2, as well as the lack of support for security functions when working with a gatekeeper.

Pros and cons of H.323

Advantages

The H.323 standard is comprehensive and flexible. It can be used when developing solutions for audio or for complete networks conference calls for transmitting video/audio/data signals. There are many benefits to implementing conferencing using H.323:

· H.323 technology provides high-quality, scalable multimedia-based conferencing. H.323 multimedia conferencing can support applications such as participatory bitmap editing, collaboration over data transmission or video conference.

· H.323 technology allows for interoperability between H.320 and H.323-based equipment from different manufacturers.

· H.323 technology takes advantage of existing investments in corporate network infrastructure.

· H.323 technology can be used to organize long-distance and international telephone connections to reduce their cost.

· H.323 technology allows for more efficient use of ISDN technology using H.320 gateways and fewer ISDN lines.

· On a corporate intranet, H.323 can provide more reliable connections and reduce support issues.

· H.323 technology also offers more sophisticated network conferencing management capabilities.

· H.323 technology does not depend on hardware and operating system.

SIP technology is somewhat similar to the Q.931 and H.225 components of H.323 technology. There are some flaws H.323 vs. SIP:

· H.323 technology takes longer to establish a connection.

· H.323 technology requires about 12 packets to establish a connection (while SIP requires about 4 packets).

· H.323 technology requires both TCP and UDP during connection setup.

· Implementing H.323 is much more complex than implementing SIP.

· Third party call control is not currently available using H.323.

Bibliography

1. http://ru.wikipedia.org/wiki/H.323

2. http://www.protocols.ru/files/Protocols/H323.pdf

3. http://www.ericsson.com/hr/etk/revija/Br_2_2005_RU/protokol.pdf

4. http://www.bytemag.ru/articles/detail.php?ID=6653

5. http://mobile.asterisk.ru/knowledgebase/H.323

6. http://www.intuit.ru/department/network/iptele/

The H.323 protocol provides the basis for transmitting data, video and audio information over IP networks, including the Internet. H.323 is recommended by the International Telecommunication Union (ITU) as a set of standards for transmitting multimedia information over local networks that do not support guaranteed quality of service (QoS). Most modern networks are of this type - examples include networks based on the TCP/IP and IPX protocols in Ethernet, Fast Ethernet and T environments. o ken Ring. Therefore, H.323 protocols are an important part of building a LAN to support multimedia applications. Such applications will include H.225.0-RAS, Q.931-H.245, RTP/RTCP and audio/video/data codecs (audio codecs G.711, G.723.1, G.728, etc. etc., video codecs H.261, H.263 with compression and decompression, as well as T.120 data codecs).

Multimedia streams are transmitted based on the RTP/RTCP protocols. RTP provides the actual transmission of media streams, while RTCP supports the transmission of data for management and control. Signaling (excluding RAS) is carried using the TCP protocol. The following protocols deal with signaling:

RAS manages registration, access, and state;
Q.931 provides for the establishment and termination of connections;
H.245 is responsible for negotiating capabilities and channel usage.

In addition to the listed protocols, H.323 uses protocols that provide support for additional functions:

H.235 provides security and authentication;

H.450.x - additional services.

Protocol Location H.323 in OSI model shown in the figure.

Protocol stack position H.323 V reference model OSI

RTP

RFC 1889

The RTP (Real-time Transport) protocol provides transport functions for applications that transmit real-time data (such as voice or video) using unicast or multicast addresses. RTP does not reserve resources or guarantee QoS for real-time service. The RTCP transmission control protocol allows monitoring of data delivery (including for large networks with multicast) and provides a minimum set of management and identification functions. The RTP and RTCP protocols are designed to operate independently of the underlying transport and network protocols. RTCP protocol supports the use of translators and level mixers RTP.

The RTP header format with a fixed structure is shown in the figure.

Bits			Octet

		Counter CSRC
	Payload type
Serial number
Timestamp
SSRC
CSRC

RTP structure

V

RTP protocol version.

P

Fill flag. P=1 indicates that the end of the packet contains one or more alignment octets that are not part of the payload.

X

Extension bit. At X=1 A header with a fixed structure is followed by an additional header of a specific format.

Counter CSRC

Shows the number of CSRC IDs following the header.

M

A marker whose interpretation is determined by the profile. Markers allow you to mark important events (for example, frame boundaries in a packet stream).

Type of content

The content type identifier specifies the format of the information portion of the RTP packet and determines how applications interpret the packet. The default static mapping of content type codes to data formats is specified by the profile. Additional codes data types can be dynamically set by other means (not related to RTP).

Serial number

This field is incremented by one for each subsequent RTP packet sent. The number can be used by the recipient to detect packet loss and restore the correct packet sequence.

Timestamp

Reflects when the first octet in an RTP data packet is sampled. The sampling time value must be taken from a variable (hours) that increases continuously and linearly over time. This value is used for synchronization and detection of delivery time fluctuations (jitter). The level of clock resolution must be sufficient to ensure the desired synchronization accuracy and corresponding accuracy in determining delivery time fluctuations (as a rule, just changing the clock readings per video frame is not enough).

SSRC

Indicates the synchronization source (the identifier is chosen randomly, taking into account that two synchronization sources in the same RTP session should not have the same SSRC identifiers).

CSRC

A list of information source identifiers containing pointers to sources of useful information included in the package.

RTCP

RFC 1889 http://www.cis.ohio-state.edu/htbin/rfc/rfc1889.html

The RTP control protocol is based on periodic transmission of control packets to all session participants using the same mechanism that is used to transmit data packets. The underlying protocol layer must support multiplexing of data packets and control packets (for example, by using different port numbers in UDP).

Bits		Octet

Version	Counter of accepted reports
Package type
Length

RTCP structure

Version

The RTP version number that is the same for RTCP packets and RTP data packets. Currently version 2 is in use.

P

Fill flag. P=1 indicates that the end of the packet contains one or more alignment octets that are not part of the payload. The last octet of the padding field contains the number of padding octets that should be ignored. Padding may be required when using some fixed block size encryption algorithms. In a multipart RTCP packet, padding may only be required for the last of the individual packets because the multipart packet is encrypted as a single unit.

Counter of accepted reports

The number of report blocks contained in the package. A zero field value is allowed.

Package type

The packet type field contains the constant 200, indicating that this packet is an RTCP SR.

Length

The length field specifies the size of the RTCP packet in 32-bit words minus 1 (including header and padding). Reducing the actual packet size makes 0 a valid length value and avoids loopback when scanning a composite RTCP packet, and counting in 32-bit words avoids checking if the size is a multiple (in octets) of 4.

RAS

H.225:

The RAS (Registration, Admission and Status) channel is used for messages used in the gateway discovery and endpoint registration processes. The latter process is used to map the endpoint addresses to the transport addresses of the signaling transport channels. Because the RAS channel does not provide guaranteed delivery, H.225.0 recommends the use of timeouts and retry counters for various messages. An endpoint or gateway that fails to respond to a request within a specified time (timeout) may use RIP (Request in Progress) messages to indicate that the request has not yet been processed. The endpoint or gateway that receives the RIP resets the timer and retry counter.

RAS messages use ASN.1 syntax.

H.225

H.225: http://www.itu.int/itudoc/itu-t/rec/h/h225-0.html

H.225 is a standard for narrowband video telephony services defined in the H.200/AV.120 recommendations. The standard deals with situations where the transmission path includes at least one packet network that is configured to provide non-guaranteed QoS (such networks also do not support protection and recovery mechanisms beyond those specified in the H.320 recommendations for terminals). H.225.0 describes the organization of voice, video, data and control information flows in packet networks to provide conversational services using H.323 equipment.

The H.225 packet structure follows the Q.931 standard and is shown in the figure.

Bits		Octet

Protocol discriminator

		3 (-4)
	Message Type
Information elements

H.225 structure

Protocol discriminator

Used to distinguish messages that control user-to-network calls from other messages.

Link size

Challenge link

Identifies a call or request to register/disable a device on the local user-network interface to which a particular message applies. The length of the link can be one or two octets.

Message Type

The type field specifies the function of the transmitted message. The following message types are used:

000	xxxxx	Message when establishing connections
	00001	ALERTING (warning)
	00010	CALL PROCEEDING
	00111	CONNECT
	01111	CONNECT KNOWLEDGE
	00011	PROGRESS	Job
	00101	SETUP
	01101	SETUP ACKNOWLEDGE (installation confirmation)
001	xxxxx	Messages when transmitting information
	00110	RESUME (resume)
	01110	RESUME ACKNOWLODGE (renewal confirmation)
	00010	RESUME REJECT (refusal to resume)
	00101	SUSPEND (temporary stop)
	01101	SUSPEND ACKNOWLODGE
	00001	SUSPEND REJECT
	00000	USER INFORMATION (user information)
010	xxxxx	Messages when connections are lost
	00101	DISCONNECT
	01101	RELEASE
	11010	RELEASE COMPLETE
	00110	RESTART (restart)
	01110	RESTART ACKNOWLEDGE (restart confirmation)
011	xxxxx	Other messages
	00000	SEGMENT
	11001	CONGESTION CONTROL
	11011	INFORMATION (information)
	01110	NOTIFY
	11101	STATUS (status)
	10101	STATUS ENQUIRY

Information Elements (IE)

The protocol defines two categories of information elements: one-octet and variable-length. The information element formats are shown in the figure.

Bits			Octet

	EI ID	IE Content

Single octet information element format (type 1)

Bits		Octet

	IE ID

Single octet information element format (type 2)

Bits		Octet

	IE ID
IE Content Length
IE Content

Variable Length Information Element Format

H.245

H.245: http://www.itu.int/itudoc/itu-t/rec/h/h245.html

H.245 defines a transmission line for non-telephone signals. The protocol includes transmission and reception characteristics, as well as the preferred mode for the receiving side, logical channel signaling, monitoring and indication. To ensure reliable delivery of data, audio and video signals, acknowledgment procedures are defined at the signaling level.

H.225 messages follow ASN.1 syntax.

Messages of type MultimediaSystemControlMessage can be defined as a request, response, command, or indication. The additional message sets listed below are also used.

Master Slave Determination (definition of the leader and slave).
Terminal Capability.
Logical Channel Signaling.
Multiplex Table signaling.
Request Multiplex Table signaling.
Request Mode.
Round Trip Delay (signal delay in both directions).
Maintenance Loop.
Communication Mode.
Conference Request and Response (request and response for the conference).
Terminal ID.

Commands and Indications (commands and indicators).

H.261

H.261: http://www.cis.ohio-state.edu/htbin/rfc/rfc2032.html

The H.261 protocol describes video streams for transmission using the Real Time Transport Protocol (RTP). At a lower level, any protocols capable of supporting RTP traffic can be used.

The header format is shown in the figure.

Bits					Octet

SBIT		EBIT
GOBN			MBAP
MBAP	QUANT			HMVD
HMVD		VMVD

H.261 header structure

SBIT

Start bit. The number of significant bits to be ignored in the first data octet.

EBIT

The final bit. The number of least significant bits to be ignored in the last data octet.

I

INTRA frame data encoding flag. If this bit is set to 1, the stream contains only blocks encoded as INTRA frames. A zero value of the flag indicates that this stream may or may not contain blocks encoded as an INTRA frame. The value of this bit cannot change during the entire RTP session.

V

Motion Vector flag. The value is set to zero when motion vectors are not used in this thread. A single flag value indicates the possibility of the presence of displacement vectors. The value of this bit cannot change during the entire RTP session.

GOBN

GOB number indicating the start of the packet. The value of this field is 0 if the packet begins with a GOB header.

MBAP

The MBAP (Macroblock Address Predictor) field encodes the macroblock address prediction (i.e., the last MBA value contained in the previous packet). The field value is in the range 0-32 (to predict acceptable values MBA is 1-33), but since the bitstream cannot be fragmented between the GOB header and MB 1, the predictor at the beginning of the packet cannot be 0. This leaves the range 1-32, which is offset by -1 to so that a 5-bit field is sufficient. If the packet begins with a GOB header, MBAP=0.

QUANT

The QUANT field shows the MQUANT or GQUANT value before the start of the packet. If the packet begins with a GOB header, the QUANT field is set to zero.

HMVD

The HMVD (Horizontal Motion Vector Data) field is a reference to the horizontal motion vector data (MVD). HMVD=0, if the V flag is 0, the packet starts with a GOB header, or the last MB placed in the previous packet has an MTYPE value other than MC. The HMVD value should be between –15 and +15.

VMVD

The VMVD (Vertical Motion Vector Data) field is a reference to the vertical motion vector of the MVD data. The value of VMVD=0, if the V flag is 0, the packet starts with a GOB header, or for the last MB placed in the previous packet, the MTYPE value is not equal to MC. The VMVD value should be between –15 and +15.

H.263

RFC 2190 (RTP): http://www.cis.ohio-state.edu/htbin/rfc/rfc2032.html

H.263: http://www.itu.int/itudoc/itu-t/rec/h/h263.html

The H.263 protocol defines a format for encapsulating an H.263 bitstream into RTP (Real-time Transport Protocol) packets. There are three modes defined for the H.263 payload. An RTP packet can use one of three H.263 video stream modes depending on the desired network packet size and H.263 encoding options. The shortest H.263 header (Mode A) supports GOB (Group of Block) fragmentation. H.263 long headers (Modes B and C) support stream breaking into macroblocks (MBs).

For each RTP packet, a fixed-length RTP header is followed by an H.263 content header, followed by a compressed H.263 bitstream. The size of the H.263 content header depends on the mode used. The diagram of the RTP H.263 video package is shown in the figure.

RTP H.263 video package

In Mode A, the H.263 header contains 4 bytes. This mode supports RTP fragmentation. Mode B uses an 8-byte H.263 header and each packet starts at the MB boundary without the PB option. The 12-byte H.263 header defines Mode C, which supports fragmentation at MB boundaries for frames with the PB option.

The mode of each H.263 stream header is indicated by the values of the F and P fields. Packets of different modes may be intermingled. The title format for mode A is shown in the following figure.

Bits			Octet

	SBIT	EBIT

R (continued)

H.263 header structure for Mode A.

F

A flag indicating the H.263 stream header mode.

P

Flag for the optional PB mode defined in the H.263 standard.

Regular type I or P frame.

1 Frame PB.

If F=1, the P field shows the mode:

Regular type I or P frame.

0 Mode B.

1 Mode C.

SBIT

Start bit. The number of significant bits that should be ignored in the first byte of data.

EBIT

The final bit. The number of least significant bits that should be ignored in the last byte of data.

SRC

The source format (bits 6, 7, and 8 of the TYPE field defined by H.263) that specifies the resolution of the current image.

I

Picture encoding type (bit 9 in PTYPE, defined by H.263):

0 Intra-coding (internal).

1 Inter-coding.

U

The U field has a value of 1 if the current picture header has had (1) the Unrestricted Motion Vector option set, specified by bit 10 in the PTYPE field as defined by H.263.

S

The S field has a value of 1 if the current picture header has been set to (1) the Syntax-based Arithmetic Coding option, specified by bit 11 in the PTYPE field as defined by H.263.

A

The A field has a value of 1 if the current picture header has been set to (1) the Advanced Prediction option, specified by bit 12 in the PTYPE field as defined by H.263.

R

DBQ

A differential quantization parameter used to calculate the quantization parameters for a B frame based on the quantization parameter for a P frame when using the PB frames option. The value of this field shall be equal to DBQUANT (defined in H.263). The field value is set to zero in cases where the PB frames option is not used.

TRB

TR

The header format for mode B is shown in the following figure.

Bits						Octet

		SBIT			EBIT
			QUANT
GOBN
MBA (continued)
				HMV1
HMV1 (continued)			VMV1
VMV1 (continued)		HMV2
HMV2	VMV2

H.263 header structure for Mode B.

Fields F, P, SBIT, EBIT, SRC, I, U, S and A are defined in the same way as for mode A.

QUANT

The quantization value for the first MB encoded at the beginning of the packet. If the packet begins with a GOB header, the QUANT field is 0.

GOBN

GOB number at the beginning of the package. The GOB number is set differently for different resolutions.

MBA

Address inside GOB of the first MB in the packet (counted from zero in scan order). For example, the third macroblock in any GOB will have MBA=2.

R

The field is reserved and must have a null value.

HMV1, VMV1

Prediction of the vertical and horizontal motion vector for the first macroblock in a given packet. When four motion vectors with the advanced prediction option are used for the current macroblock, these vectors are motion vector predictors for block number 1 in the macroblock. Each 7-bit field encodes the motion vector predictor in half resolution using 2's complement.

HMV2, VMV2

Predictions of vertical and horizontal motion vectors for block number 3 in the first macroblock of this packet when four motion vectors are used with the advanced prediction option. This is necessary because block number 3 in a macroblock requires different motion vector predictions from other macroblocks. The described fields are not used in cases where the MB has only one movement vector. Each 7-bit field encodes the motion vector predictor in half resolution using 2's complement.

The header format for mode C is shown in the figure.

Bits						Octet

		SBIT			EBIT
			QUANT
GOBN
MBA (continued)
				HMV1
HMV1 (continued)			VMV1
VMV1 (continued)		HMV2
HMV2	VMV2

RR (continued)

H.263 header structure for Mode C.

Fields F, P, SBIT, EBIT, SRC, I, U, S, A, DBQ, TRB and TR are defined in the same way as for mode A; fields QUANT, GOBN, MBA, HMV1, VMV1, HMV2, VMV2 - as for mode B.

R.R.

The field is reserved and must have a null value.

H.235

H.235: http://www.itu.int/itudoc/itu-t/rec/h/h235.html

The H.235 protocol extends the H.3xx series of recommendations to add security services such as Authentication and Privacy (data encryption). H.235 must work with other H-series protocols that use H.245 as the control protocol.

All H.235 messages are encrypted in the same way as ASN.1.

Ministry of Education of the Russian Federation

MOSCOW STATE INSTITUTE

ELECTRONICS AND MATHEMATICS (TECHNICAL UNIVERSITY)

Abstract on the subject

Computer network management

“Internet telephony. H.323 protocol"

Checked by Kharlamov A.G.

Performer Group S-94

Murchie A.E.

Moscow 2010

Introduction

In just a few years, IP telephony technologies have evolved significantly, and the solutions common today are significantly different from previous ones. On the one hand, this is due to the development of hardware solutions, in particular the emergence of powerful backbone and transit routers and high-speed telecommunication channels. On the other hand, one cannot help but note the emergence of such qualitatively new technologies as dynamic routing taking into account the quality of service in multiservice IP networks and resource reservation for monitoring the quality of service of transit routers.

As it develops, IP telephony undergoes important qualitative changes: from an additional service, it gradually turns into a kind of basic service, which may soon become one of the components of multiservice technology.

It should also be kept in mind that IP telephony is not just an alternative to regular telephony. The relevance of the development of IP telephony solutions is due not only to the possibility of reducing the costs of telephone conversations and infrastructure maintenance (although this, of course, is important). In a strategic plan, IP telephony can become a single technical platform that will combine solutions for data and voice transmission, as well as for processing and subsequent use of this information in all business processes. Thus, the development of IP telephony in a certain sense is a means of increasing labor productivity and business development.

Protocol H .323

In 1990, the first international standard in the field of video conferencing, the H.320 specification, was approved to support video conferencing over ISDN. Then the ITU-T approved a whole series of recommendations related to video conferencing. This series of recommendations, often referred to as H.32x, includes standards H.321-H.324 in addition to H.320, which are intended for different types of networks. In the second half of the 90s, IP networks and the Internet received intensive development. They have evolved into a cost-effective data transmission medium and have become almost ubiquitous. However, unlike ISDN, IP networks are poorly suited for transmitting audio and video data. The desire to use the established structure of IP networks led to the appearance in 1996 of the H.323 standard, which contains descriptions of terminal devices, equipment and network services designed to implement multimedia communications in packet-switched networks (for example, Intranet or Internet). H.323 terminal devices and network equipment can transmit data, voice and video information in real time. The H.323 recommendation does not define: the network interface, the physical medium for transmitting information, and the transport protocol used in the network. The network over which communication between H.323 terminals occurs can be a segment or multiple segments with a complex topology. H.323 terminals can be integrated into personal computers or implemented as stand-alone devices. But support for voice exchange is a mandatory feature for any H.323 device.

· bandwidth management;

· Possibility of network interaction;

· platform independence;

· support for multipoint conferences;

· support for multicast transmission;

· standards for codecs;

· support for multicast addressing.

Bandwidth Management

Internet conferences

Platform independence

H.323 is not tied to any hardware or software technology solutions. Applications that interact with each other can be created on different platforms and with different operating systems.

Multipoint conference support

The H.323 recommendations allow for a conference with three or more participants. Multipoint conferences can be held either with or without a central controller - MCU (multipoint conference unit).

Multicast support

Codec Standards

H.323 sets standards for encoding and decoding audio and video streams to ensure compatibility between equipment from different manufacturers. At the same time, the standard is quite flexible. Requirements are formulated, the fulfillment of which is mandatory, and there are optional features, if used, it is also necessary to strictly follow the standard. In addition, the manufacturer may include additional features in multimedia products and applications if they do not contradict the mandatory and optional requirements of the standard.

Compatibility

There may be cases where conference participants want to communicate with each other without worrying about compatibility issues among themselves. The H.323 recommendations support the discovery of common end-user equipment capabilities and establish the best possible encoding, calling, and control protocols common to conference participants.

Rice. 4.2.

The gateway is not a required component of an H.323 network. It is only necessary when it is necessary to establish a connection with a terminal of a different standard. This communication is ensured by the translation of protocols for establishing and terminating connections, as well as data transfer formats. According to H.323, a multimedia gateway is an optional element in an H.323 conference. It can perform many different functions. Its typical function, for example, is the task of converting transmission protocol formats (for example, H.225.0 and H.221). H.323 gateways are widely used in IP telephony to interface IP networks and digital or analogue switched telephone networks (ISDN or PSTN). If there is no gateway in the network, one of its functions must be implemented - converting the PSTN number into the transport address of the IP network using other means. From the side of networks with IP packet routing, as well as from the PSTN side, the gateway can participate in connections as a terminal or conference control device.

Multipoint conference configurations can be centralized, decentralized, hybrid, or mixed.

Rice. 4.3.

Centralized multipoint conference requires an MCU device. Each terminal exchanges audio, video, data and control commands with the MCU in a point-to-point manner. The MCU controller, using the H.245 protocol, determines the capabilities of each terminal. The MP processor generates the multimedia streams necessary for each terminal and sends them out. In addition, the processor can provide conversions for streams from different codecs at different data rates.

The hybrid conferencing scheme is a combination of the previous two. The H.323 terminals participating in the conference multicast the audio-only or video-only stream to the remaining participants without sending it to the MCU. The remaining streams are transmitted using a point-to-point scheme between the terminals and the MCU. In this case, both the controller and the MCU processor are involved.

Rice. 4.4.

In a mixed conferencing scheme, one group of terminals can work according to a centralized scheme, and another group - according to a decentralized one.

Zone Controller (or Gatekeeper)- a recommended, but not required, device that provides network management and acts as a virtual telephone exchange.

The zone controller provides call control services for H.323 endpoints, such as address translation and bandwidth control according to the RAS protocol. The zone controller in an H.323 network is not a required component. However, if it is present in the network, then terminals and gateways must use its services. The H.323 standard defines both the required services of an area controller and the additional (optional) functionality that it can provide.

An optional feature of the zone controller is call routing. Endpoints send ringing messages to the zone controller, which routes them to destination endpoints. Alternately, endpoints can send call signaling messages directly to each other. This opportunity is valuable for current control requests and management of requests on the network. Routing calls through a zone controller provides better network efficiency because the controller can make routing decisions based on a number of factors, such as load balancing among gateways.

The services offered by the zone controller are defined in the RAS and include address broadcast, reception control, bandwidth control and area control. H.323 networks that do not have a gateway controller do not have these capabilities. H.323 networks containing IP phones and gateways must contain a zone controller to translate incoming E.164 phone addresses into transport addresses. The zone controller is a logical component of H.323, but it can also be implemented as part of a gateway.

Mandatory Zone Controller Features

Address translation
A call originated within an H.323 network can be used to address the desired terminal using its alias (short name). A call originating outside the H.323 network and received through a gateway to be addressed to the recipient terminal may use a telephone number in accordance with the E.164 recommendation (for example, 310-442-9222). This recommendation is used to address ISDN subscribers. The zone controller converts the received E.164 telephone number or alias into the network address (for example, 204.252.32.156 for an IP network) of the destination terminal. The destination endpoint can be reached using this network address.
Registration Management
The zone controller can manage the registration of endpoints in the H.323 network. In this case, RAS messages are used: registration request ( ARQ), confirmation ( ACF) and deviation ( ARJ). Registration control can be a dummy function that admits all endpoints to the H.323 network.
Bandwidth Management
The controller provides bandwidth management using RAS messages: bandwidth request (BRQ), acknowledge (BCF) and reject (BRJ). For example, if the network manager has defined a threshold for the number of simultaneous connections for an H.323 network, the zone controller may refuse to establish new connections unless that threshold is reached. As a result, it is possible to limit the total allocated bandwidth to some portion of the total bandwidth of the data network, leaving the remaining bandwidth for data applications. Bandwidth management can also be a dummy function that simply receives requests without processing them.
Optional Zone Controller Features
- Call management
  The zone controller can route calls between H.323 endpoints. In a point-to-point conference, the area controller can handle H.225 ringing messages. Alternatively, the zone controller may allow endpoints to independently exchange H.225 call signaling messages directly with each other.