Qsig

Qsig is an inter-PBX signaling system designed for multiple PBX system platform networks. The proprietary nature of IFTN solutions restricted customer configuration flexibility to a single supplier’s product platform. Qsig in its current form originated during the 1990s as a standardization effort by the IPSN Forum, a group of Western European PBX equipment suppliers, with Siemens and Alcatel at the forefront of the movement. IPSN work efforts were handed off to the ECMA and the International Telecommunications Union (ITU) for the formalization of issuing standards and specifications. Qsig is based on the ITU’s Q.93x series of recommendations for basic services and generic features and Q.95x series for supplementary services.

The major benefits for developing Qsig were outlined in the Qsig handbook originally published by the IPNS Forum.

Vendor independence. The nonproprietary nature of Qsig, based on open international standards and supported by all of the leading global PBX suppliers, allows customers to configure an intelligent communications system network when using equipment from more than one supplier.

Guaranteed interoperability. A memorandum of understanding (MoU) signed by the leading global suppliers signifies commitment to Qsig specifications, facilitates interoperability performance tests, and assures customers that they will be able to operate communications networks with a mix of supplier equipment.

Free-form topology. Qsig does not impose the use of a specific network topology, so it can be implemented with any network configuration: meshed, star, main/satellite, etc. Existing networks can, regardless of their topology, be upgraded to Qsig. Newly designed networks can be installed with the most effective and economical topology.

Unlimited number of nodes. There are no nodal limits for a Qsig network. New nodes can be added as needed.

Flexible numbering plan. Qsig does not impose any number plan restrictions for the network, thereby allowing customers to freely adopt customized numbering plans.

Flexible interconnection. Qsig will work over any type transmission network for linking PBX systems, including two- and four-wire analog tie lines, digital leased lines (including ISDN PRI and BRI), radio and satellite links, and VPN services provided by interchange carriers. Associated transmission delays are managed and controlled according to Qsig specifications.

Public ISDN synergy. There is network service compatibility between public and private ISDN transmission facilities. Applications developed for desktop terminals connected directly to a public ISDN network will also be available to desktop terminals provisioned within the Qsig-based customer private network.

Supplementary services for corporate users. Qsig supports private communications features beyond those defined for public ISDN networks, including caller name ID, call intrusion, do not disturb, path replacement, operator services, mobility services, and call completion on no reply.

Feature transparency. Features and functions supported by any network node can be transparently supported across the network to station users configured behind other network nodes. Qsig is structured and organized to adapt to service levels offered by different PBX systems, and it allows each network node to provide only the required level of service. There is an exchange of high-level services between any two nodes, via transit nodes with lower service levels: transit nodes pass communications and control signals between systems.

Innovation. Qsig does not restrict individual PBX manufacturers from developing customized, unique features. A special mechanism within Qsig, generic functional procedures (QSIG GF), provides a standardized method for transporting nonstandard Qsig features. As defined in Qsig GF, the basic rules related to feature transparency allow end-to-end communication through the network, regardless of network structure. Qsig does not prevent the use of innovative, proprietary system features across the customer network and allows for customized new feature development negotiation between PBX suppliers and customers.

Multiapplication domain. Qsig is not restricted to PBX systems and can support applications requiring other peripheral communications equipment, such as VMS, fax servers, data processing equipment, and multipoint conferencing systems.

Evolution. Qsig has an evolutionary path to support communications features, functions, and applications that are developed in the future.

Qsig Architecture

Qsig standards specify a signaling system at the ITU-T ISDN “Q” reference point, which is intended primarily for use on a common channel, although Qsig can be implemented over any suitable inter-PBX connection platform. The “Q” reference point, the logical signaling point between two PBXs, was defined explicitly for the Qsig. The physical connection point to the PBX system is made at the “C” reference (also a new ISDN reference point). There are three sublayer protocols at Layer 3, including the Qsif GF procedures. Qsig GF protocol provides a standardized mechanism to exchange signaling information for the control of supplementary services and additional network features (ANFs) over customer networks.

Qsig basic call (BC) message sequence is an intermediary transit node linking two endpoint PBX systems. Qsig BC is a symmetrical protocol designed for peer-to-peer networks, and it includes transit node capability.

ECMA also has been working on enhancements to its Qsig specifications to support broadband PBX networks. B-Qsig is an extension of Qsig, using many standards as possible available from the ITU-T and ATM forums.

Qsig Supplementary Services and ANFs

The following is a listing of the Qsig supplementary services and ANFs:

• Advice of charge

• Call completion

• Call forwarding and diversion

• Call interception

• Call intrusion

• Call offer

• Call transfer

• Call waiting

• Direct dialing in

• Do not disturb

• Identification services

• CLID presentation

• Connected line identification presentation

• Calling/connected line identification restriction

• Calling name identification presentation

• Calling/connected name identification restriction

• Mobile

• Multiple subscriber number

• Operator services

• Path replacement

• Recall

• Subaddressing

• User-to-user signaling

H.323 Architecture Protocols and Procedures

H.323 is an umbrella recommendation that depends on several other standards and recommendations to enable real-time multimedia communications. The following are the key H.323 reference recommendations.

Call Signaling and Control

• H.323: Packet-based multimedia communications systems

• H.225: Call control protocol

• H.235: Security

• H.245: Media control protocol

• Q.931: Digital subscriber signaling

• H.450.1: Generic functional protocol for the support of supplementary services in H.323

• H.450.2-11: Supplemental features: blind call transfer and call diversion, consulting (450.2); call forward, activation/deactivation, interrogation (450.3); call hold (450.4); call park, call pickup (450.5); call waiting (450.6); message waiting (450.7); name interrogation (450.3), call hold (450.4); call park/call pickup (450.5); call waiting (450.6); message waiting (450.7); name identification service (450.8); call completion (450.9); call offer (450.10); call intrusion (450.11)

H323 Annexes

• Annex D: Real time fax over H.323

• Annex E: Multiplexed call signaling

• Annex F: Simple Endpoint Terminal (SET)

• Annex G: Text SET

• Annex H: Mobility

• Annex I: Operation over low QoS networks

• Annex J: Secure SET

• Annex K: HTTP service control transport

• Annex L: Stimulus signaling

• Annex M: Qsig tunneling

• Annex N: QoS

Audio Codecs

• G.711: PCM audio codec 56/64 kbps

• G.722: Audio codec for 7 Khz at 48/56/64 kbps

• G.723: Speech codec for 5.3 and 6.4 kbps

• G.728: Speech codec for 16 kbps

• G.729: Speech codec for 8/13 kbps

Video Codecs

• H.261: Video codec for ≥ 64 kbps

• H.263: Video codec for ≤ 64 kbps

H.323 Version 1 was approved in 1996. Version 1 centered on multimedia communications, such as voice and video over IP data networks. Version 2 was approved in January 1998, and addressed deficiencies in Version 1 and introduced new functionality within existing protocols, such as Q.931, H.245, and H.225, and entirely new protocols. The most significant advances were in security, fast call set-up, supplementary services, and T.120/H.323 integration. There was more efficiency about getting media streams to transfer at a faster rate. The major functions introduced were Fast Connect (also known as Fast Start) and H.245 tunneling (transferring minimal call signals to more quickly establish connection). Version 3, approved in May 1999, had several new annexes or sections focused on H.323-compliant devices for large-scale production networks. It covered bandwidth management and QoS issues and focused on “smart” networks and “dumb” endpoints (master/slave relationship). Some specific areas addressed were:

• Connection over UDP

• Simple endpoint type

• Interdomain communications

• H.263 and packetization

• H.GCP decomposition architecture

H.323 Version 4 was approved in November 2000 and contained many enhancements in a number of important areas, including reliability, scalability, and flexibility. Many of the new features facilitate more scalable gateway and MCU solutions to meet the market requirements for IP telephony. Version 4 addressed the following issues:

• Gateway decomposition

• Multiplexed stream transmission

• Supplementary services

• Additive registrations

• Alternate gatekeepers

• Usage information reporting

• Endpoint capacity

• Tones and announcements

• Mapping aliases

• Indicating desired protocols

• Bandwidth management

• Reporting call status

• 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

H.323 Architecture Components

H.323 specifies components, protocols, and procedures for real-time point-to-point and multipoint multimedia communications over packet-based networks. It also sets interoperability guidelines for communication between H.323-enabled networks and the H.32X-based family of conferencing standards.

An H.323 implementation requires four logical entities or components. These are terminals, gateways, gatekeepers, and MCUs. A fifth component element, known as border elements, is optional. Terminals, gateways, and MCUs are collectively known as endpoints. Although an H.323 network can be configured with only terminals, the other components are essential to provide greater practical usefulness of the services.

Terminal

A terminal, or a client, is an endpoint where H.323 data streams and signaling originate and terminate. In an IP-PBX system, it may be an IP telephone; a non-IP communications device, such as an analog telephone with an IP adapter module; or a PC client softphone with an H.323 compliant stack. A terminal must support audio communication; video and data communication supports are optional.

Gateway

A gateway is an optional component in an H.323-enabled network. A gateway will be required if at least one of the terminals does not conform to H.323 standards but is designed for a different type of network. The gateway is usually located at the interface between the two networks. Through the provision of gateways in H.323, it is possible for H.323 terminals to interoperate with other H.32X-compliant conferencing terminals. For example, a PC client softphone station user can talk to a station user on an analog telephone. A gateway provides data format translation, control signaling translation, audio and video codec translations, and call setup and termination functionality on both sides of the network. Gateway equipment is composed of a Media Gateway Controller (MGC) and a Media Gateway (MG), which may co-exist or exist separately. The MGC handles call signaling and other nonmedia-related functions, and the MG handles the media. There are different types of gateways required to support H.310, H.320, H.321, H.322, or H.324 endpoints.

Gatekeeper

A gatekeeper is an optional component of an H.323-enabled network. Gatekeepers ensure reliable, commercially feasible communications and are used for admission control and address resolution. The gatekeeper may allow calls to be placed directly between endpoints or it may route the call signaling through itself to perform functions, such as followme/find-me, forward on busy, etc. A gatekeeper is often referred to as the controller of the H.323 enabled network, because it provides central management and control services. When a gatekeeper exists, all end- points (terminals, gateways, and MCUs) must be registered with it. Registered endpoints’ control messages are routed through the gatekeeper.

The gatekeeper and the endpoints it administers form a management zone. A gatekeeper provides several services to all endpoints in its zone.

Address translation. A gatekeeper maintains a database for translation between aliases, such as international phone numbers, and network addresses.

Admission and access control of endpoints. This control can be based on bandwidth availability, limitations on the number of simultaneous H.323 calls, or the registration privileges of endpoints.

Bandwidth management. Network administrators can manage bandwidth by specifying limitations on the number of simultaneous calls and by limiting authorization of specific terminals to place calls at specified times.

Routing capability. A gatekeeper can route all calls originating or terminating in its zone. This capability provides numerous advantages: accounting information of calls can be maintained for billing and security purposes; a gatekeeper can reroute a call to an appropriate gateway based on bandwidth availability; and rerouting can be used to develop advanced services such as mobile addressing, call forwarding, and voice mail diversion.

Multipoint Control Unit

The MCU is an optional component of an H.323-enabled network. It is responsible for managing multipoint conferences (two or more endpoints engaged in a conference). It consists of a mandatory Multipoint Controller (MC) that manages the call signaling and optional Multipoint Processors (MPs) to handle media mixing, switching, or other media processing. Although the MCU is a separate logical unit, it may be combined into a terminal, gateway, or gatekeeper.

The MC provides a centralized location for multipoint call set-up. Call and control signaling are routed through the MC so that endpoint capabilities can be determined and communication parameters negotiated. An MC may also be used in a point-to-point call, which can later be extended into a multipoint conference. The MC may also determine whether to unicast or multicast the audio and video streams depending on the capability of the underlying network and the topology of the multipoint conference.

The MCU is required in a centralized multipoint conference, where each terminal establishes a point-to-point connection with the MCU. The MCU determines the capabilities of each terminal and sends each a mixed media stream. In the decentralized model of multipoint conferencing, an MC ensures communication compatibility, but the media streams are multicast and the mixing is performed at each terminal.

Border Elements

Border elements are often colocated with a gatekeeper. They exchange addressing information and participate in call authorization between administrative domains. Border elements may aggregate address information to reduce the volume of routing information passed through the network and assist directly in call authorization/authentication between two administrative domains or via a clearinghouse.