Atos Unify OpenScape Voice

Atos Unify OpenScape Voice - программное приложение корпоративной голосовой связи

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программное приложение корпоративной голосовой связиOpenScape Voice is a native SIPbased real-time Voice over IP system scalable up to 100,000 users per system and a virtually unlimited number of users when OpenScape Voice systems are networked. It provides a complete and feature-rich set of business class features and perfectly fits into data center installations running as a virtualized appliance on VMware hypervisors. Alternatively, it runs on highly reliable, redundant and fault-tolerant hardware. OpenScape Voice can be deployed on premise or in data centers as Private Cloud or as multi tenant capable Hosted/Public Cloud solution.

OpenScape Voice is a carrier-grade enterprise voice solution meaning 99.999% reliability – that translates to less than 5 1/2 minutes of downtime per year! The server nodes are designed so that if one fails, the other server node can support 100% of the call load. The server nodes can operate with100% call failover support even when they are geographically separated, greatly reducing the costs, and the amount of time implementing a disaster recovery strategy. And remote offices can be protected with an OpenScape Branch solution – a survivable branch office solution for OpenScape Voice. OpenScape Branch not only offers survivability, but it includes a media server, firewall, Session Border Controller, and integrated PSTN gateway, all in a single appliance form factor. The value of OpenScape Branch goes beyond survivability, its activity contributes to lower the overall deployment, bandwidth and service costs.

Comprehensive Unified Communications

OpenScape Voice is always part of a solution landscape. The most basic solution includes:

Deployment scenarios

OpenScape Voice is designed to mcover multiple customers and target market deployment scenarios. The key deployment scenarios are:

Configurations

OpenScape Voice Integrated Simplex

This configuration consists of a system that provides the mediumsized voice solution (with or without UC) in a single server. The OpenScape Voice and UC applications are deployed as a single node platform; as such there is no carrier-grade reliability due to the lack of redundancy. Additionally, the following deployment highlights provide the ability to run on the same physical platform:

This model represents an offering that would be of interest to a customer that wants a medium sized VoIP business solution (up to 5,000
users) at a low cost, and therefore is willing to accept some risk of downtime (due to no redundancy). This is also a configuration that is prevalent in the "try-and-buy" program.

OpenScape Voice Duplex

This deployment model illustrateshow the OpenScape Voice can be operated as a more robust and scalable duplex system. It provides carrier-grade reliability by running two platforms in a redundant twonode cluster that executes in an active-active mode. Should one of
the nodes fail, then the remaining partner node would assume the call load of the failed partner (and would handle 100% of the call traffic) and would continue to provide uninterrupted call processing. No calls would be dropped due to the failover from duplex to simplex operation, or when the system reestablishes duplex operation. The duplex mode also allows for the possibility of maintaining call processing operation while an OpenScape Voice upgrade is performed.

Currently, a mix of physical nodes and virtual nodes is not supported. A cluster of 2 physical nodes or a cluster of 2 virtual nodes are the
only scenarios supported. This model is appropriate for larger customers (up to 100,000 users) as well as for customers that want carrier-grade reliability.

OpenScape Voice virtualized architecture

The most important features provided by virtualization are the reduced number of servers and the capability of our solution to be hardware-agnostic. Therefore, OpenScape Voice/UC Suite operation in a virtual environment enables the following capabilities: Server consolidation The applications and virtual machines deployed onto a VMware host can use different guest operating systems, i.e. OpenScape Voice (Linux) and OpenScape Concierge (Windows) can both be deployed onto the same VMware host and share its physical resources. Hardware independence Having many hardware server vendors and models in a data center environment adds complexity and cost to the operation, therefore, customers often look to standardize their IT hardware infrastructure. Virtualization allows customers to deploy applications onto any hardware platform, assuming it has been certified by VMware and it meets the resource requirements of the application, as described in this document.

Application and server platform

At the heart of the OpenScape Unified Communications is the OpenScape Voice real-time, SIPbased, Voice over IP application that provides the carrier-grade level of redundancy, reliability and scalability required for missioncritical deployments. OpenScapeVoice operates on commercial servers over QoS managed networks. The OpenScape Voice VoIP system provides the following key features:

• SIP B2BUA
• Enterprise telephony features
• User management and address translation functions
• Interface to monitor and control media transactions including pure telephony
• Interface for advanced services, such as presence services, billing services, collaboration services, etc.
• Gateway selection and hunting
• Routing and translation functions comparable to a carrier-grade solution

OpenScape Voice is designed as an open standards platform that runs on standard rack-mountable computing hardware. The base system software runs on the SUSE Linux Enterprise Server operating system – SLES12 64 bit. This is combined with cluster control software to run all parts of the system as a redundant unit. The system runs on a single server or a dual server cluster, depending on the number of users and customer requirements.

Hardware redundancy and cluster connectivity

OpenScape Voice controls and supervises call setup; the actual media payload (voice and/or video) is carried over the LAN/WAN between endpoints. The administration, call control, and billing traffic are carried over redundant pairs of network interface cards through redundant, interconnected L2/L3 switches that provide redundant networking. The OpenScape Voice redundant configuration can be deployed as follows:

Security

OpenScape Voice supports SRTP for media encryption. SRTP secures voice communication by encrypting the media packets between media devices.

End-to-end media encryption is implemented using a "best effort" mechanism that is dependent on SRTP support from the media devices that are involved in the connection. An encrypted SRTP connection is established when both media endpoints support SRTP and use a common key management protocol (e.g., MIKEY0 or SDES); if an SRTP connection cannot be established, the call will still be completed but with an unencrypted RTP.

SRTP MIKEY (Profile 0) is supported on connections between nearly all media endpoints of the OpenScape Unified Communications. With OpenScape Voice, SRTP SDES (Profile 1) is supported for connections between nearly all media endpoints of the OpenScape Unified Communications solution and is the preferred SRTP key management protocol to use. OpenScape Voice also supports media encryption for connections that are signaled over the SIP-Q interface between itself and:

Solution media devices that do not support SRTP or do not support a compatible key management protocol should negotiate down to RTP.

OpenScape Voice supports enhanced SDP backward compatibility for best effort SRTP that allows for support of third-party SIP endpoints that do not support SRTP and do not properly handle SRTP to RTP fallback which might otherwise have resulted in call failures.

SRTP requires a secure signaling connection to be used between the media device and the OpenScape Voice server. For SIP devices, TLS is used, and for the OpenScape Media Server, IPSec is used to secure the signaling connection. All Session Border Controllers (SBCs) that are approved for use with OpenScape Voice support SRTP media encryption using transparent media relay, or "passthrough". In addition, OpenScape  SBC (V2 and later) can support SRTP termination of MIKEY0 and SDES key management, which allows for SRTP to RTP termination and also SRTP mediation between MIKEY0 and SDES key exchange methods for media connections routed via the SBC. This interworking is useful, for example, to maintain maximum media stream security within the enterprise network when using SIP trunks to a service provider that does not support SRTP, or to ensure security for remote subscribers (e.g., home workers) that access OpenScape Voice via an un-secure network.

Security: TLS

OpenScape Voice provides Transport Layer Security (TLS) for protecting signaling communications on SIP endpoint, SIP server, and
SIP-Q server interfaces. OpenScape Voice also supports optional use of TLS to secure the transport of XML messages on the SOAP server management interface. This feature also provides for client user authentication and rolebased authorization for controlling access to OpenScape Voice management functions. The system's static capacity for TLS is 50,000 endpoints. Dynamic capacity depends on customer feature configuration and call rate.

Security: IPSec

OpenScape Voice supports optional use of IPSec for protecting the OpenScape Voice SOAP and SNMP management interfaces to the external OpenScape Voice Assistant and CMP, as well as for protecting the MGCP signaling interface to a media server.

Security: Event logging

Security event logging can be provided by using the standard Syslog mechanisms for both platform and application or optionally by using the Linux Audit OS module.

OpenScape Software Assurance

OpenScape Software Assurance assures that customers are kept on the latest software version of OpenScape products. Continuous software upgrades guarantee longterm software stability and up-todate security features and improve the OpenScape Unified Communication interfaces towards other products and solutions.

Upgrade/Migration to OpenScape Voice V10

Upgrades require an upgrade license per user license purchased in the previous release.
For new installations, the current available system server deployment options are:

Earlier server version simplex or duplex customers who wish to migrate to OpenScape Voice V10 software will be required to change out their platform to a supported Lenovo or Fujitsu server:

Network connectivity

SIP trunking to service providers

Many enterprises are already using VoIP; however, many use it only for communication on the enterprise LAN. SIP trunking takes the VoIP concept beyond this LAN application. The full potential for IP communications can be realized only when the communication is taken outside of the corporate LAN. The OpenScape SBC provides secure connection of OpenScape Voice to carrier-based SIP trunking services.

SIP Private Networking

SIP Private Networking uses the SIP-Q protocol currently used for OpenScape Voice-to-OpenScape Voice/4000/Business connectivity. This protocol provides feature transparency among users in these networked systems.

QSIG networking

QSIG networking provided by the OpenScape Branch supports SIPQ, which permits OpenScape Voice to interwork with OpenScape Voice, OpenScape 4000, OpenScape Business or a QSIG PBX

Call Admission Control features

The integrated Call Admission Control (CAC) features provide for management of the bandwidth used for the transport of media traffic (such as RTP audio, T.38 fax,
and video) through the bottleneck links that may exist in an enterprise network. This feature ensures that real-time media calls are only established when the necessary bandwidth resources are available on all access links that exist between the two communicating endpoints. The following are examples of the functionality the Call Admission Control feature provides:

Supported gateways

For all calls made to the LegacyPSTN TDM network, a gateway on the enterprise edge is required. The survivable OpenScape Branch family of integrated gateways provide access to the Legacy PSTN network.

Features

Keyset telephone user features

Keyset telephone user features provide multiple line capability and other associated functions for a SIP endpoint configured as a keyset. Keysets are sometimes known as multiline telephones. Any of the OpenScape Desk Phone CP SIP phone family can be configured as keysets.

Brochures

Atos Unify OpenScape Voice V10 (Brochure)

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