Siemens | OpenScape Voice OpenStage 20 E SIP | OpenScape Voice Datasheet

OpenScape Voice V4R1
Simple yet Sophisticated
OpenScape Voice is the world's only Enterprise and Carrier-Class
Voice Application that is fully integrated with, and offered as part of,
a complete Unified Communications Solution. It provides a comprehensive solution for building or migrating a Large Enterprise, Carrier
or Hosting Service Provider Voice Communications Network.
The essence of OpenScape Voice is its architecture, which forms the
foundation for a new IT-integrated and business process-focused
communications solution, providing the lowest Total Cost of Ownership (TCO) and Best Return on Investment (RoI).
Communication for the open minded
Siemens Enterprise Communications
The OpenScape Voice
Solution Landscape
OpenScape Voice is an enterprise-class
voice application that is fully integrated
with, and offered as part of, a complete
unified communications solution, the
OpenScape Unified Communications (UC)
Suite. Supporting open standards, it is
designed not only for centralized deployment within a distributed enterprise, but as
a highly viable option for site-based deployments as well.
OpenScape Voice is designed to provide
the architectural strength to such a framework through its rich feature set, scalability, resiliency, adherence to open standards, and manageability. As an enabler of
Information Communication Technology
(ICT) convergence, OpenScape Voice creates technology choices, allowing customers to implement well thought out communication strategies at their own pace.
OpenScape Voice forms the foundation for
a new multimedia business communications paradigm: pervasive, intuitive, interactive and effective. It allows customers to
build strong pillars of multimedia communications functionality that provide business process integration (BPI) and business
process enhancements through communications-enabled business processes (CEBP).
It boasts a very credible claim to being a
core component of enterprise communications based on open standards, becoming a
business tool to optimize and enhance
enterprise communications and to enrich
its processes – a tangible change from the
traditional TDM, converged and IP PBXs.
OpenScape Voice serves enterprises of
mid- to very large size and multi-tenant
hosted services offered by Service Providers (SP). It serves as a core component of
communications and is able to offer
choices not only in unified communications, but to unify communications.
Best for Your Business
OpenScape Voice offers a key solution at
the infrastructure level, interworking with
a number of components to provide Voice
over IP communications to the enterprise.
The OpenScape Voice solution is in turn
part of a broader solution set in the customer’s environment, and as such, works
with a variety of applications to enhance
and support the customer's business practices.
The entire OpenScape UC Suite portfolio is
optimized for the demands of businesses –
easy to put into practice, reliable in performance, and easy to use. With it, you
become even more efficient.
Key Attributes of OpenScape Voice
The key attributes are as follows:
Simple yet Sophisticated:
• Simple
– To deploy and manage
– To migrate and upgrade
– Per-user licensing model
• Sophisticated
– Architecture
– Feature set
– Networking options
– Interworking and applications
Mature yet Innovative:
• Mature
– Proven, stable and reliable
– In service worldwide
– Large enterprise, carriers and MSPs
– Comprehensive deployment options
• Innovative
– Cloud and social media
– Expanded networking architecture
– Built on a sure and proven foundation
Standards yet Open options:
• Standards
– Most open solution
– Standards compliance
• Open options
– Configuration options
– Applications options
– Networking and branch options
– APIs for building new options
Economic yet Green
• Economic
– Lowest TCO, best RoI
– Simple but powerful management
– IT and data-center integrated
• Green
– With OpenStage devices provide lowest carbon footprint per user
– Lowest power-cost per user
With OpenScape Voice, businesses benefit
from the investments already made in their
customers, partners, employees, and communications infrastructure. OpenScape
Voice V4 further demonstrates how SIP
platforms can reduce communication
costs. The separation of voice and data no
longer exists. Only one infrastructure
needs to be maintained. Processes and
applications are more reliable and can be
shared across the enterprise.
As a product, OpenScape Voice V4 has
experienced a natural technological evolution far in advance of its competitors. It has
been built from the ground up to ensure
that leading-edge software technologies
could be employed, rather than a patchwork of engineering. Therefore, OpenScape Voice is able to provide seamless,
standards-based integration with multivendor systems and applications.
You decide when, where, how, and to what
extent to invest in innovative technology.
You can choose from a broad range of IP
convergence platforms, optiPoint and
OpenStage phones, and an
OpenScape Personal Edition solution (softclient). You set the pace in accordance with
your demands and ideas.
Benefits of Installing
OpenScape Voice V4
• High availability and cost effective solution for enterprises in the medium to
very large range
• Carrier-grade reliability and resiliency
• Scalability to tens of thousands of users
• Open unified communications platform
– Any media, any time, anywhere
– Support of open standards
• Excellent CAPEX and OPEX efficiencies
of scale
• Seamless migration path from converged IP to SIP
• Web Services architecture
– Access for end-user self-management
– Integration with other Web-based
applications and management
• Global licensing
• Batch command file and mass provisioning interface
• Communications as a Service (CaaS)
IP Communication
mode, and can switchover automatically
without loss of active calls or billing
OpenScape Voice V4 offers a wide range of
options for transforming your corporate
communications solution into real-time IP
communication. You can reduce your IP
infrastructure costs even further by using
high-performance gateways and standardized compression procedures. OpenScape
Voice V4's “Any-to-Any” IP payload switching ensures that you get the highest availability and quality.
OpenScape Voice also provides a Survival
Authority (SA), a separate component
which normally resides on the OpenScape
Voice Assistant administration server. The
SA can assist in determining the proper
cluster response in the event that communication between the two nodes is severed
due to a network failure. Activation of the
Survival Authority is optional in the case of
collocated cluster nodes, but required in
the case of geographic separation of the
Resiliency, Reliability and
Resiliency is about how well systems
behave under stress conditions (e.g. overload). OpenScape Voice has a WorkerWorker software architecture and behaves
far better with overloads and fault conditions than traditional Worker-Standby systems.
Reliability is about how often things fail
and how quickly they are restored to normal operation. It is a key feature of
OpenScape Voice that it does not lose a single call-in-progress or a single billing record
on any single failure.
Recovery is about how fast the system
recovers after faults or overloads. Again
OpenScape Voice, because of its hardware
and software architecture recovers
extremely fast, and is superior to all its
OpenScape Voice software runs on highly
reliable, fault-tolerant industry-standard
servers under the Linux SLES 10 64-bit
operating system. Clustering software protects against hardware and software failures, and controls failover of redundant
Ethernet links and cluster nodes (redundancy is optional for systems below 5000
lines). By ensuring that all functions and
applications maintain unrestricted availability, OpenScape Voice provides a new
level of quality in IP communications.
OpenScape Voice controls and supervises
call setup; the actual voice traffic is carried
over the LAN/WAN between endpoints.
Administration/signaling and billing traffic
is carried over a redundant pair of network
interface cards through redundant, interconnected L2/L3 switches that provide
redundant networking.
The two servers may be collocated or geographically separated. If geographically
separated, the connections between the
two nodes may be established at the layer
3 level using IP routing protocols.
OpenScape Voice utilizes Fujitsu’s PRIMECLUSTER clustering software and Resilient
Telco Platform (RTP) middleware to provide
a highly reliable platform which can operate in both active-active and active-standby
Environmentally Friendly
Two servers versus many – OpenScape
Voice uses only two servers for fully redundant call control. The environmental costs,
measured in terms of the power consumption and CO2 output associated with the
manufacture, acquisition, operation, maintenance and disposal of two servers, are
significantly lower than for a site-based
communications system or a system that
requires more servers.
The unique scalability up to 100,000 users
with only two servers is achieved through
software-based growth, not by adding
more hardware. This leads to:
• Less overhead power usage in the Data
• Lower heating, ventilating and air conditioning (HVAC) requirements in the Data
• Less rack space use in the Data Center
Features in
OpenScape Voice V4
New Features in
OpenScape Voice V4R1
The many new or enhanced features being
introduced with OpenScape Voice V4 can
effectively be grouped into the following
functional categories:
There are several important new solution
components and features being introduced
in V4R1. These include:
• Enhancements to the OpenScape Voice
• Support for integration or interworking
with other products in the OpenScape
UC Solution Landscape
• New smaller server configuration for
simplex configuration for up to 800 SIP
Enhancements to the
OpenScape Voice Softswitch
• Call diversion for invalid destinations
• Connected outgoing line presentation
(COLP) enhancements
• Continuous trace tool
• Deployment Service (DLS) V2.0 R4
• Emergency call handling enhancements
• HiPath MetaManagement enhancements
• Media encryption enhancements
• Remote patching with HiSPA (HiPath
Serviceability Platform for Applications)
Integration or Interworking with Other
Products in the OpenScape UC Solution
• Application-provided billing party via SIP
• Application-provided call correlation via
• Display enhancements for CCBS/NR
• Geographic node separation: low-bandwidth layer-3 cluster interconnect links
• Interworking with HiPath 3000
• Mediatrix 4102 analog adapter support
• Mediatrix gateway enhancements:
media encryption
• OpenScape Branch support
• OpenScape Contact Center integration
• OpenScape Media Server enhancements
– CALEA/LI support
– Support for additional
• OpenScape UC Application integration
• OpenStage support
• Radisys Convedia CMS-3000: security,
QoS and language enhancements
• RG 8700 enhancements: media encryption
• SIP trunking customization options
• Session border controller enhancements
• SIP signaling manager: internal audit
• OpenScape Branch V1R2 (four size
options - up to 50, 250, 1,000 and 6,000
users; see OpenScape Branch)
• OpenScape Voice Entry Edition (for
enterprises of up to 800 users)
• New Attendant Solution (OpenScape
Concierge) that does not need Contact
Center (Automatic Call Distribution
• New Executive / Assistant Application
• Digital alarm and communication server
(DAKS) added as a tested element to the
Solution Landscape
• HiPath 3000 V8 added as a tested element to the Solution Landscape (see
OpenScape Voice V4 Gateways)
• Additional feature interworking with
HiPath 4000, especially Network-Wide
Call Pickup (see OpenScape Voice V4
OpenScape Voice Entry Edition
OpenScape Voice Entry is an attractive
entry-level simplex deployment configuration of the OpenScape Voice Application
• is targeted at smaller enterprises and
supports up to 800 SIP users and 400
• uses the IBM x3250 M2 server, and cannot be extended beyond it's maximum
of about 800 users
• includes an integrated Session Border
Controller (SBC) that can be used for SIP
OpenScape Concierge
The OpenScape Concierge Attendant Console as part of OpenScape Contact Center
Extensions provides comprehensive Attendant function for OpenScape Voice (with
and now also without the use of an external ACD system) and also provides a seamless migration from an IP/TDM platform to
OpenScape Voice. It comprises the following features:
• User-friendly attendant console solution
for OpenScape Voice and HiPath 4000
• Easy installation and configuration
• Character-Separated Value (CSV)
importer for flexible customer data synchronization
• Standard Lightweight Directory Access
Protocol (LDAP) synchronization tool for
customer phone book integration
• Fast call pickup mode for pure attendant
• Optimization of availability through
integration in the OpenScape Contact
Center routing strategy possible
• Operation of the solution on networked
HiPath / OpenScape Voice systems
• Detailed display of customer data for
incoming calls
• Extensive search function for efficient
database research
• Presentation of additional information
about destination stations
• Use of ACD and Computer Telephony
Integration (CTI) functions in the intuitive user interface
• Snapshot function while switching, station status is already known before
• UC-based status information and status
change for UC contacts
• Pictures for contacts can be shown
• Incorporation of a Web browser for
using Web-based information sources
• 252 individual repertory keys (6 sections) with integrated presence function, also for stations in system network
• Up to 20 always visible "speed dial" buttons
• Administration tool for managing the
data volume
• Adaptable layout of Concierge interface
• Extensive real-time and historic reporting, if you use Concierge with
OpenScape Contact Center (OSCC)
• Standard real-time and historic reporting, if you use Concierge without OSCC
• Personal and group park queue
• Appending calls to busy stations
Executive / Assistant Application
The Executive / Assistant Application is an
XML application developed to work only on
the OpenStage Phones 60 and 80 and
interacts with OpenScape Voice to provide
an intuitive implementation of the Executive / Assistant function that up to now has
only been available on TDM systems. It
comprises the following functions:
• Streamline the executive's calling processes with the support of one or more
• Assistants control and manage calls for
executives, providing support with a
great degree of flexibility
• One or more assistants can answer all
incoming calls for the executive, handling them exactly as the executive
• Incoming calls for the executive are
directly forwarded to the assistant, or
both the executive and assistant are signaled simultaneously
• Assistant can always monitor all incoming calls for the executive and react
OpenScape Voice V4
OpenScape Voice V4
Media Servers
OpenScape Voice V4
System management tools for OpenScape
Voice V4 include the following:
• OpenScape Voice Assistant
• RTP Command Line Interface (CLI)
• Deployment Service (DLS)
OpenScape Voice V4 offers the following
media server options:
• RadiSys Convedia CMS-3000 media
server for up to 360 ports
• OpenScape Media Server
up to 75 ports for the internal media
server and up to 500 ports for the standalone media server
for up to 500 channels with G.711 codec
To access the Public Switched Telephone
Network (PSTN), the OpenScape Voice V4
solution provides the following gateway
options for media and signalling:
• HiPath 4000 Survivable Media Gateway
• HiPath 3000 Media Gateway including
HiPath 3000 V8
• RG 8700 Survivable Media Gateway
• RG 2700 Survivable Media Gateway
• Mediatrix Gateways
OpenScape Voice Assistant
For all user configurations, OpenScape
Voice Assistant is the strategic Web-based
tool for administering OpenScape Voice V4.
For installations with fewer than 5000
users, the Assistant can be installed on the
same server as the OpenScape Voice software and the integrated OpenScape Media
Server. In installations with more than
5000 users, it is necessary to install OpenScape Voice Assistant as well as the OpenScape Media Server on a separate, external
RTP Command Line
OpenScape Voice system provisioning and
administration can be performed using a
traditional Command Line Interface (CLI).
Features/functions which must be activated or provisioned only once are still
managed using the RTP CLI, e.g., tracing
and other maintenance functions. The RTP
CLI is always accessible via a secure shell,
and can be accessed via Assistant or
directly from the maintenance port of
OpenScape Voice.
Deployment Service
The Deployment Service (DLS) is a management tool used for administering workpoints in the OpenScape Voice network.
DLS is a Java-based application with a Webbased user interface, and is functionally
integrated into OpenScape Voice Assistant.
The DLS is required to support the mobility
feature on Siemens SIP endpoints. It provides options to migrate existing workpoints and to implement mobile user standards. Other important functions of the
DLS include software deployment, inventory data management, configuration
management, and Plug and Play support.
Multiple media servers can be employed
for large installations or for added reliability and scalability.
RadiSys Convedia
The RadiSys Convedia CMS-3000 is a turnkey, high performance media server for
enterprise-sized OpenScape Voice network
OpenScape Media Server
The OpenScape Media Server is an integral
part of OpenScape Voice systems for
medium-size enterprises supporting from
300 to 50,000 subscribers, per single
server. This software-only server solution
provides tones, announcements and user
prompts to support the functionality of
OpenScape Voice features. Announcements are generated in the language
requested by OpenScape Voice, or in a
configurable default language. The OpenScape Media Server also supports redundancy, station controlled conferencing,
and media encryption using Secure RTP
(SRTP) and the MIKEY key management
The OpenScape Media Server can be
installed on the same server as the OpenScape Voice application for systems with
fewer than 5000 subscribers, requiring no
additional hardware; on an external server
(the same server as OpenScape Voice Assistant); on a separate, standalone server.
HiPath 4000 and HiPath 3000 Survivable
Media Gateways (SMG)
In branch offices with a HiPath 4000 or
HiPath 3000, survivability is made possible
through the use of the OpenScape Branch
SIP proxy functionality. This proxy takes the
registrations from the phones and the
HiPath 4000 gateway and passes them to
OpenScape Voice via the WAN. If OpenScape Voice drops out or does not respond
in a timely manner, the local SIP proxy takes
over and tries to mediate the calls, including routing PSTN calls through the
HiPath 4000 gateway. When connectivity
to OpenScape Voice is re-established, the
OpenScape Branch resumes forwarding
the requests to OpenScape Voice as usual.
Interworking of HiPath 4000 with networkwide call pickup is now supported.
RG 8700 Survivable Media Gateway
The RG 8700 provides a complete Siemens
solution for OpenScape Voice, as well as
basic survivability for branch offices in the
event of network failure. Survivability, a
standard feature of the RG 8700 gateway,
is accomplished through the use of SIP
phones that are dual-registered with OpenScape Voice and the RG 8700. If the
RG 8700 can no longer communicate with
OpenScape Voice, it switches to survivable
mode, allowing the dual-registered SIP
phones access to the PSTN trunks and, conversely, allowing incoming calls from the
PSTN to be distributed directly to the SIP
The RG 8700 family of Survivable Media
Gateways comprises 3 models that interwork with OpenScape Voice V4: RG 8716
with up to 16 T1/E1 spans, RG 8708 with
up to 8 T1/E1 spans, and RG 8702 with up
to 2 T1/E1 spans. No license is required.
The RG 8700 V1.3 software adds SIP-Q
functionality for connectivity to
HiPath 4000 and third party products
which support QSIG.
RG 2700 Survivable Media Gateway
The RG 2700 gateway, designed for organizations with a head office and small- to
medium-size branch offices, is used for
cross-site networking. This SMG includes a
built-in SIP proxy that provides continued
inbound and outbound calling service for
up to 30 subscribers when the connection
to the central OpenScape Voice system is
temporarily lost.
Mediatrix Gateways for Small Branch
Offices (SBO)
The OpenScape Branch connects these
locations to the OpenScape Voice and provides the survivability for the small office
scenario. It also supports SIP Trunking functionality and can also interwork with the
gateways from Mediatrix.
Customers can also continue to use their
previously installed third-party SIP gateways with OpenScape Voice. The supported functionality depends on how these
gateways adhere to the relevant SIP standards. Interoperability testing may be
required to confirm feature behavior. The
HiPath Ready Lab is available to vendors
seeking to certify their products with
OpenScape Voice.
OpenScape Branch
The OpenScape Voice V4 solution provides
the following branches:
• OpenScape Branch 50
• OpenScape Branch 250
• OpenScape Branch 1000
• OpenScape Branch 6000
OpenScape Branch 50
This new hardware is being released to support branch scenarios with less than 50
OpenScape Branch 250 and 1000
New with OpenScape Voice V4, is the support of the new OpenScape Branch 250,
and OpenScape Branch 1000 (for up to 250
and 1000 users respectively) where the
remote branch office gets empowered with
remote survivability, SIP Trunking, local
tones, announcements and conference
and Session Border Controller functionalities. During the total loss or partial service
degradation between the remote branch
and the HQ, the OpenScape Branch assures
continued communication services with a
feature rich set of survivable capabilities.
OpenScape Branch 6000
As there is a market demand for OpenScape
Branch configurations larger than the current limit of 1000, OpenScape Branch 6000
has been introduced to support up to 6000
subscribers per branch using either of the
following servers:
• IBM x3550 M2
• Fujitsu Primergy RX330 S1
Session Border
A session border controller (SBC) enables
VoIP networks to extend SIP-based applications beyond an enterprise’s network
boundaries, such as for example, when the
SIP clients of an OpenScape Voice system
reside in different IP networks. For the
branch office location the OpenScape
Branch’s Session Border Controller (SBC)
functionality is also a very efficient and
cost effective solution.
SIP Endpoints
The following Siemens SIP endpoints are
• OpenStage 15/20/20E/40/60/80
• optiPoint 410 S and 420 S
• optiPoint WL2 professional S (wireless)
• OpenScape Personal Edition
OpenStage 40, the flexible office phone, is
customizable for various workplace environments – desk sharers, work teams, call
center staff, and so on.
OpenStage 40
OpenStage 60 incorporates an open application platform and personalization
options, and is especially well-suited for
executive-assistant environments and
users who interact with mobile devices.
OpenStage 60
Selected third-party phones may also be
certified through the Siemens Ready Lab.
OpenStage™ is the name for Siemens' new
generation of IP phones, setting the benchmark for open, unified communications in
a productivity-enhancing business tool.
OpenStage phones have an intuitive and
innovative interface that is available in a
wide variety of languages; all models are
fully compliant with IEEE 802.3af Power
Over Ethernet (PoE) standard.
The OpenStage family of SIP telephones
comprises following models:
OpenStage 15 is a full-featured speakerphone with display and illuminated feature
keys that could be used for up to 8 line
appearances, for example.
OpenStage 80, the high end model, incorporates premium features, materials and
components, and a productivity-enhancing
open platform for applications. It is
designed with the needs of the top-level
manager and executive in mind.
OpenStage 80
OpenStage 15
Eco-Friendly Endpoints
OpenStage 20, the economy model, is a
full-featured speakerphone and a universal
solution for efficient and professional
telephony. The new OpenStage 20 E variant offers open listening only.
OpenStage 20
OpenStage phones have been designed
with the environment in mind. Environmental protection standards have been
fully adhered to in regard to materials and
the manufacturing process, power usage
during operation, and disposal when the
time comes. This new family of devices is
designed to reduce power consumption by
as much as 35%.
optiPoint Phone Family
optiPoint 410 S and 420 S
The feature that distinguishes the optiPoint
410 S / 420 S family of SIP phones, in particular, is the wide range of customizable
models, from the optiPoint 410 entry S for
basic telephony to the optiPoint 420
advance S for high-volume callers with
sophisticated needs. A total of five different telephone models are available to suit
all workstation requirements. A choice of
expansion options and accessories provide
the ability to accommodate future needs.
optiPoint WL2 professional S
The SIP-compliant optiPoint WL2 professional S is a single-line WLAN handset that
supports converged mobile voice and data
applications on a single wireless infrastructure. It is interoperable with all standards
compliant WLAN infrastructure products
for seamless wireless connectivity and
OpenScape Personal Edition
OpenScape Personal Edition is an IP Softphone for installation on portable laptop
and desktop PCs.
The Personal Edition serves as an entry
point into OpenScape UC Application and
can be used as a stepping stone for the subsequent deployment of OpenScape Enterprise Edition.
The new user interface has the look and
feel of Windows® Office 2007 and offers
the user a wide range of technical and
graphical features that can effectively
replace desktop phones entirely. The IP
Softphone thus provides the ideal solution
for normal users who want to eliminate
their desktop phones or mobile users who
are not tied down to a specific workplace
and view their Notebooks as their office.
Analog Adapters
Analog adapters from Mediatrix allow users
with existing analog phones, analog fax
machines and modems to connect to the
OpenScape Voice SIP environment, thereby
preserving their investments.
Other OpenScape UC
Server Applications
OpenScape UC Application
HiPath ComAssistant
The OpenScape Unified Communications
(UC) Application is a high-functionality
collaboration application that fits into an
enterprises’s existing voice and data infrastructure, tying together phones, voice
mail, e-mail, text messaging, directories,
calendaring, instant messaging and
conferencing services.
HiPath ComAssistant is a Web-based call
control and communication filtering
application that enables users to manage
incoming voice and e-mail communications from their desktop.
The tight integration between the
OpenScape UC Application and OpenScape
Voice allows users to take advantage of
market-leading collaboration and mobility
features, and provides the ability to leverage advanced user and group presence features.
HiPath ComAssistant offers computer
telephony integration (CTI) features such
as “click & dial”, call logging, LDAP address
book search, and One Number Service
(ONS). With a choice of two easy-to-use
graphical user interfaces (GUI), HiPath
ComAssistant provides home and business
users with rules-based communication
filters and routing capabilities to optimize
accessibility and increase efficiency.
OpenScape Xpressions
HiPath MetaManagement
OpenScape Xpressions combines voice,
fax, e-mail and text (Short Message Service, SMS) services on a Windows
2003/2008 platform and transforms them
into a Unified Messaging system for use
together with OpenScape Voice.
The HiPath MetaManagement Suite provides a comprehensive and all-embracing
management solution for the standardized
administration of all HiPath platforms and
Built using a modular, scalable client/server
architecture, OpenScape Xpressions can be
configured to meet users’ individual communication needs. New functionality in
OpenScape Xpressions V6 supports Unified
Communication features like Audio- and
Web-Conferencing and the OpenScape
Web Client. OpenScape Xpressions can
alternatively be used as central conferencing system.
HiPath Accounting Management
(HiPath AM) is the accounting application
for processing and analyzing call data for
incoming and outgoing voice and VoIP calls
over different network operators (carriers)
as well as internal connections in HiPath
standalone systems and networks.
HiPath Fault Management (HiPath FM)
supports and simplifies network management by graphically displaying the complete communications network, showing
the status of each element. Special plug-ins
optimize the detection, diagnosis and
removal of failures. HiPath FM also monitors hardware and software from other
manufacturers, interfacing via SNMP
(using the manufacturer-specific enterprise
HiPath User Management provides a simplified “umbrella solution” for the creation,
deletion and modification of user data and
communication resources across all HiPath
platforms and applications in a HiPath network. All relevant user data are stored in a
Directory Service and are available for all
HiPath applications with an LDAP interface.
HiPath Quality of Service (QoS) Management provides comprehensive, easy to use
functions for configuring, monitoring and
analyzing all HiPath VoIP components in a
HiPath network with respect to the relevant
QoS parameters.
OpenScape Contact Center
OpenScape Contact Center is the Siemens
contact center application for the OpenScape Voice and HiPath switching platforms. It provides an intuitive agent interface with powerful visual management
Communications as a
OpenScape Voice Server
Technical Data
Siemens’ Communications as a Service
(CaaS) is much richer than mere hosted
telephony. CaaS offers a modular approach
to building applications, allowing enterprises to select the feature sets they need
today, with the flexibility to change them
or add to them in the future.
The OpenScape Voice V4 software runs on
highly reliable, fault-tolerant industry-standard servers, providing carrier-grade reliability. A typical hardware configuration
consists of a two-node cluster of PRIMERGY
RX330 S1 servers from Fujitsu, running in a
fully redundant load-sharing operation. For
installations with up to 5000 users, redundancy is optional, so the second server is
not required.
The flexibility inherent in CaaS allows customers to not only grow, but to do so at
their own pace. CaaS provides growth
choices ranging from basic telephony to
business process-embedded, presencebased rich communications environments;
from contact centers with remote agents
optimized through group- or skill-based call
routing to multimedia-based and presenceenhanced contact center solutions.
Whether your goal is interoperability with
an existing communications infrastructure
in order to optimize existing investments,
or inexpensive migration to a survivable
remote office, these choices and many
other data center deployment options are
made possible through Siemens’ commitment to open IT-based communications.
The operating system is SUSE Linux Enterprise Server 10 Service Pack 2
(SLES 10 SP2). A SolidTech database runs
on each server.
Each RX 330 S1 server has two (2)
Dual-Core or Quad-Core (Q1 2009) AMD
OpteronTM processors, up to 32 GB of
DDR2-667 direct addressable memory, two
(2) L2/L3 Ethernet switches and eight (8)
10/100/1000 base-T Ethernet links, set up
as pairs connected to the Ethernet switches
(two external L2/L3 Ethernet switches are
required for a redundant configuration.)
Note: OpenScape Voice V4 continues to
support IBM’s System x3650 T servers;
However, customers using the x3650 T
platform must expand the DDR2 memory
to 8 GB.
Supported Standards
IETF Standards
The OpenScape Voice platform and its standard solution components (phones and
application servers) support the relevant
aspects of the following standards specific
to Voice over IP (VoIP):
• RFC 1213: Management Information
Base for Network Management of
TCP/IP-based internets: MIB-II
• RFC 1442: Structure of Management
Information for Version 2 of the Simple
Network Management Protocol
• RFC 3016: RTP Payload Format for
MPEG-4 Audio/Visual Streams
• RFC 3047: RTP Payload Format for ITU-T
Recommendation G.722.1
• RFC 3168: The Addition of Explicit Congestion Notification (ECN) to IP
• RFC 1443: Textual Conventions for Version 2 of the Simple Network Management Protocol (SNMPv2)
• RFC 3204: MIME Type for ISUP and QSIG
• RFC 1889 & RFC 1890: RTP - Real-Time
• RFC 3261: SIP: Session Initiation
• RFC 2131: Dynamic Host Configuration
• RFC 3262: Reliability of Provisional
Responses in SIP
• RFC 2234: Augmented BNF for Syntax
Specifications: ABNF
• RFC 3263: Session Initiation Protocol
(SIP): Locating SIP Servers
• RFC 2246: The TLS Protocol
• RFC 3264: SDP Offer/Answer Model
• RFC 2327: Session Description Protocol
• RFC 3265: SIP-specific Event Notification
• RFC 2474: Definition of the Differentiated Services Field (DS Field) in the IPv4
and IPv6 Headers
• RFC 3267: Real-Time Transport Protocol
(RTP) Payload Format and File Storage
Format for the Adaptive Multi-Rate
(AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio Codecs
• RFC 2475: An Architecture for Differentiated Services
• RFC 2597: Assured Forwarding PHB
• RFC 2705: Media Gateway Control
Protocol (MGCP)
• RFC 2780: IANA Allocation Guidelines
For Values In the Internet Protocol and
Related Headers
• RFC 2806: URLs for Telephone Calls
• RFC 3260: New Terminology and Clarifications for Diffserv
• RFC 3272: Overview and Principles of
Internet Traffic Engineering
• RFC 3288: Using the Simple Object
Access Protocol (SOAP) in Blocks Extensible Exchange Protocol (BEEP)
• RFC 3311: SIP UPDATE Method
• RFC 3323: SIP Privacy Mechanism
• RFC 3515: SIP REFER Method
• RFC 2833: RTP Payload for DTMF Digits,
Telephony Tones and Telephony Signals
• RFC 3605: Real Time Control Protocol
(RTCP) attribute in Session Description
Protocol (SDP)
• RFC 2848: The PINT Service Protocol:
Extensions to SIP and SDP for IP Access
to Telephone Call Services
• RFC 3711: The Secure Real-time Transport Protocol (SRTP)
• RFC 2865: Remote Authentication Dial
In User Service (RADIUS)
• RFC 2976: SIP INFO Method
• RFC 3725: SIP Third Party Call Control
• RFC 3761: The E.164 to Uniform
Resource Identifiers (URI) Dynamic Delegation Discovery System (DDDS) Application (ENUM)
• RFC 3824: Using E.164 Numbers with
• RFC 3830: MIKEY: Multimedia Internet
• RFC 3842: SIP Message Waiting
• RFC 3852: Cryptographic Message Syntax (CMS)
• RFC 3892: The Session Initiation Protocol (SIP) Referred-By Mechanism
• RFC 3952: Real-time Transport Protocol
(RTP) Payload Format for internet Low
Bit Rate Codec (iLBC) Speech
• RFC 3959: The Early Session Disposition
Type for the Session Initiation Protocol
• RFC 4028: Session Timers in SIP
CSTA Standards (ECMA)
• RFC 4049: BinaryTime: An Alternate Format for Representing Date and Time in
• ECMA-269: Services for Computer
Supported Telecommunications
Applications (CSTA) Phase III
• RFC 4235: An INVITE-Initiated Dialog
Event Package for the Session Initiation
Protocol (SIP)
• ECMA-323: XML Protocol for
• RFC 4353: Framework for Conferencing
with the Session Initiation Protocol (SIP)
• RFC 4568: Session Description Protocol
(SDP) Security Descriptions for Media
• ECMA-354: Application Session Services
• ECMA TR/82: Scenarios for
• RFC 4575: A Session Initiation Protocol
(SIP) Event Package for Conference State
• RFC 3960: Early Media and Ringing Tone
Generation in the Session Initiation Protocol (SIP)
Copyright © Siemens Enterprise
Communications GmbH & Co. KG
Siemens Enterprise
Communications GmbH & Co. KG
is a Trademark Licensee of Siemens AG
Hofmannstr. 51, D-80200 München, 12/2009
Reference No.: A31002-H8040-D100-2-7629
The information provided in this document contains
merely general descriptions or characteristics of
performance which in case of actual use do not always
apply as described or which may change as a result of
further development of the products. An obligation to
provide the respective characteristics shall only exist if
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availability. Right of modification reserved.
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