Technology Overview

The Aastra call Manager (MX-One TSE) delivers fully integrated communications by enabling data, voice, and video to be transmitted over a single network infrastructure using standards-based Internet Protocol (IP). Leveraging the framework provided by IP Networks; the MX-One TSE delivers unparalleled performance and capabilities to address current, and emerging communications needs in the enterprise environment. The Aastra communications Manager family of products is designed to optimize feature functionality, reduce configuration and maintenance requirements, and provide interoperability with a wide variety of other applications. The Aastra communications Manager provides this capability while maintaining a high level of availability, quality of service (QoS), and security for your network.
Aastra IP Telephony is a leading converged network IP telephony solution for organizations that want to increase productivity and reduce the costs associated with managing and maintaining separate voice and data networks. The flexibility and sophisticated functionality of the IP network infrastructure provides the framework that permits rapid deployment of emerging applications such as desktop IP Telephony, unified messaging, video telephony, desktop collaboration, enterprise application integration with IP phone displays, and collaborative IP contact centres. These applications enhance productivity and increase enterprise revenues.

Solution Architecture Design

IP Network Infrastructure
IP telephony places strict requirements on IP packet loss, packet delay, and delay variation (or jitter). Therefore, you need to enable Quality of Service (QoS) mechanisms available on IP Telephony ready switches and routers throughout the network. For the same reasons, redundant devices and network links that provide quick convergence after network failures or topology changes are also important to ensure a highly available infrastructure.

Local Area Network Infrastructure
Campus LAN infrastructure design is extremely important for IP telephony operation on a converged network. The LAN infrastructure design requires following basic configuration and design best practices for deploying a highly available network. Further, the LAN infrastructure design requires deploying end-to-end QoS on the network.

Designing a LAN requires building a robust and redundant network from the top down. By structuring the LAN as a layered model and developing the LAN infrastructure one step of the model at a time, you can build a highly available, fault tolerant, and redundant network. Once these layers have been designed you can add network services such as DHCP and TFTP to provide additional network functionality.

Wide Area Network Infrastructure
WAN infrastructure design is extremely important for IP telephony operation on a converged network. Infrastructure design requires following basic configuration and design best practices for deploying a WAN that is as highly available as possible and that provides guaranteed throughput. Furthermore, WAN infrastructure design requires deploying end-to-end QoS on all WAN links.

WAN Topology Design
WAN deployments for voice networks may be hub-and-spoke or an arbitrary topology. A hub-and-spoke topology consists of a central hub site and multiple remote spoke sites connected into the central hub site. In this scenario, each remote or spoke site is one WAN-link hop away from the central or hub site and two WAN-link hops away from all other spoke sites. An arbitrary topology may contain multiple WAN links and any number of hops between the sites. In this scenario there may be many different paths to the same site or there may be different links used for communication with some sites compared to other sites.
Topology-unaware call admission control requires the WAN to be hub-and-spoke, or a spoke-less hub in the case of MPLS VPN. This topology ensures that call admission control, provided by Aastra communications Manager, works properly in keeping track of the bandwidth available between any two sites in the WAN. In addition, multiple hub-and-spoke deployments can be interconnected via WAN links.
Topology-aware call admission control may be used with either hub-and-spoke or an arbitrary WAN topology. This form of call admission control requires parts of the WAN infrastructure to support Resource Reservation Protocol (RSVP).

WAN Quality of Service
Before placing voice, video, and presence traffic on a network, it is important to ensure that there is adequate bandwidth for all required applications. Once this bandwidth has been provisioned, voice priority queuing must be performed on all interfaces. This queuing is required to reduce jitter and possible packet loss if a burst of traffic oversubscribes a buffer. This queuing requirement is similar to the one for the LAN infrastructure.

Typically the WAN requires additional mechanisms such as traffic shaping to ensure that WAN links are not sent more traffic than they can handle, which could cause dropped packets. Link efficiency techniques can be applied to WAN paths. For example, link fragmentation and interleaving (LFI) can be used to prevent small voice packets from being queued behind large data packets, which could lead to unacceptable delays on low-speed links.

The goal of these QoS mechanisms is to ensure reliable, high-quality voice by reducing delay, packet loss, and jitter for the voice traffic.

Aastra Call Manager (MX-One Telephony Server (MX-One TSE)

Aastra MX-ONE™ is a complete vehicle for communications. MX-ONE not only provides a complete telecommunications solution, it is also the tool for Mobility and Unified Communications. MX-ONE is based on open software and hardware environment; standard server and LINUX™ SUSE operating system.
Today’s users are expected to have all types of communications services combined and integrated in the same application GUI. Furthermore users expect to have all services when they are on the move. Since long MX-ONE offers mobile extension where mobile phones are features wise integrated with the office communications system.MX-ONE takes mobility and Unified Communications a step further by combining the two. The combination of mobility and Unified Communications leads to full freedom for the employee as to how and where to perform his duties irrespective of device and location. The management solution of MX-ONE is based on industry standards that an IT department will feel familiar with.

MX-One TSE is call manager that is scalable from 50 to 500 000 extensions, employing a true server-gateway architecture. When using an HP DL360 server (or equivalent), up 15 gateways (3U and\or 7U) can be connected to the server, providing a system capacity of 15000 extensions and 64 E1 or 87 T1 trunks. Further to this, 124 servers with 15 gateways per servers can be logically connected to each other to provide a system with a capacity of 500000 extensions and 7936 E1 or 10788 T1 trunks. Theorically the system can scale to 1860000 extensions, however has only been tested to 500000 extensions.

Over and above all of the traditional MX-One TSW and MD110 features and functionality that has been ported to MX-One TSE, the system has been enhanced with rich new features and functionality, supporting over 500 in total. Listed below are some of the newer capabilities.

Support for SIP on both the user and trunk sides
Due to open standards being used in this system, SIP support is extended to 3rd SIP compliant phones, wired or WLAN.

SIP DECT allows for connection of base stations via the IP network infrastructure thus removing the need for additional cabling to support a wireless telecommunication requirement

HLR/VLR redundancy
IP/SIP extension implementation in MX-ONE is desig¬ned in accordance with the HLR/VLR architecture used in mobile networks. An IP user has a “home server” that corresponds to the HLR (Home Location Register) in mo¬bile networks. The user can be handled by any server in the system as long as the home server can be accessed. A VLR (Visitor Location Register) is created in the visited server and part of the user data is copied from the HLR to the VLR.The Gatekeeper/SIP Proxy database redundancy feature in MX-ONE allows an IP user to register with any available server in the network and the IP user can be reached by incoming calls even if the “home server” is out of service.

Server bonding redundancy
By server bonding two or more Ethernet interfaces look like one logical interface to the MX-ONE server, all in order to im¬prove availability and performance. Thanks to this method, MX-ONE offers a higher level of reliability. In the case one in¬terface or switch fails the other one takes over.

CSTA V3 – XML support
The latest version of MX-ONE supports CTI monitoring in accordance with Computer-Supported Telecommunications Applications 3, also called CSTA Phase 3. The CSTA Phase 3 is based on the ECMA-269 standard. The existing CSTA Phase 1/TSAPI implementation is also supported, as was the case in previous releases. Further to this is now possible to obtain CTI information directly from MX-One Call Manager, with out the need for for application link or open application server, although these methods are still supported. Important to is the either metod can be deployed not both.

Dual Mode WLAN/Cellular 3G/GSM
Dual mode will integrate your mobile workforce as IP extensions towards MX-ONE. The Dual mode solution, can seamlessly handover active calls between Cellular (GSM/3G) and VoIP WiFi networks.

Signalling and media encryption is supported via TLS and SRTP and Communications to the PBX is suecured via the use of HTTPS and SSH

Hospitality application
The MX-ONE basic package can be integrated with a Hospitality Media Gateway from 3rd party vendors to form a complete hospitality solution packed with features to handle all customers’ voice and data needs. The solution comes with industry standards based interfaces, enabling it to be fully integrated with a front-office system, making guest information available to those who need it. This hospitality solution may be tailored to fit the needs of hotels, hospitals, cruise ships, conference centers and exhibitions, as well as other customers wanting to offer this type of functionality (such as university campuses).

Microsoft OCS and IBM Sametime Integration
MX-ONE also offers integration with Microsoft® OCS (Office Communications Server). An extension in MX-ONE can be fully visible on OCS clients, and feature-wise, fully integrated. The integration with OCS is based on the Microsoft “Dual Forking” standard. Other metods employed to facilte Microsoft OCS intergration is RCC (remote call control) ie the MOC clients uses the phone to facilate call functions and direct SIP connection ie SIP trunk established between OCS server and call Manager and ext exists in both call manager and OCS server

MX-ONE™ Telephony System Building Blocks
The MX-ONE Telephony System consists of two basic components: the server and the media gateway. A server and a media gateway can be combined to form either a complete system or a LIM that is included in a large multi-LIM system. So as to always provide cost-efficient solutions, several hardware options have been developed. Both the server and the media gateway are available with a variety of options.

Aastra Dialogue 5446 IP Phone

Dialog 4425 – IP Vision

Dialog 4222 Office

Aastra 6730i

Aastra 6731i

Aastra 6753i

Aastra 6755i

Aastra 6757i

SIP 6739i Pone

Polycom Sound Station 7000

Media Gateway Classic
Aastra Media Gateway Unit

3U Media Gateway Lite

Aastra Server Unit (ASU-E)

Commercially available hardware (HP DL360 or equivalent)

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