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SME VoIP System Guide Installation & Configuration Network Deployment Operation & Management Technical Reference Document Version 1.5 Dec-2010 RTX Telecom A/S, Denmark t n e m u c o D s i h T t u o b A
r e t p a h C 1 SME VoIP System Guide, Version 1.5 Proprietary and Confidential Trademarks RTX and the combinations of its logo thereof are trademarks of RTX Telecom A/S, Denmark. Other product names used in this publication are for identification purposes and maybe the trademarks of their respective companies. Disclaimer The contents of this document are provided in connection with RTX products. RTX makes no representations with respect to completeness or accuracy of the contents of this publication and reserves the right to make changes to product descriptions, usage, etc., at any time without notice. No license, whether express, implied, to any intellectual property rights are granted by this publication Confidentiality This document should be regarded as confidential, unauthorized copying is not allowed Dec-2010 RTX Telecom A/S, Denmark, All rights reserved http://www.rtx.dk t n e m u c o D s i h T t u o b A
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 2 Contents 2 SME VoIP System Guide .................................................................................................................................... 1 Contents ............................................................................................................................................................ 3 About This Document ................................................................................................................................ 8 1 1.1 Audience .................................................................................................................................................. 8 1.2 When Should I Read This Guide .............................................................................................................. 8 1.3 Important Assumptions ........................................................................................................................... 8 1.4 Whats Inside This Guide ......................................................................................................................... 8 1.5 Whats Not in This guide.......................................................................................................................... 9 1.6 Abbreviations ......................................................................................................................................... 10 1.7 References/Related Documentation ..................................................................................................... 10 1.8 Document History .................................................................................................................................. 10 1.9 Documentation Feedback ..................................................................................................................... 10 Introduction System Overview ............................................................................................................. 11 2.1 Hardware Setup ..................................................................................................................................... 11 2.2 Components of SME VoIP System ......................................................................................................... 12 RTX Base Stations ........................................................................................................................ 12 2.2.1 2.2.2 SME VoIP Administration Server/Software ................................................................................. 12 2.2.3 RTX Wireless Handset .................................................................................................................. 13 2.3 Wireless Bands ...................................................................................................................................... 13 2.4 System Capacity (in Summary) .............................................................................................................. 13 2.5 Advantages of SME VoIP System ........................................................................................................... 14 SME System Deployment Scenarios ........................................................................................................ 15 3.1 Multi-cell System ................................................................................................................................... 15 3.2 Multi-cell Setup ..................................................................................................................................... 15 3.3 Case Studies ........................................................................................................................................... 16 Case ##1: Isolated Buildings ........................................................................................................ 16 3.3.1 3.3.2 Case ##2: Location with co-located partners .............................................................................. 17 Case ##3: Large to Medium Sized Enterprises ............................................................................ 18 3.3.3 3.3.4 Case ##4: Large Enterprises at Different Locations ..................................................................... 18 SME VoIP Network Planning/Optimization ............................................................................................. 20 4.1 Network Requirements ......................................................................................................................... 20 4.2 Deployment Considerations .................................................................................................................. 20 4.3 Site Planning .......................................................................................................................................... 20 Deployment kit ............................................................................................................................ 20 4.3.1 Location Probability ..................................................................................................................... 20 4.3.2 4.3.3 Handover Mechanics/Planning ................................................................................................... 21 4.4 Cell Coverage / Capacity Planning ......................................................................................................... 21 Cell Coverage ............................................................................................................................... 21 4.4.1 4.4.2 Capacity Planning ........................................................................................................................ 22 4.5 Network Dimensioning .......................................................................................................................... 23 4.6 Environmental Considerations .............................................................................................................. 23 3 4 t n e m u c o D s i h T t u o b A
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 3 9 8 5 6 4.7 Recommended Base station/Repeater Placement ............................................................................... 24 4.8 Network Assessment/Optimisation ...................................................................................................... 25 Deployment Mechanics Multi-cell SME Network ................................................................................. 26 Installation of Base Stations/Repeater .................................................................................................... 29 6.1 Package - Contents/Damage Inspection................................................................................................ 29 6.2 RTX Base station Mechanics .................................................................................................................. 30 6.3 RTX Base Unit - Reset feature ................................................................................................................ 31 6.4 Installing the Base Station ..................................................................................................................... 31 7 Making Handset Ready ............................................................................................................................ 33 7.1 Package - Contents/Damage Inspection................................................................................................ 33 7.2 Before Using the Phone ......................................................................................................................... 34 7.3 Using the Handset ................................................................................................................................. 35 Core Network Server(s) Configuration .................................................................................................... 36 8.1 Server setup ........................................................................................................................................... 36 8.2 Requirements ........................................................................................................................................ 36 8.3 DNS Server Installation/Setup ............................................................................................................... 36 8.4 DHCP Server Setup ................................................................................................................................ 36 8.4.1 Hint: Getting DHCP Server to Work ............................................................................................. 37 8.5 TFTP Server Setup .................................................................................................................................. 38 8.5.1 TFTP Server Settings .................................................................................................................... 38 8.6 SIP Server Setup ..................................................................................................................................... 39 8.6.1 FreePBX SIP Server....................................................................................................................... 39 SME VoIP Administration Interface ......................................................................................................... 42 9.1 Home/Status .......................................................................................................................................... 42 9.2 Extensions .............................................................................................................................................. 43 9.3 Servers ................................................................................................................................................... 45 9.4 Network ................................................................................................................................................. 46 9.4.1 IP Settings .................................................................................................................................... 46 VLAN Settings .............................................................................................................................. 47 9.4.2 Boot Server Options .................................................................................................................... 48 9.4.3 9.4.4 NAT Settings ................................................................................................................................ 49 9.4.5 SIP/RTP Settings ........................................................................................................................... 49 9.5 Management Settings Definitions ......................................................................................................... 50 9.6 Firmware Update Definitions ................................................................................................................ 51 9.7 Time Server ............................................................................................................................................ 52 9.8 Multi-cell Parameter Definitions ........................................................................................................... 53 Settings for Base Unit .................................................................................................................. 53 9.8.1 9.8.2 DECT System Settings .................................................................................................................. 54 SIP System Settings ...................................................................................................................... 55 9.8.3 9.8.4 MAC-units in Chain ...................................................................................................................... 55 9.9 Settings Configuration File Setup ....................................................................................................... 56 9.10 Debug Logs .......................................................................................................................................... 57 9.11 SIP Logs ................................................................................................................................................ 58 Firmware Upgrade Management ........................................................................................................ 59 10.1 Network Dimensioning ........................................................................................................................ 59 10.2 TFTP Configuration .............................................................................................................................. 60 10 t n e m u c o D s i h T t u o b A
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 4 11 12 13 14 15 10.3 Create Firmware Directories ............................................................................................................... 61 10.4 Login to Base SME Configuration Interface ......................................................................................... 61 10.5 Firmware Update Settings ................................................................................................................... 63 10.6 Base Station(s) Firmware Upgrade ...................................................................................................... 65 10.7 Handset (s) Firmware Upgrade............................................................................................................ 66 10.8 Verification of Firmware Upgrade ....................................................................................................... 66 10.9 Reboot the Base station(s) .................................................................................................................. 68 Registration Management - Handset .................................................................................................. 70 11.1 Hardware required .............................................................................................................................. 70 11.2 Software required equipment ............................................................................................................. 71 11.3 Add server............................................................................................................................................ 71 11.4 Register handset to base ..................................................................................................................... 73 VLAN Setup Management ................................................................................................................... 77 12.1 Introduction ......................................................................................................................................... 77 12.2 Backbone/ VLAN Aware Switches ....................................................................................................... 78 12.3 How VLAN Switch Work: VLAN Tagging .............................................................................................. 79 12.4 Implementation Cases ......................................................................................................................... 79 12.5 Base station Setup ............................................................................................................................... 80 12.6 Configure Time Server ......................................................................................................................... 81 12.7 VLAN Setup: Base station .................................................................................................................... 82 Multi-cell Setup & Management ......................................................................................................... 83 13.1 SME Configuration Interface ............................................................................................................... 83 13.2 Adding Base stations via SME Configuration Interface ....................................................................... 83 13.2.1 Time Server Setup ........................................................................................................................ 85 13.2.2 SIP Server (or PBX Server) Setup ................................................................................................. 86 13.2.3 Multi-cell Setup ........................................................................................................................... 87 13.3 Synchronizing the Base stations .......................................................................................................... 90 13.4 Summary of Procedure Creating a Chain ......................................................................................... 92 13.5 Stage 1 ................................................................................................................................................. 92 13.6 Stage 2 ................................................................................................................................................. 92 13.7 Stage 3 ................................................................................................................................................. 93 13.8 Stage 4 ................................................................................................................................................. 94 13.9 Practical Configuration of Multi-cell System ....................................................................................... 95 13.9.1 Case ##1: Isolated Buildings ........................................................................................................ 95 Case ##2: Location with co-located partners .............................................................................. 96 13.9.2 Case ##3: Large to Medium Sized Enterprises ............................................................................ 97 13.9.3 13.9.4 Case ##4: Large Enterprises at Different Locations ..................................................................... 98 Functionality Overview ...................................................................................................................... 100 14.1 System Feature List............................................................................................................................ 100 14.2 Detail Feature Description ................................................................................................................. 101 Network Operations .......................................................................................................................... 104 15.1 Introduction ....................................................................................................................................... 104 15.2 System Start Up ................................................................................................................................. 104 15.3 Terminal Attachment......................................................................................................................... 104 15.4 Outgoing Calls .................................................................................................................................... 104 15.5 Incoming Calls 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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 5 17 16 15.6 Handover ........................................................................................................................................... 105 15.6.1 RTP Stream Remains at Initial Base Station ............................................................................... 105 15.7 Roaming ............................................................................................................................................. 106 Operation Setup Bases/Handsets/SIP Sever .................................................................................. 108 16.1 Power Up ........................................................................................................................................... 108 16.2 Power Down ...................................................................................................................................... 108 16.3 Call Operations .................................................................................................................................. 109 16.3.1 Initiating Calls ............................................................................................................................ 109 Call Holding ................................................................................................................................ 110 16.3.2 Call Transfer (Blind) ................................................................................................................... 111 16.3.3 16.3.4 Call Bridging (Attended Transfer) .............................................................................................. 111 Call Conference (Conference) .................................................................................................... 112 16.3.5 Handset - Service Menu Management .............................................................................................. 113 17.1 Service Menu Site Survey Mode ..................................................................................................... 113 17.2 Service Menu Parameter Definitions ................................................................................................ 114 17.2.1 Master Reset .............................................................................................................................. 114 17.2.2 Site Survey Mode ....................................................................................................................... 114 17.2.3 HS Logs ....................................................................................................................................... 115 Status ......................................................................................................................................... 116 17.2.4 IPEI ............................................................................................................................................. 116 17.2.5 17.2.6 Demo mode ............................................................................................................................... 117 Appendix ........................................................................................................................................................ 118 Appendix A......................................................................................................................................... 119 18 Handset...................................................................................................................................................... 119 Base Station ............................................................................................................................................... 120 Web interface ............................................................................................................................................ 120 Charge unit ................................................................................................................................................ 121 SME VoIP System Guide, Version 1.5 Proprietary and Confidential t n e m u c o D s i h T t u o b A
r e t p a h C 6 SME VoIP System Guide, Version 1.5 Proprietary and Confidential t n e m u c o D s i h T t u o b A
r e t p a h C 7 1 About This Document This document describes the configuration, customization, management, operation, maintenance and trouble shooting of the SME VoIP System. It also describes effective practices that should be done to deploy an optimal SME System. 1.1 Audience Who should read this guide? First, this guide is intended for networking professionals responsible for designing and implementing RTX based enterprise networks. Second, network administrators and IT support personnel that need to install, configure, maintain and monitor elements in a live SME VoIP network will find this document helpful. Furthermore, anyone who wishes to gain knowledge on fundamental features in the UMBER system can also benefit from this material. 1.2 When Should I Read This Guide Read this guide before you install the core network devices of VoIP SME System and when you are ready to setup or configure SIP server, NAT aware router, advanced VLAN settings, base stations, and multi cell setup. This manual will enable you to set up components in your network to communicate with each other and also deploy a fully functionally VoIP SME System. 1.3 Important Assumptions This document was written with the following assumptions in mind:
1) You have understanding of network deployment in general 2) You have working knowledge of basic TCP/IP/SIP protocols, Network Address Translation, etc.. 3) A proper site survey has been performed, and the administrator have access to these plans 1.4 Whats Inside This Guide We summarize the contents of this document in the table below:
Where Is It?
Chapter 2 Purpose Content Chapter 3 Chapter 4 Introduction to the SME VoIP Network Description of System Deployment Scenarios SME VoIP Network Planning/Optimization Chapter 5 Deployment Mechanics Multi-cell SME VoIP System Guide, Version 1.5 Proprietary and Confidential To gain knowledge about the different elements in a typical SME VoIP Network Provides the reader an idea of different possibilities available to the user in deploying the system. To learn radio network planning techniques including dimensioning, detailed capacity, coverage planning and network optimisation Examine practical cases of how Multi-cell SME VoIP network can be deployed t n e m u c o D s i h T t u o b A
r e t p a h C 8 Chapter 6 SME Network Installation of Base station/Repeater Chapter 7 Making Handsets Chapter 8 Chapter 9 Chapter 11 Ready Core Network Servers Configuration SME VoIP Administration Interface Firmware Upgrade/Downgrade Management Chapter 12 Registration Management -
Handsets Considerations to remember before unwrapping and installing base units and repeaters To determine precautions to take in preparing handsets for use in the system To learn about operating the handset and base stations including detail description of handset MMI. To learn about the Configuration Interface and define full meaning of various parameters needed to be setup in the system. Provides a procedure of how to upgrade firmware to base stations and/or handsets Learn how to add servers via the Configuration Interface and how to register handset to base stations Chapter 13 VLAN Setup Examines how to setup VLAN in the SME network Management Chapter 14 Multi-Cell Setup &
Learn how to setup multiple bases into a multi-cell network Chapter 15 Management System Functionality Overview Chapter 16 Network Operations Chapter 17 Operations Setup Handset/Base/SIP Server To gain detail knowledge about the system features. To study the operation of network elements during system start up, location registration, etc. To study the operation of handset and base stations/SIP server. Chapter 18 Handset Service To learn how to enable and use the hidden menus in the handset Menu Management 1.5 Whats Not in This guide This guide provides overview material on network deployment, how-to procedures, and configuration examples that will enable you to begin configuring your VoIP SME System. It is not intended as a comprehensive reference to all detail and specific steps on how to configure other vendor specific components/devices needed to make the SME VoIP System functional. For such a reference to vendor specific devices, please contact the respective vendor documentation. t n e m u c o D s i h T t u o b A
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 9 1.6 Abbreviations For the purpose of this document, the following abbreviations hold:
DHCP:
DNS:
HTTP(S):
(T)FTP:
IOS:
PCMA:
PCMU:
PoE:
RTP:
RPORT:
SIP:
SME:
VLAN:
TOS:
URL:
UA:
Dynamic Host Configuration Protocol Domain Name Server Hyper Text Transfer Protocol (Secure)
(Trivial) File Transfer Protocol Internetworking Operating System A-law Pulse Code Modulation mu-law Pulse Code Modulation Power over Ethernet Real-time Transport Protocol Response Port (Refer to RFC3581 for details) Session Initiation Protocol Small and Medium scale Enterprise Virtual Local Access Network Type of Service (policy based routing) Uniform Resource Locator User Agent 1.7 References/Related Documentation
[1]:
[2]:
[3]:
[4]:
Deployment Kit Guide Version 2.5 Hosted PBX Solution - Deployment Kit Version 0.5 Handset operation Manual V0.1 1.8 Document History Revision 1.5 1.4 Issue Date 17-Dec-2010 12-Nov-2010 Author MYA MYA MYA MYA MYA 27-Oct-2010 05-Oct-2010 23-Sep-2010 1.1 1.0 0.1 Comments Complete review and modifications of all sections New Input: 17.3 Call Operations, Operations Setup Handset/Base/SIP Server. Total re-write of overall of manual First version, SIP version, EU DECT Initial Version 1.9 Documentation Feedback We always strive to produce the best and we also value your comments and suggestions about our documentation. If you have any comments about this guide, please enter them through the Feedback link on the RTX Telecom website. We will use your feedback to improve the documentation. t n e m u c o D s i h T t u o b A
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 10 Introduction System Overview 2 In a typical telephony system, the network setup is the interconnection between Base-stations, fat routers, repeaters, portable parts, etc. The back-bone of the network depends on the deployment scenario but a ring or hub topology is used. The network has centralized monitoring, and maintenance system. The system is easy to scale up and supports from 1 to 40 bases in the same network. Further it is able to support up to 200 registered handsets. The Small and Medium Scale Enterprise (SME) VoIP system setup is illustrated below. Based on PoE interface each base station is easy to install without additional wires other than the LAN cable. The system supports the next generation IP DECT CAT-IQ repeater with support up to 5 channels simultaneous call sessions. The following figure gives a graphical overview of the architecture of the SME VoIP System:
2.1 Hardware Setup SME network hardware setup can deployed as follows:
Base-station(s) are connected via Layer 3 and/or VLAN Aware Router depending on the deployment requirements. The Layer 3 router implements the switching function. The base-stations are mounted on walls or lamp poles so that each base-station is separated from each other by up to 50m indoor (300m outdoor). Radio coverage can extended to 400m using repeaters that are installed up to 300m from the base-station(s). The base-station antenna mechanism is based on space diversity feature which improves coverage. The base-stations uses complete DECT MAC protocol layer and IP media stream audio encoding feature to provide up to 10 simultaneous calls. w e i v r e v O m e t s y S n o i t c u d o r t n SME VoIP System Guide, Version 1.5 Proprietary and Confidential 11 I
r e t p a h C 2.2 Components of SME VoIP System RTX SME VoIP system is made up of (but not limited to) the following components:
At least one RTX Base Station is connected over an IP network and using DECT as air-core interface. RTX IP DECT wireless Handset. RTX SME VoIP Configuration Interface; is a management interface for SME VoIP Wireless Solution. It runs on all IP DECT Base stations. Each Base station has its own unique settings. Other 3rd party vendor products which will be described in details in Chapter 8 2.2.1 RTX Base Stations The Base Station converts IP protocol to DECT protocol and transmits the traffic to and from the end-nodes
(i.e. wireless handsets) over a channel. It has 12 available channels. In a multi-cell setup, each base station has:
8 channels have associated DSP resources for media streams. The remaining 4 channels are reserved for control signalling between IP Base Stations and the SIP/DECT end nodes (or phones). Base Stations are grouped into clusters. Within each Cluster, Base Stations are synchronized to enable a seamless handover when a user moves from one base station coverage to another. For synchronization purposes, it is not necessary for Base Stations to communicate directly with each other in the system. E.g. a Base Station may only need to communicate with the next in the chain. It is advisable for a Base Station to identify more than one Base Station to guarantee synchronization in the situation that one of the Base Stations fails. The 4 control signalling channels are used to carry bearer signals that enable a handset to initiate a handover process. 2.2.2 SME VoIP Administration Server/Software This server is referred to as SME VoIP Configuration Interface. The SME VoIP Configuration Interface is a web based administration page used for configuration and programming of the base station and relevant network end-nodes. E.g. handsets can be registered or de-
registered from the system using this interface. The configuration interface can be used as a setup tool for software or firmware download to base stations, repeaters and handsets. Further, It is used to check relevant system logs that can be useful to w e i v r e v O m e t s y S n o i t c u d o r t n I
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 12 administrator. These logs can be used to troubleshoot the system when the system faces unforeseen operational issues. 2.2.3 RTX Wireless Handset The handset is a lightweight, ergonomically and portable unit compatible with Wideband Audio (G.722), DECT, GAP standard, CAT-iq vb & ve-profile compliant. The handset includes Colour display with graphical user interface. It can also provide the subscriber with most of the features available for a wired phone, in addition to its roaming and handover capabilities. Refer to the relevant handset manual for full details handset features. 2.3 Wireless Bands The bands supported in the SME VoIP are summarized as follows:
Frequency band:
1880 1930 MHz (DECT) 1880 1900 MHz (10 carriers) Europe/ETSI 1910 1930 MHz (10 carriers) LATAM 1920 1930 MHz (5 carriers) US Other frequency bands can be customized via Base station low-level debug programming. 2.4 System Capacity (in Summary) SME network capacity of relevant components can be summarised as follows:
Description Min ## of Bases Single Cell Setup Max ## of Bases in Multi-cell Setup Single Cell Setup: Max ## of Repeaters Multi-cell Setup: Max ## of Repeaters Max ## of Users per Base Capacity 1 40 3 per Base station 3x40=120 30 Max ## of Users per SME VoIP System 30x40=1200 (limited to 200) Multi-cell Setup: Max ## of Synchronisation levels Multi-cell Setup: Max ## of Users Max ## Simultaneous Calls (Single Cell) Max ## Simultaneous Calls (Multi-cell Setup) 6 8 10 per Base station 8x40=320 call sessions SME telephony network composed of one base station Telephony network that consists of more than one base stations Is the air core interface between two base stations. Quick Definitions-
Single Cell Setup:
Multi-cell Setup:
Synchronisation Level:
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r e t p a h C w e i v r e v O m e t s y S n o i t c u d o r t n 2.5 Advantages of SME VoIP System They include (but not limited to):
1. Simplicity. Integrating functionalities leads to reduced maintenance and troubleshooting, and significant cost reductions. 2. Flexibility. Single network architecture can be employed and managed. Furthermore, the architecture is amenable to different deployment scenarios, including Isolated buildings for in-building coverage, location with co-located partners, and large to medium scale enterprises deployment for wide coverage. 3. Scalability. SME network architecture can easily be scaled to the required size depending on customer requirement. 4. Performance. The integration of different network functionalities leads to the collapse of the protocol stack in a single network element and thereby eliminates transmission delays between network elements and reduces the call setup time and packet fragmentation and aggregation delays. SME VoIP System Guide, Version 1.5 Proprietary and Confidential w e i v r e v O m e t s y S n o i t c u d o r t n I
r e t p a h C 14 3 SME System Deployment Scenarios In principle, there is no one best solution for deployment of SME Telephony network. Different solutions
(i.e. deployments scenarios) exist depending on the customer requirements. Before describing commonly used cases in SME System deployment, we first describe a multi-cell system and its configuration. 3.1 Multi-cell System A multi-cell system has a coordinated installation of intra-system synchronized base stations, which enables seamless inter-cell handover for moving handsets. It provides, besides the increased mobility, higher total capacity and higher quality than a corresponding number of standalone base stations. This is due to a combination of the inter-cell synchronization and the seamless handover features of a multi-cell system. All handsets subscribed to a multi-cell system will benefit from high voice quality, full coverage and full mobility over the whole multi-cell system coverage area. 3.2 Multi-cell Setup A total of 40 Base stations can be hosted in a typical SME multi-cell system. At each time, a maximum of 6 levels can be formed in a multi-cell chain. Synchronisation Level is the air core interface between two base stations. Using the figure below, synchronisation level 1 is when Base unit 1 synchronizes to Master Base unit, and Level 2 is when Base unit 2 uses Base unit 1 as its synchronization source, in that order. The relevant synchronisation levels can be defined by specifying the hierarchy of bases in multi-cell systems. The hierarchy of bases can be specified using the DECT sync source parameter in the SME VoIP Configuration Server (Refer to Chapter 13 for detail procedure of Multi-cell setup). s o i r a n e c S t n e m y o p e D m e t s y S E M S l SME VoIP System Guide, Version 1.5 Proprietary and Confidential 15
r e t p a h C 3.3 Case Studies The following requirements must be considered when deploying SME Telephony Network:
The type of end users (e.g. Business type, Educational Campus, etc.) Distances end users require Vendors Business Situation In this section we describe common cases or situations where SME Telephony solutions can be implemented. Typical case studies includes (but not limited to):
3.3.1 Case ##1: Isolated Buildings This is the simplest scenario a SME VoIP System can be deployed in e.g. branch or small office, Retail outlet or store. It consists of a standalone base station and a number of handsets registered to it. This setup is optimal for isolated buildings. A typical illustration is shown in the figure below. From this illustration, it is not possible to roam or handover to other bases in this setup. s o i r a n e c S t n e m y o p e D m e t s y S E M S l SME VoIP System Guide, Version 1.5 Proprietary and Confidential 16
r e t p a h C 3.3.2 Case ##2: Location with co-located partners This scenario can be deployed in e.g. Department shops, Retail location with co-located photo kiosk or pharmacy and huge apartment complexes. This setup consists of more than one base station and each handset is registered to a specific base. Roaming and handovers are permitted in this setup. Here is diagram to illustrate Case ##2. In this illustration, two slave bases synchronises to a Master DECT source. s o i r a n e c S t n e m y o p e D m e t s y S E M S l 17
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 3.3.3 Case ##3: Large to Medium Sized Enterprises This scenario is a multi-cell setup can be deployed in e.g. Corporate headquarters, Harbour areas, High School Campus. In the illustration shown below, handsets can roam and handover to other bases. 3.3.4 Case ##4: Large Enterprises at Different Locations In this scenario, multi-cell systems are deployed at different locations; geographically separated from each other. An example of this setup includes Government departments/agencies geographically separated from each other; University campuses or Hospital(s) in different regions. Roaming and handovers are permitted in this setup even though the each multi-cell system is geographically separated from each other. Each base station has a unique identifier. Handsets registered to one multi-cell setup can be used in other geographically separated setups because visited base stations retrieves the base station identifier which the visiting handset is originally registered to. In addition, the relevant SIP server functions must be enabled to so roaming and handover can be allowed at different geographies. s o i r a n e c S t n e m y o p e D m e t s y S E M S l SME VoIP System Guide, Version 1.5 Proprietary and Confidential 18
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r e t p a h C 4 SME VoIP Network Planning/Optimization In this chapter, we describe SME VoIP radio network planning techniques including dimensioning, detailed capacity and coverage planning, and network optimisation. 4.1 Network Requirements Network requirement is essential to determine elements necessary to achieve the overall expectations of the customer. Typical network requirements includes (but not limited to):
The geographical area to be covered The type or architecture of building and/or topology, etc. The estimated traffic on each zone or region or building The blocking criteria in each traffic area. The relevant quality targets expected to be achieved 4.2 Deployment Considerations The following radio considerations must be examined before deploying a SME VoIP System. These includes
(but not limited to):
Building Penetration:
When a signal strikes a building it is diffracted or absorbed; therefore to some extend the signal is reduced. The amount of absorption is dependent of the kind of building and its environment, the amount of solid structure. This is an important consideration in coverage planning. Interference Sources:
Signals to receiving antenna can be weakened by virtue of interference from other signals. These signals may be from the same network or other man-made objects. Interference sources must be identified and avoided or minimized. 4.3 Site Planning 4.3.1 Deployment kit Based on propagation models, the coverage of areas is done with the use of radio planning tools. In the RTX SME VoIP Network, the radio planning tool available is called Deployment Kit. Detail description and use of this document is available in a separate document(s) [1][2]. 4.3.2 Location Probability The quality of coverage is determined by location probability. For practical purposes, the location probability of 50% is equal to the sensitivity of receiver in a specific region. This is also measured by the Deployment Kit [1][2]. n o i t a z i m i l i t p O
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 20 4.3.3 Handover Mechanics/Planning Handsets should seamlessly move between coverage areas. In other words, handset should be able to move in a multi-cell setup of base stations and/or repeaters from one base station to another without terminating or causing hindrance while receiving continuous service and maintaining call-sessions in progress. For efficient handover of conversations between Base stations in multi-cell setup, deploy Base stations with wide overlap between them (i.e., plan for some areas to be covered by more than one Base station). Overlaps are necessary to maintain seamless handover and to establish synchronization chains. A good example may be a cafeteria during lunch hour where temporary concentrations of handsets may occur. The overlap carries the excess call load to adjacent Base stations to provide uninterrupted services to subscribers. 4.4 Cell Coverage / Capacity Planning 4.4.1 Cell Coverage Due to the unexpected nature of RF propagation in an indoor environment, an actual on-site test must be performed before the deploying and/or installing core network elements. An extensive guide to effective RF coverage planning is outside the scope of this document. This should be noted:
The Base stations provides typical RF coverage of up to 50 meters/164 feet in a typical indoor office environment and up 300 meters/984 feet in an open area (line-of-sight-LOS), extending in all directions from the Base stations (i.e. Omni directional). The exact coverage range depends on the building architecture, wall material and surroundings. The figure below show the correct installation of base stations spaced at the recommended distances from each other:
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r e t p a h C 21 Typically, installations such as office buildings, hotels and hospitals should be equipped with both base stations and repeaters on several floors to create uniform and complete radio coverage. Open areas can be covered with a sparse network of base stations. In such deployments, the base stations and/or repeaters cover an extended range due to the extended line-of-sight radio propagation capability The figure below shows an example of an installation where base stations are not spaced at the right distances:
4.4.2 Capacity Planning This is where the Network provider estimates how many calls will be initiated in a typical window/time frame and how many users will be initiating them. Another aspect of capacity planning should address the user growth pattern of a typical SME VoIP network. How many users will be using this network in for example two years window, etc. Based on these estimations, the network dimensioning can be planned orderly bearing in mind the SME capacity. The capacity of the SME VoIP System is summarised as follows (this should guide the network planner):
Description Min ## of Bases Single Cell Setup Max ## of Bases in Multi-cell Setup Single Cell Setup: Max ## of Repeaters Multi-cell Setup: Max ## of Repeaters Max ## of Users per Base Max ## of Users per SME VoIP System Multi-cell Setup: Max ## of Synchronisation levels Capacity 1 40 3 per Base station 3x40=120 30 30x40=1200 6 n o i t a z i m i l i t p O
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 22 Multi-cell Setup: Max ## of Users Max ## Simultaneous Calls (Single Cell) Max ## Simultaneous Calls (Multi-cell Setup) 8 10 per Base station 8x40=320 call sessions 4.5 Network Dimensioning After the network requirements are clearly defined, the number of users that are expected to use the network must be estimated. Based on that, you should estimate and identify the number and type of equipments required in order to cater for the capacity, coverage and quality requirements. The more accurate the dimensioning, the more efficient the network rollout. Handsets/SIP End-Nodes:
In a typical setup, the system can support up to 200 handsets depending on the configuration. Base stations/Repeaters:
The system can easily scale up to 40 base stations. Depending on the network setup, coverage can be extended by up to 5 repeaters. The planner should bear in mind that base stations can support 10 simultaneously call sessions while repeaters can support 5 call sessions. Core Network Equipments:
These equipments are at the premises of the service operator or data center. Depending on the network requirements the following devices should be available:
VLAN/NAT aware router(s), Session Border Controller, DHCP/TFTP/FTP Servers, STUN Server, Media Server, Access Gateway, SIP Server, etc. 4.6 Environmental Considerations In this section, we enumerate some environmental conditions that need to be considered prior to planning, deployment and optimisation of the SME network. The considerations are as follows:
Ensure that the installation area is clean, dry, and protected from weather extremes. Ensure that the floor of the installation area is finished with linoleum, vinyl, ceramic, wooden flooring, computer floor tiles, or polished sealed concrete. Ensure that the ceiling of the installation area is finished or treated to prevent particle discharge. Ensure that the installation area is well lit, and that the light source is uniformly diffused without shadows. Adequate lighting should provide a comfortable reading level and allow the identification of wire insulator colours without undue eye fatigue. Lighting should be comparable to an office work environment, with a minimum level of 21 meter/68.9 feet at each work surface. As a rule of thumb, in a room with a 2.5 meters/8.2 feet ceiling, one 1.2 meters/4 feet fluorescent tube provides sufficient illumination for 1.9-2.4 square meters/20.5-25.9 square feet. Ensure that ventilation of the installation area is capable of maintaining an ambient temperature of 0-
40C/32-104F, and a relative humidity of 20-80% non-condensing, while the system is operating. The maximum power rating of a base station under full load should not exceed 315W/1070 BTU/Hr. These figures are for each cabinet only, and do not take into account heat generated by other equipment. In particular, charging fully-discharged batteries may generate a considerable amount of heat, depending on battery capacity and rate of charge. Refer to the equipment manufacturer data for more information. n o i t a z i m i l i t p O
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 23 Ensure that the installation area is free of caustic or corrosive liquids, substances, or materials. If batteries will be installed as part of the system, ensure that adequate precautions are taken (such as special ventilation) to prevent corrosive emissions from the batteries. Check local building codes for additional requirements. 4.7 Recommended Base station/Repeater Placement There is no one strategy for deploying base stations. These are some recommended Base station and/or Repeater placement strategies:
Around Corridors:
Base stations/repeater should be deployed vertically preferably at corridor intersections where propagation patterns follow the corridor patterns. The base station/repeater should point towards the corridor and preferably in the middle height between the floor and the actual ceiling. In case there are high objects in the area, the base station/repeater should be installed above those objects but still kept distant from the ceiling. Multi-Storey Buildings:
Base stations and repeaters can be installed on opposite sides of the floors to take advantage of the floor-
to-floor coverage. The coverage design cannot rely entirely on floor-to-floor propagation; each case must be verified due to variations in local attenuation patterns. Open Areas/ Large Halls:
Base stations and repeaters can be deployed in open areas for buildings that contain a central open space area with windows to the other areas. This provides a good coverage for the rooms in the inner circle on all floors (e.g. hotels). In large halls, Base stations/repeater should be installed vertically in the middle of the space below the drop ceiling. Mounting Positions:
When Base stations and repeaters are mounted vertically on a wall, the radio coverage in front of these devices is twice as large as the coverage at the rear. Repeaters should be installed in the middle of corridors and small rooms. Metallic Structures/Objects:
Base stations and repeaters should not be deployed near large metallic objects. Reinforced Concrete Structures:
These structures have a high attenuation factor inside the building. They reduce the radio coverage range of the Base stations and repeaters and therefore require a higher number of base stations or repeaters in the building. Lighter types of construction materials require fewer base stations since attenuation figures are considerably lower. Others Recommendations:
Maximum distance between two base stations varies depending on material and construction of buildings, but there must always be synchronization chains and radio coverage overlap between the two base stations or handover between radio units. The time it takes a person to cross the common coverage area must be 10 seconds or more, as the handset needs time to scan for an alternative base stations. Ensure that the installation area is located no closer than 6.1meters/20.0 feet from electric devices that produce large electro-magnetic fields (EMF) or high levels of radio frequency energy. Possible EMF n o i t a z i m i l i t p O
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 24 sources are radio transmitters, electric arc welding machines, copying machines, electric motors, refrigeration units, power transformers, electric load centers, and main circuit breaker panels. Ensure that the electrical service is sufficient and located in close proximity to the Base stations. 4.8 Network Assessment/Optimisation This involves monitoring, verifying and improving the performance of the SME VoIP network. Depending on the network setup and varying deployment conditions and network usage some requirements have to be monitored and corrected. The main focus of network optimisation should be telephony quality, handovers, network traffic and other related measurements. The quality of the network is ultimately determined by the satisfaction of users of the network. Therefore before SME VoIP Networks are handed over to customers, Network providers must perform walk or drive testing using the appropriate measurement kits. The walk or drive testing kit includes 2 or 3 test phones, a walk/drive testing software (deployment kit software) and a suggested walk path, etc. Collect statistics of the network an example is illustrated in the table below:
Parameters Comments Value
## Call Setup failures
## Dropped calls
## HO successes
## HO failures Traffic Blocking Rate
(%) Traffic Blocking (Erl) Receiver level (dBm) Receiver Quality (%) After collecting the necessary information, you should fine tune signalling and radio resource sharing parameters. Network optimisation is a continuous process during and after the launch of the network. For example, if it is found that an area within a building has low signal level. There should be an immediate scrutiny of base station and/or repeater locations, heights and tilts. The problem is sorted out by moving the relevant devices and altering the tilts of these devices. For buildings/halls constructed with high signal attenuation materials, deploying additional base stations will be one of the solutions. n o i t a z i m i l i t p O
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 25 5 Deployment Mechanics Multi-cell SME Network In this chapter we provide short description of practical cases of how Multi-cell SME VoIP network can be deployed unfold synchronisation details of how master bases latches to one or more slaves. Case ##1: Synchronisation Chain with One Master Sync. a) The Synchronisation chain must always overlap with other Base stations in order to latch each other in Sync. b) In this illustration Base station ##39 is the Sync Master c) A maximum of 7 devices (Including the master bases) can be chained at a time d) The other slave base stations or repeaters are connected to the Sync Master through the e) synchronisation chain If one of the base or repeater units in the sync chain is broken or not working, then the units that follows non-working device are cut off from the sync chain. Therefore handover is not possible between the non working units and working ones. f) However, handover is possible in the deployment figure below, because e.g. both BS#01 and BS#02 overlaps with BS#39. Therefore when BS#01 is not active, handover between BS#39 and the rest of the slaves is still permitted. k r o w t e N E M S l l e c
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r e t p a h C 26 SME VoIP System Guide, Version 1.5 Proprietary and Confidential Case ##2: Synchronisation Chain without Alternative Sync Paths a) Assuming Master Sync source is Base station ##39. A maximum of 40 base stations can be deployed in one setup (depending on the network requirement not all base stations should be chained). b) A maximum of 7 devices can be chained at one time. c) Depending on the system setup, it is recommended to place the Sync source Master in the middle of the building and to assign numbers/addresses, radio ID (RPN), etc., to each base station or repeater for easy identification. Continuous line: Shows the primary sync paths, with the relevant bases chained in the multi-cell network. Dotted line: Alternative sync paths, but cannot be used because the relevant base stations have not been chained. Case ##3: Synchronisation Chain with Alternative Sync Paths The illustration below shows a multi-cell network with alternative Synchronisation paths. A failure of one base unit does not mean handset or users cannot perform handovers to other active cells. BS#39 is the SYNC Master, if BS#05 is down, most user handovers can be formed via 3 other alternative cells (i.e. BS#06, BS#02 and BS#04) without any problems at all. Furthermore observe the following:
BS#04 and BS#01 are Primary with alternative sync to BS#39. BS#05 is primary sync to BS#04 while alternative sync is BS#01 or BS#02 BS#03 is primary sync to BS#02 while alternative sync is BS#05 or BS#06. k r o w t e N E M S l l e c
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 27 In the illustration below:
BS#24 is primary sync to BS#25 while alternative sync is BS#20 BS#22 is primary sync to BS#20 while alternative sync is BS#21 or BS#23. SME VoIP System Guide, Version 1.5 Proprietary and Confidential k r o w t e N E M S l l e c
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r e t p a h C 28 6 Installation of Base Stations/Repeater After planning the network, next is to determine the proper places or location the relevant base stations will be installed. Therefore, we briefly describe the how to install the base station in this chapter. 6.1 Package - Contents/Damage Inspection Before Package Is Opened:
Examine the shipping package for evidence of physical damage or mishandling prior to opening. If there is a proof of mishandling prior to opening, you must report it to the relevant support center of the regional representative or operator. Contents of Package:
Make sure all relevant components are available in the package before proceeding to the next step. Every shipped base unit package/box contains the following items:
2 x mounting screws and 2 x Anchors 1 x Metal plate(s) 1 x Plastic stand 1 x Cat. 5 cable (Ethernet cable) Base unit Damage Inspection:
The following are the recommended procedure for you to use for inspection:
1. Examine all relevant components for damage. 2. Make a defective on arrival - DOA report or RMA to the operator. Do not move the shipping carton until it has been examined by the operator. If possible send pictures of the damage. The operator/regional representative will initiate the necessary procedure to process this RMA. They will guide the network administrator on how to return the damaged package if necessary. If no damage is found then unwrap all the components and dispose of empty package/carton(s) in accordance with country specific environmental regulations. 3. SME VoIP System Guide, Version 1.5 Proprietary and Confidential 29 r e t a e p e R
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r e t p a h C RTX Base station Provides RF Channels to Handsets The base station supports 12 RF channels simultaneously for all DECT bands, summarised below:
The RF communication is provided according to the band standard at the site:
Frequency band:
1880 MHz 1930 MHz (DECT) 1880 1900 MHz (10 carriers) Europe/ETSI 1910 1930 MHz (10 carriers) LATAM 1920 1930 MHz (5 carriers) US Other frequency bands can be customized via Base station low-level debug programming. 6.2 RTX Base station Mechanics The base station front end shows an LED indicator that signals different functional states of the base unit and occasionally of the overall network. The indicator is off when the base unit is not powered. The table below summarises the various LED states:
LED State Unlit Unlit/Solid red Blinking green Solid red Blinking red Solid green Blinking red Solid red State No power in unit Error condition Initialisation Factory reset warning or long press in BS reset button Factory setting in progress Ethernet connection available (Normal operation) Ethernet connect not available OR handset de/registration failed Critical error (can only be identified by RTX Engineers). Symptoms include no system/SIP debug logs are logged, etc. Press reset button of base station. Orange r e t a e p e R
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 30 6.3 RTX Base Unit - Reset feature It is possible to restart or reset the base station unit by pressing a knob at the rear side of the unit. Alternatively, it can be reset from the SME Configuration Interface. We do not recommend this; but unplugging and plugging the Ethernet cable back to the PoE port of the base station also resets the base unit. 6.4 Installing the Base Station First determine the best location that will provide an optimal coverage taking account the construction of the building, architecture and choice of building materials. Next, mount the Base Station on a wall to cover range between 50 - 300 meters (i.e. 164 to 984 feet), depending whether its an indoor or outdoor installation. Please refer to chapter 4 for important information regarding network requirements, deployment considerations, site planning, cell coverage/capacity planning, environmental considerations and recommended Base station placement. Mounting the Base Stations/Repeaters:
We recommend the base station be mounted an angle other than vertical on both concrete/wood/plaster pillars and walls for optimal radio coverage. Avoid mounting the base units upside down as it significantly reduces radio coverage. Mount the base unit as high as possible to clear all nearby objects (e.g. office cubicles and cabinets, etc.). Occasionally extend coverage to remote offices/halls with lower telephony users by installing Repeaters. Make sure that when you fix the base stations with screws, the screws do not touch the PCB on the unit. Secondly, avoid all contacts with any high voltage lines. r e t a e p e R
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r e t p a h C 31 SME VoIP System Guide, Version 1.5 Proprietary and Confidential Documentation of Installation:
It is highly recommended to document the deployment of Multi-cell network; writing the locations or relevant geographical addresses each base unit has been installed. This is useful for maintenance purposes, so installed base units can be located easily on the field. SME VoIP System Guide, Version 1.5 Proprietary and Confidential r e t a e p e R
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r e t p a h C 32 7 Making Handset Ready In this chapter we briefly describe how to prepare the handset for use, install, insert and charge new batteries. Please refer to an accompanying Handset User Guide for more information of the features available in the Handset. 7.1 Package - Contents/Damage Inspection Before Package Is Opened:
Examine the shipping package for evidence of physical damage or mishandling prior to opening. If there is a proof of mishandling prior to opening, you must report it to the relevant support center of the regional representative or operator. Contents of Package:
Make sure all relevant components are available in the package before proceeding to the next step. Every shipped base unit package/box contains the following items:
2 x mounting screws and 2 x Anchors 1 x Handset hook 1 x A/C Adaptor 1 x Battery 1 x charger 1 x Handset Unit, 1 x Battery cover Damage Inspection:
The following are the recommended procedure for you to use for inspection:
1. Examine all relevant components for damage. 2. Make a defective on arrival - DOA report or RMA to the operator. Do not move the shipping carton until it has been examined by the operator. The operator/regional representative will y d a e R t e s d n a H g n i k a M
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 33 initiate the necessary procedure to process this RMA. They will guide the network administrator on how to return the damaged package if necessary. If no damage is found then unwrap all the components and dispose of empty package/carton(s) in accordance with country specific environmental regulations. 3. 7.2 Before Using the Phone Here are the pre-cautions users should read before using the Handset:
Installing the Battery 1. Never dispose battery in fires, otherwise it will explode. 2. Never replace the batteries in potentially explosive environments, e.g. close to inflammable liquids/
gases. 3. ONLY use approved batteries and chargers from the vendor or operator. 4. Do not disassemble, customise or short circuit the battery Using the Charger Each handset is charged through the use of a handset charger. The charger is a compact desktop unit designed to charge and automatically maintain the correct battery charge levels and voltage. The charger Handset is powered by AC supply from 110-240VAC that supplies 5.5VDC at 600mA. When charging the battery for the first time, it is necessary to leave the handset in the charger for at least 10 hours before the battery is fully charged and the handset ready for use. Handset In the Charger For correct charging, ensure that the room temperature is between 0C and 25C/32F and 77F. Do not place the handset in direct sunlight. The battery has a built-in heat sensor which will stop charging if the battery temperature is too high. If the handset is turned off when placed in charger, only the LED indicates the charging. When handset is turned off, the LED flashes at a low frequency while charging and lights constantly when the charging is finished. There will be response for incoming calls. If the handset is turned on when charging, the display shows the charging status. Open Back Cover 1. Press down the back cover and slide it towards the bottom of the handset. 2. Remove Back Cover from Handset Handset Serial Number The serial number (IPEI/IPUI number) of each handset is found either on a label, which is placed behind the battery, or on the packaging label. First, lift off handset back cover and lift the battery and read the serial number. y d a e R t e s d n a H g n i k a M
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 34 The serial number is needed to enable service to the handset. It must be programmed into the system database via the SME VoIP Configuration interface. Replace Battery Remove Back Cover from Handset. Remove the old battery and replace with a new one. 7.3 Using the Handset Please refer to chapter 16 for detail description of how to use the handset. y d a e R t e s d n a H g n i k a M
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 35 8 Core Network Server(s) Configuration In this chapter we describe how to setup the various server elements in the system. 8.1 Server setup In the SME network, the server environment is installed as a centralized system. The main server types hosted on the network include SIP, DNS/DHCP and HTTP/TFTP Servers. These servers can be hosted both in one or multiple windows and/or Linux Server environment. Management servers are normally installed to monitor and manage the network in detail. Each Base-
station status can be checked. Each Repeater and each Subscriber Terminal can be monitored over the air from a centralized location. Further, new software can be uploaded to all system elements from the centralized location (typically a TFTP server) on an individual basis. This includes Subscriber Handsets where the latest software is downloaded over the air. 8.2 Requirements Regardless of whether or not you will be installing a centrally provisioned system, you must perform basic TCP/IP network setup, such as IP address and subnet mask configuration, to get your organizations phones up and running. 8.3 DNS Server Installation/Setup Name server is a name server service installed in a server for mapping or resolution of humanly memorable domain names and hostnames into the corresponding numeric Internet Protocol (IP) addresses. The customer should refer to the platform vendor either windows or Linux vendor for detail step-by-step guide on how to install and configure Domain Name System for internet access. In this section, we briefly describe hints on how to setup DNS behind NAT or Firewall. 8.3.1.1 Hints on how to Configure DNS behind a Firewall/NAT Proxy and Network Address Translation (NAT) devices can restrict access to ports. Set the DNS to use UDP port 53 and TCP port 53. For windows Servers, set the RCP option on the DNS Service Management console and configure the RCP to use port 135. These settings should be enough to resolve some of potential issues that may occur when you configure DNS and firewalls/NAT. 8.4 DHCP Server Setup A DHCP Server allows diskless clients to connect to a network and automatically obtain an IP address. This server is capable of supplying each network client with an IP address, subnet mask, default gateway, an IP address for a WINS server, and an IP address for a DNS server. This is very often used in enterprise networks to reduce configuration efforts. All IP addresses of all computers/routers/bases are stored in a database that resides on a server machine. n o i t a r u g i f n o C
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 36 The network administrator should contact the relevant vendors for detail information or step-by-step procedure on how to install and setup DHCP process or service on windows/Linux servers. In this section, we will provide some hints of how to resolve potential problems to be encountered you setup DHCP Servers. 8.4.1 Hint: Getting DHCP Server to Work Windows Server:
1) Clients are unable to obtain an IP address If a DHCP client does not have a configured IP address, it generally means that the client has not been able to contact a DHCP server. This is either because of a network problem or because the DHCP server is unavailable. If the DHCP server has started and other clients have been able to obtain a valid address, verify that the client has a valid network connection and that all related client hardware devices
(including cables and network adapters) are working properly. 2) The DHCP server is unavailable When a DHCP server does not provide leased addresses to clients, it is often because the DHCP service has failed to start. If this is the case, the server may not have been authorized to operate on the network. If you were previously able to start the DHCP service, but it has since stopped, use Event Viewer to check the system log for any entries that may explain the cause. Next, restart the DHCP service, click Start, click Run, type cmd, and then press ENTER. Type net start dhcpserver, and then press ENTER. Linux Platform:
Troubleshooting DHCP, check the following:
1) Incorrect settings in the /etc/dhcpd.conf file such as not defining the networks for which the DHCP server is responsible;
2) NAT/Firewall rules that block the DHCP bootp protocol on UDP ports 67 and 68;
3) Routers failing to forward the bootp packets to the DHCP server when the clients reside on a separate network. Always check your /var/logs/messages file for dhcpd errors. 4) Finally restart the dhcpd service daemon n o i t a r u g i f n o C
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 37 8.5 TFTP Server Setup There are several TFTP servers in the market place, in this section we describe how to setup a commonly used TFTP Server. 8.5.1 TFTP Server Settings The administrator must configure basic parameters of the TFTP application:
Specify UDP 69 port - for TFTP incoming requests and TCP 12000 - for remote management of the server. For file transmission the server opens UDP ports with random numbers. In case the option Enable NAT or firewall support is activated on the server, the server uses the same port for files transmission and listening to the TFTP incoming requests (UDP 69 port on default). Specify the interface bindings, TFTP root directory, port which the TFTP Server will listen, timeout and number of retries, and TFTP options supported by the server. Configure the relevant TFTP virtual folder in the server. The TFTP virtual folder is the file folder, visible for TFTP clients under a certain name. You can set security settings separately for every virtual TFTP folder. Next, set rights to access TFTP folders according to the relevant clients. n o i t a r u g i f n o C
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 38 8.6 SIP Server Setup SIP server is one of the main components of an SME network, dealing with the setup of all SIP calls in the network. A SIP server is also referred to as a SIP Proxy or a Registrar. Although the SIP server is the most important part of the SIP based phone system, some servers only handles call setup and call tear down. It does not actually transmit or receive any audio. This is done by the media server in RTP. The RTX SME family of network phones are fully interoperable with the most of SIP Server applications. There are many off-the-shelf vendor and open source SIP servers. In this section, we will briefly explain settings required to take full advantage of FreePBX SIP Server feature set. The settings are similar for other SIP servers. 8.6.1 FreePBX SIP Server FreePBX is an easy to use GUI (graphical user interface) that controls and manages Asterisk, which the most popular open source telephony engine software. The administrator should refer to the relevant detail step-by-step procedure of how to install FreePBX SIP server. This section briefly describes SIP Server setup parameters. 1) SIP Server Setup Settings NAT Description This option determines the settings for users connecting to an asterisk server. Possible values: Yes, No, Never, Route NAT=route Asterisk will send the audio to the port and IP where its receiving the audio from. Instead of relying on the addresses in the SIP and SDP messages. This will only work if the phone behind NAT send and receive audio on the same port and if they send and receive the signalling on the same port. (The signalling port does not have to be the same as the RTP audio port). NAT=No Asterisk will add an RPORT to the via header of the SIP messages NAT=never This will cause asterisk not to add an RPORT in the VIA line of the sip invite n o i t a r u g i f n o C
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 39 Other NAT Settings Codecs Video Codecs Media & RTP Settings header Choose the relevant option or enter the settings in IP configuration, External IP, Local Network. Some SIP Servers supports dynamic codec support. Codecs are algorithm used to compress or decompress speech or audio signals. The user should select the relevant Codecs and other speech compression techniques whose traffic will be routed to the SME network. The user should enable this option if SME network supports video telephony. This option should be enabled to provide for deliver media streams (e.g., audio and video) or out-of-band events signaling (DTMF in separate payload type). 2) Extensions This feature allows administrators create handset profiles in the SME network. In other words, Extensions describes the Dial plan for the PBX SIP system. Enter the relevant parameters n o i t a r u g i f n o C
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r e t p a h C 41 9 SME VoIP Administration Interface The SME VoIP Administration Interface is also known as SME VoIP Configuration. It is the main interface through which the system is managed and debugged. The SME VoIP Configuration Interface is an in-built HTTP Web Server service residing in each base station This interface is user friendly interface and easy to handle even to a first time user. This chapter seeks to define various variables/parameters available for configuration in the network 9.1 Home/Status We describe the parameters found in the front end of the SME VoIP Administration Interface. Feature Home/Status Extensions Servers Network Description This is the front end of the Base stations HTTP web interface. This page shows the summary of current operating condition and settings of the Base station and Handset(s). Sets the Dial plan or phone numbers for the network. On this page the user can define which SIP/NAT server the network should connect to. Typically the user configures the Network settings from here. NAT provisioning: allows configuration of features for resolving of the NAT -
Network Address Translation. These features enable interoperability with most e c a f r e t n I i n o i t a r t s i n m d A P o V E M S I
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 42 Management Firmware Update Time Country Security Multi cell Settings Debug Log SIP Log types of routers. DHCP: allows changes in protocol for getting a dynamic IP address. Virtual LAN: specifies the Virtual LAN ID and the User priority. IP Mode: specifies using dynamic (DHCP) or static IP address for your SME network. IP address: if using DHCP leave it empty. Only write in, when you use static IP address. Subnet mask: if using DHCP, leave it empty. Only write in, when you use static IP address. DNS server: specify if using DHCP, leave it empty. Only write in the DNS server address of your Internet service provider, when you use static IP address. (DNS =
Dynamic Name Server) Default gateway: if using DHCP, leave it empty. Write in the IP address of your router, when you use static IP address. Defines the Configuration server address, Management transfer protocol, sizes of logs/traces that should be catalogued in the system. Remote firmware updates (HTTP/TFTP) settings of Base stations and handsets. Here the user can configure the Time server. It should be used as time server in relevant country for exact time. The time servers have to deliver the time to conform to the Network Time Protocol (NTP). Handsets are synchronised to this time. Base units synchronise to the master using the Time server. Specifying the country/territory where the SME network is located ensures that your phone connection functions properly. Note: The handset language and country setting are independent of each other. The users can create account credentials with which they can log in or log out of the embedded HTTP web server. Specify to connect base station or chain of base stations to the network. Make sure the system ID for the relevant base stations are the same otherwise the multi-cell feature will not work. This shows detail and complete SME network settings for base station(s), HTTP/DNS/DHCP/TFTP server, SIP server, etc. Overall network related events or logs is displayed here (only live feed is shown). SIP related logs can be retrieved from url link. It is also possible to clear logs from this feature. 9.2 Extensions In this section, we describe the different parameters available whenever the administrator is creating an extension for handset. Parameter Extension Description Handset phone number or SIP username depending on the setup. Possible value(s): 8-bit string length Example: 1024, etc. Note: The Extension must also be configured in SIP server in order for this feature to function. Username: SIP authentication username Password: SIP authentication password. Permitted value(s): 8-bit string length Authentication User Name/
Password SME VoIP System Guide, Version 1.5 Proprietary and Confidential e c a f r e t n I i n o i t a r t s i n m d A P o V E M S I
r e t p a h C 43 Display Name Mailbox Name/
Number Server Forwarding Unconditional Number Forwarding No Answer Number Forwarding On Busy Number Human readable name used for reference purposes on the HTTP web interface.(This does not display on handset) Permitted value(s): 8-bit string length Name of centralised system used to store phone voice messages that can be retrieved by recipient at a later time. Valid Input(s): 8-bit string length character (Latin characters for the Name and positive integer for the Number) Note: Mailbox Number parameter is available only when its enabled from SIP server. DNS or IP address of SIP server or Server of SME VoIP Service provider. Valid Input(s): AAA.BBB.CCC.DDD or URL e.g. www.sip-sme.com Number to which incoming calls must be re-routed to irrespective of the current state of the SIP node or handset. Note: Feature must be enabled in the SIP server before it can function in the network Number to which incoming calls must be re-routed to when there is no response from the SIP end node Note: Feature must be enabled in the SIP server before it can function in the network Number to which incoming calls must be re-routed to when SIP node is busy. Note: Feature must be enabled in the SIP server before it can function in the network e c a f r e t n I i n o i t a r t s i n m d A P o V E M S I SME VoIP System Guide, Version 1.5 Proprietary and Confidential 44
r e t p a h C Registrar Outbound Proxy Re-registration time Keep Alive DTMF Signalling Codec Priority 9.3 Servers In this section, we describe the different parameters available in the Servers configurations menu. Parameter NAT Adaption Description To ensure all SIP messages goes directly to the NAT gateway in the SIP aware router. NAT Adaption option should be No or otherwise choose Yes Possible value(s): Yes, No SIP Server proxy DNS or IP address Permitted value(s): AAA.BBB.CCC.DDD:<Port-Number> or <URL>:<Port-###>
Note: Specifying the Port Number is optional. This is a Session Border Controller DNS or IP address (OR SIP server outbound proxy address) Set the Outbound proxy to the address and port of private NAT gateway so that SIP messages sent via the NAT gateway. Permitted value(s): AAA.BBB.CCC.DDD or <URL> or <URL>:<Port-###>
Examples: 192.168.0.1, 192.168.0.1:5062, nat.company.com and sip:nat@company.com:5065. The window period (in seconds) when base stations SIP registers with SIP server. Permitted value(s): Positive integer This directive defines the window period (30 secs.) to keep opening the port of relevant NAT-aware router(s), etc. Valid Input(s): Enable, Disable Conversion of decimal digits (and '*' and '#') into sounds that share similar characteristics with voice to easily traverse networks designed for voice Valid Option(s) In band: Meta-data (e.g.: tone digits) and control information sent in the same voice band, using the same VoIP codec as the human voice (e.g. G.711, G.729, etc.) SIP INFO: Carries application level data along SIP signalling path (e.g.: Carries DTMF digits generated during SIP session OR sending of DTMF tones via data packets in the same internet layer as the Voice Stream, etc.). RFC 2833: DTMF handling for gateways, end systems and RTP trunks (e.g.:
Sending DTMF tones via data packets in different internet layer as the voice stream) Both: Enables SIP INFO and RFC 2833 modes. Defines the codec priority that base stations uses for audio compression and transmission. Possible Option(s): PCMU, PCMA, iLBC. e c a f r e t n I i n o i t a r t s i n m d A P o V E M S I SME VoIP System Guide, Version 1.5 Proprietary and Confidential 45
r e t p a h C 9.4 Network In this section, we describe the different parameters available in the network configurations menu. 9.4.1 IP Settings Parameter DHCP/Static IP IP Address Subnet Mask Description If DHCP is enabled, the device automatically obtains TCP/IP parameters. Possible value(s): Static, DHCP DHCP: IP addresses are allocated automatically from a pool of leased address. Static IP: IP addresses are manually assigned by the network administrator. If the user chooses DHCP option, the other IP settings or options are not available. 32-bit IP address of device (e.g. base station). 64-bit IP address will be supported in the future. Permitted value(s): AAA.BBB.CCC.DDD Is device subnet mask. Permitted value(s): AAA.BBB.CCC.DDD This is a 32-bit combination used to describe which portion an IP address refers to the subnet and which part refers to the host. A network mask helps users know which portion of the address identifies the network and which portion of the address identifies the node. Default Gateway Devices default network router/gateway (32-bit). Permitted value(s): AAA.BBB.CCC.DDD e.g. 192.168.50.0 IP address of network router that acts as entrance to other network. This device e c a f r e t n I i n o i t a r t s i n m d A P o V E M S I
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 46 DNS (Primary) DNS (Secondary) Screenshot provides a default route for TCP/IP hosts to use when communicating with other hosts on hosts networks. Main server to which a device directs Domain Name System (DNS) queries. Permitted value(s): AAA.BBB.CCC.DDD or <URL>
This is the IP address of server that contains mappings of DNS domain names to various data, e.g. IP address, etc. The user needs to specify this option when static IP address option is chosen. This is an alternate DNS server. 9.4.2 VLAN Settings Enable users to define devices (e.g. Base station, etc) with different physical connection to communicate as if they are connected on a single network segment. The VLAN settings can be used on a managed network with separate Virtual LANs (VLANs) for sending voice and data traffic. To work on these networks, the base stations can tag voice traffic it generates on a specific voice VLAN using the IEEE 802.1q specification. Parameter VLAN id Description Is a 12 bit identification of the 802.1Q VLAN. Permitted value(s): 0 to 4094 (only decimal values are accepted) A VLAN ID of 0 is used to identify priority frames and ID of 4095 (i.e. FFF) is reserved. Null means no VLAN tagging or No VLAN discovery through DHCP. This is a 3 bit value that defines the user priority. Values are from 0 (best effort) to 7 (highest); 1 represents the lowest priority. These values can be used to prioritize different classes of traffic (voice, video, data, etc). Permitted value(s): 8 priority levels (i.e. 0 to 7) VLAN User Priority Screenshot e c a f r e t n I i n o i t a r t s i n m d A P o V E M S I SME VoIP System Guide, Version 1.5 Proprietary and Confidential 47
r e t p a h C 9.4.3 Boot Server Options Parameter Boot Server DHCP Option Customer DHCP Option Content Option Type Screenshot Description Static:
The base station uses the IP settings configured manually in the boot server through the Network Menu. Option 66:
This the option code contained in the clients initial boot file. The network device searches for option 66 (string type) from the response received from the DHCP server. Custom:
The network device searches for the option number specified by the Boot Server Option parameter, and the type specified by the Boot Server Option Type parameter (below) in the response received from the DHCP server. Custom+Option 66:
The 1st choice option for network device will be to use the custom option if present and the 2nd choice is Option 66 if the custom option is not present. If the DHCP server sends nothing, the following scenarios are possible:
If a boot server value is stored in flash memory and the value is not 0.0.0.0, then the value stored in flash is used. Otherwise the network device sends out a DHCP INFORM query.
- If a single alternate DHCP server responds, this is functionally equivalent to the scenario where the primary DHCP server responds with a valid boot server value. The network device prefers the custom option value over the Option 66 value, but if no custom option is given, the device will use the Option 66 value.
- If no alternate DHCP server responds, the INFORM query process will retry and eventually time out. Permitted value(s): Static, Option 66, Custom, Custom+Option 66 This is a value (a positive integer) used to explain a DHCP action. When this parameter is set to Custom, this parameter specifies the DHCP option number in which the network device will look for its boot server. Permitted value(s): 128 through 254 (Cannot be the same as VLAN ID Option) This parameter should not matter when the Boot server DHCP option is set/enabled. This makes it possible for a user to choose some customized Boot Server Option types. These options are customized by the user itself in the configuration file. When this parameter is set to Custom, this parameter specifies the content type of the DHCP option in which a device should retrieve from its boot server. Permitted value(s): IP address, String The IP Address, must specify the boot server. The String can be URL, FTP, TFTP, HTTPS, etc. The address can be followed by optional directory. e c a f r e t n I i n o i t a r t s i n m d A P o V E M S I SME VoIP System Guide, Version 1.5 Proprietary and Confidential 48
r e t p a h C 9.4.4 NAT Settings We define some options available when NAT aware routers are enabled in the network. Parameter Enable RPORT Description Define whether RPORT should be used in SIP messages. Enabled by setting to Yes and disable by choosing No Permitted values: Yes, No. This option is disabled by default. Keep alive time This defines the frequency of how keep-alives are sent to maintain NAT bindings. Permitted values: Positive integer default is 90, unit is in seconds 9.4.5 SIP/RTP Settings These are some definitions of SIP/RTP Server settings:
Parameter Local SIP port Description SIP server port number reserved for a specific customer by the network operator. Port used for first user agent (UA) instance. Succeeding UA's will get succeeding ports. Permitted values: Port number default 5060. Priority of call control signalling traffic based on both IP Layers of Type of Service
(ToS) byte. ToS is referred to as Quality of Service (QoS) in packet based networks. Permitted values: Positive integer default is 104 Usually the first RTP port to use for RTP audio streaming. Permitted values: Port number default 50004 (depending on the setup). The number of ports that can be used for RTP audio streaming. Permitted values: Positive integers, default is 20 Priority of RTP traffic based on the IP layer ToS (Type of Service) byte. ToS is referred to as Quality of Service (QoS) in packet based networks. See RFC 1349 for details. cost bit" is not supported. o Bit 7..5 defines precedence. o Bit 4..2 defines Type of Service. o Bit 1..0 are ignored. Setting all three of bit 4..2 will be ignored. Permitted values: Positive integer default is 160 SIP ToS/QoS RTP port RTP port range RTP TOS/QoS e c a f r e t n I i n o i t a r t s i n m d A P o V E M S I SME VoIP System Guide, Version 1.5 Proprietary and Confidential 49
r e t p a h C Screenshot 9.5 Management Settings Definitions The administrator can configure base stations to perform some specific functions such as configuration of file transfers, firmware up/downgrades, password management, and SIP/debug logs. Parameter Title name of the web site Description This option is optional. It indicates the title that should appear at the top window of the browser i.e. <Enter_title_here>-Browser_name By default: SME VoIP Configuration Server/device that provides configuration file to base station. Type: DNS or IP address Permitted value(s): AAA.BBB.CCC.DDD (Currently only IPv4 are supported) or URL
(e.g.: firmware.rtx.net) The protocol assigned for configuration file and firmware downloads (or uploads) Valid Input(s): HTTP, TFTP Note: Choose TFTP if Bases/handsets firmware are updated over-air via TFTP Server The folder location or directory path that contains the configuration file of the Configuration server. The configuration upload script is a file located in e.g. TFTP server or Apache Server which is also the configuration server. Permitted value(s): /<configuration-file-directory>
Example: /CfgUpload Note: Must begin with (/) slash character. Either / or \ can be used. Password that should be entered in order to have access to the configuration server. Permitted value(s): 8-bit string length Enable this to save low level system debug messages in the configuration server. Further, the user can specify the kind of contents that should be saved. Choose Boot Log option to save only boot related messages (i.e. logs during start up of the system). Choose Everything option if you want to save all debug logs including boot logs. Valid Input: Disable, Boot Log, Everything The Debug log will be saved in the file format <Time_Stamp>b.log in the TFTP server as specified in the upload script. The upload script is a routine enabled at the TFTP/HTTP server. Enable this option to save low level SIP debug messages in the configuration server. Enable by choosing the option: Everything. Valid Input: Disable, Everything Configuration server address Management Transfer Protocol HTTP Management upload script HTTP Management password Upload of Debug Log Upload of SIP Log e c a f r e t n I i n o i t a r t s i n m d A P o V E M S I
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 50 Trace Server Trace Server IP Address The SIP logs are saved in the file format: <MAC_Address><Time_Stamp>SIP.log Enable this option to save mail traces. Mail server traces are low level internal log messages or traces used for debug purposes by RTX engineers. Valid Input: Disable, Enable The trace server uses the same address as the configuration server. Permitted value(s): AAA.BBB.CCC.DDD (Currently only IPv4 are supported) or url
(e.g.: firmware.rtx.net) 9.6 Firmware Update Definitions In this page, the system administrator can configure how base stations and SIP nodes upgrade/downgrade to the relevant firmware. Parameter Firmware update server address Description IP address or DNS of firmware update files source Valid Inputs: AAA.BBB.CCC.DDD or <URL>
Example: firmware.rtx.net or 10.10.104.41 Location of firmware on server (or firmware update server path where firmware update files are located). Example: /East_Fwu Note: Must begin with (/) slash character Version of firmware to be upgraded (or downgraded) on Base station or handset. Base units are referred to as gateways over here. Valid Input(s): 8-bit string length. E.g. 0034, etc. Firmware path Required Version e c a f r e t n I i n o i t a r t s i n m d A P o V E M S I SME VoIP System Guide, Version 1.5 Proprietary and Confidential 51
r e t p a h C 9.7 Time Server In this section, we describe the different parameters available in the Time Server menu. The Time server supplies the time is used in the debug logs and SIP trace information pages. It is also used to determine when to check for new configuration and firmware files. NOTE:
You should set the time server for multi-cell configuration (mandatory). It is not necessary to set the time server for standalone base stations (optional). Parameter Time Server Description DNS name or IP address of NTP server. Enter the IP/DNS address of the server that distributes reference clock information to its clients including Base stations, Handsets, etc. Valid Input(s): AAA.BBB.CCC.DDD or URL (e.g. time.server.com) Currently only IPv4 address (32-bit) nomenclature is supported. The window time in seconds within which time server refreshes. Valid Inputs: positive integer Refers to local time in GMT or UTC format. Min: -12:00 Max: +13:00 Enter the start and stop dates if you select Automatic. The system administrator can Enable or Disable DST manually. You determine when DST actually changes. Choose the relevant date or day of the week, etc. from the drop down menu. Month that DST begins Valid Input(s): Gregorian months (e.g. January, February, etc.) Numerical day of month DST comes to effect when DST is fixed to a specific date Time server refresh interval Time Zone Daylight Saving Time (DST) DST Fixed By Day DST Start Month DST Start Date SME VoIP System Guide, Version 1.5 Proprietary and Confidential 52 e c a f r e t n I i n o i t a r t s i n m d A P o V E M S I
r e t p a h C DST Start Time DST Start Day of Week DST Start Day of Week, Last in Month DST Stop Month DST Stop Date DST Stop Time DST Stop Day of Week Last in Month Valid Inputs: positive integer DST start time in the day Valid Inputs: positive integer Day within the week DST begins Specify the week that DST will actually start. The month that DST actually stops. The numerical day of month that DST turns off. Valid Inputs: positive integer (1 to 12) The time of day DST stops Valid Inputs: positive integer (1 to 12) The week within the month that DST will turn off. 9.8 Multi-cell Parameter Definitions In this section, we describe the different parameters available in the Multi-cell configurations menu. 9.8.1 Settings for Base Unit Description of Settings for Specific Base units is as follows:
Parameter Multi cell system Enable this option to allow the Base unit to be set in multi-cell mode (can be set Description either as master or slave in the multi-cell chain system refer to MAC-units in e c a f r e t n I i n o i t a r t s i n m d A P o V E M S I
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 53 System chain ID Synchronization time (s) Multi cell debug Chain section for details). Valid Inputs: Enable, Disable This is an identifier (in string format e.g. 2275) that is unique for a specific multi-
cell system. Note: There can be several multi-cell systems in SME network. Up to 7 levels of base stations chains are permitted in a typical setup. Please refer to accompanied document [2] for further details and description. Valid Input: 16 bit String length This specifies the period in seconds when elements/nodes (e.g. Base units) in a specific Multi-cell should synchronise to each other. Enable this feature, if you want the system to catalogue low level multi-cell debug information or traces. 9.8.2 DECT System Settings Description of DECT Settings for Specific Base units is as follows:
Parameter DECT system RFPI Description This is a radio network identity accessed by all Base units in a specific multi-cell system. It composed of 5 octets. It is actually 5 different variables combined together. RFPI Format: XX XX XX XX XX (where XX are HEX values) Note: Only type e.g. 11 6E 60 49 04 the system reformats as 0x11 0x6E 0x60 0x49 0x04 Access Rights Class (ARC):
Defines network identity structure used by terminals especially in multi-cell environment. Fixed/default Value=1 (Private multi-cell system). RFPI: 1X XX XX XX XX Equipment Installer's code (EIC):
Code that allows terminals to distinguish between separate DECT networks. Example_RTX_EIC: 0x16E6 (May change in the future) RFPI: 11 6E 6X XX XX Fixed Part Number(FPN):
Is a geographically unique identity transmitted to DECT networks to help PP distinguish between base station communications in different cells/multi-cell systems. E.g. FPN: 0x049 RFPI: 11 6E 60 49 XX e c a f r e t n I i n o i t a r t s i n m d A P o V E M S I
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 54 Auto configure DECT sync source tree Location Area Length (Lal):
A unique code sent to the terminal during location registration to determine the size of the location or cell area. Type: 8-bit value (from 0x00 to 0xFF). E.g Lal=0x04 RFPI: 11 6E 60 49 04 Enable this to allow the network to automatically synchronise the multi-cell chain/tree. (Not available in some base station firmware(s)). Permitted Inputs: Enable, Disable 9.8.3 SIP System Settings Description of SIP Settings for Specific Base units is as follows:
Description The maximum number of handsets or SIP end nodes that are permitted to perform location registration on a specific Base unit before load is distributed to other base units. Note: A maximum of 8 simultaneous calls can be routed through each Base units in a multi-cell setup. Permitted Input: Positive Integers (e.g. 6) Enable this option so it is possible to use same extension (i.e. SIP Account) on multiple phones (SIP end nodes). These phones will ring simultaneously for all incoming calls. When a phone (from a SIP account group) initiates a handover from Base X to Base Y, this phone will de-register from Base X, and register to Base Y after a call. Note: Choose Yes when the SIP server supports this feature otherwise choose No for the Sip server does not support this feature. Permitted Input: Yes, No Parameter
## of SIP accounts before distributed load SIP Support for multiple registrations per account 9.8.4 MAC-units in Chain The definitions for various parameters settings of various chain levels for a typical multi-cell system. Parameters ID Description Base unit identity in the chained network. Permitted Output: Positive Integers The Radio Fixed Part Number, is an 8-bit DECT cell identity allocated by the RPN e c a f r e t n I i n o i t a r t s i n m d A P o V E M S I
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 55 installer. The allocated RPN within the SME must be geographically unique. Permitted Output: 0 to 255 (DEC) OR 0x00 to 0xFF (HEX) Contains the hardware Ethernet MAC address of the base station. It varies from Base station to Base stations. Base station current firmware version. Permitted Output: positive Integers (e.g. 34) Current Base station behaviour in the SME network. Possible Outputs Connected: The relevant Base station(s) is online in the network Connection Loss: Base station unexpectedly lost connection to network This Unit: Current Base station whose http Web Interface is currently being accessed The administrator should choose the relevant multi cell chain level its wants a specific Base unit be placed. Maximum number of multi-cell chain levels is 6. Base station characteristics in connection to the current multi cell network. Possible Output(s) Master: Main Base station unto which all other nodes in the chain synchronises to. Locked: The Base unit is currently synchronized and locked to the master Base unit. Searching: Base unit in the process of locating to a Master/slave as specified in Dect sync source Free Running: A locked Base unit that suddenly lost synchronisation to the Master. Unknown: No current connection information from specific Base unit MAC Address Version Status DECT Sync source Dect Property 9.9 Settings Configuration File Setup This page provides non editable information showing the native format of entire SME VoIP Configuration parameter settings. The settings format is exactly what is used in the configuration file. The configuration file is found in the TFTP server. The filename for the configuration server is <MAC_Address>.cfg. The configuration file is saved in the folder /Config in the TFTP sever. There are three ways to edit the configuration file or make changes to the settings page:
1) Using the SME VoIP Configuration interface to make changes. Each page of the HTTP web interface is a template for which the user can customise settings in the configuration file. e c a f r e t n I i n o i t a r t s i n m d A P o V E M S I
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 56 2) Retrieving the relevant configuration file from the TFTP and modify and enter new changes. This should be done with an expert network administrator. 3) Navigate to the settings page of the VoIP SME Configuration interface > copy the contents of settings > save them to any standard text editor e.g. notepad > modify the relevant contents, make sure you keep the formatting intact > Save the file as <Enter_MAC_Address_of_RFP>.cfg > upload it into the relevant TFTP server. An example of contents of settings is as follows:
~RELEASE=UMBER_FP_V0054
%GMT_TIME_ZONE%:16
%COUNTRY_VARIANT_ID%:18
%FWU_POLLING_ENABLE%:0
%FWU_POLLING_MODE%:0
%FWU_POLLING_PERIOD%:86400
%FWU_POLLING_TIME_HH%:3
%FWU_POLLING_TIME_MM%:0
%DST_ENABLE%:2
%DST_FIXED_DAY_ENABLE%:0
%DST_START_MONTH%:3
%DST_START_DATE%:1
.... ... 9.10 Debug Logs This page shows live feed of system level messages of the current base station. The messages the administrator see here depends on what is configured at the Management settings. The Debug logs can show only Boot Log or Everything that is all system logs including boot logs. The Debug log is saved in the file format <Time_Stamp>b.log in a relevant location in the TFTP server as specified in the upload script. A sample of debug logs is as follows:
0101000013 [N](01):DHCP Enabled 0101000013 [N](01):IP Address: 192.168.10.101 0101000013 [N](01):Gateway Address: 192.168.10.254 0101000013 [N](01):Subnet Mask: 255.255.255.0 0101000013 [N](01):TFTP boot server not set by DHCP. Using Static. 0101000013 [N](01):DHCP Discover completed 0101000013 [N](01):Time Server: 192.168.10.11 0101000013 [N](01):Boot server: 10.10.104.63 path: Config/ Type: TFTP 0101000013 [N](01):RemCfg: Download request of Config/00087b077cd9.cfg from 10.10.104.63 using TFTP 0101000014 [N](01):accept called from task 7 0101000014 [N](01):TrelAccept success [4]. Listening on port 10010 0101000019 [N](01):RemCfg: Download request of Config/00087b077cd9.cfg from 10.10.104.63 using TFTP 0101000019 [W](01):Load of Config/00087b077cd9.cfg from 10.10.104.63 failed e c a f r e t n I i n o i t a r t s i n m d A P o V E M S I SME VoIP System Guide, Version 1.5 Proprietary and Confidential 57
r e t p a h C 9.11 SIP Logs This page shows SIP server related messages that are logged during the operation of the SME system. The full native format of SIP logs is saved in the TFTP server as <MAC_Address><Time_Stamp>SIP.log These logs are saved in 2 blocks of 17Kbytes. When a specific SIP log is fully dumped to one block, the next SIP logs are dumped to the other blocks. An example of SIP logs is shown below:
..... Sent to udp:192.168.10.10:5080 at 12/11/2010 11:56:42 (791 bytes) REGISTER sip:192.168.10.10:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.101:5063;branch=z9hG4bKrlga4nkuhimpnj4.qx Max-Forwards: 70 From: <sip:Ext003@192.168.10.10:5080>;tag=3o5l314 To: <sip:Ext003@192.168.10.10:5080>
Call-ID: p9st.zzrfff66.ah8 CSeq: 6562 REGISTER Contact: <sip:Ext003@192.168.10.101:5063>
Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Expires: 120 User-Agent: Generic-DPV-001-A-XX(Generic_SIPEXT2MLUA_v1) Content-Type: application/X-Generic_SIPEXT2MLv1 Content-Length: 251
..... e c a f r e t n I i n o i t a r t s i n m d A P o V E M S I
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 58 10 Firmware Upgrade Management This step-by-step chapter describes how to upgrade or downgrade base station(s) and/or handset(s) to the relevant firmware provided by the RTX. RTX supports and provides to the vendor an interface that performs the following operations related to firmware on the device:
1) Verify whether the new firmware package is compatible with the device. 2) Upgrade the firmware on the device to the new firmware 3) Roll back the firmware on the device to the previous firmware version (where necessary) 10.1 Network Dimensioning In principle, a number of hardware and software components should be available or be satisfied before base station/handset update can be possible. The minimum hardware and software components that are required to be able update via TFTP include the following (but not limited to):
Handsets Base stations TFTP Server (Several Windows and Linux applications are available) DHCP Server (Several Windows and Linux applications are available) Workstation (e.g. Normal terminal or PC) Any standard browser (e.g. firefox) Public/Private Network t n e m e g a n a M e d a r g p U e r a w m r i F SME VoIP System Guide, Version 1.5 Proprietary and Confidential 59
r e t p a h C 10.2 TFTP Configuration This section illustrate TFTP Server configuration using SolarWinds vendor TFTP Server. Create the following relevant folders as shown in the snap shots and choose defaults settings for the remaining options and save. t n e m e g a n a M e d a r g p U e r a w m r i F SME VoIP System Guide, Version 1.5 Proprietary and Confidential 60
r e t p a h C 10.3 Create Firmware Directories Within the same TFTP Server Interface do the following:
STEP 1 Make a folder named BeatUs in the TFTP-Root and place the fwu file(s)/(Firmware) in this folder. The base firmware must be renamed to BeatUsSw_v00xx.fwu. STEP 2 The admin from the service providers side must create the relevant firmware directory in the server where both old and new firmware(s) can be placed in it. (See the STEP above) The firmware directory or path should be \<Server>\<FwuPath>\BeatUs\, where <Server> is the root directory of the server (e.g. C:\TFTP-Root) and <FwuPath> is a folder within the <Server> that contains the BeatUs directory. IMPORTANT:
The BeatUs directory name cannot be changed. 10.4 Login to Base SME Configuration Interface STEP 3 Connect the Base station to a private network via standard Ethernet cable (CAT-5). STEP 4 Open any standard browser and enter the address:
http://ipdect<MAC-Address-Base-Station>
e.g. http://ipdect00087B00AA10. This will retrieve the HTTP Web Server page from the base station with hardware address 00087B00AA10. STEP 5 You can also use an IP/network protocol analyser e.g. Wireshark (freeware program) to identify which IP the base has received. t n e m e g a n a M e d a r g p U e r a w m r i F
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 61 Below is shown how to see which IP address the base has received from the DHCP server. In the example we start the trace and filter on bootp. Then we power up the base which is connected to the same network as the sniffer (wireshark). After a short while an offer is given by the DHCP server, and it is possible to see that the base received the IP address 192.168.50.76 STEP 6 On the Login page, enter your authenticating credentials (i.e. username and password). By default the username and password is admin. Click OK button. t n e m e g a n a M e d a r g p U e r a w m r i F
r e t p a h C 62 SME VoIP System Guide, Version 1.5 Proprietary and Confidential STEP 7 Once you have authenticated, the browser will display front end of the SME Configuration Interface. The front end will show relevant information of the base station. STEP 8 Click on Management URL. Enter the relevant Management server information in the system. Select the Management transfer protocol TFTP drop down menu. 10.5 Firmware Update Settings STEP 9 Scroll down and Click on Firmware Update url link in the SME VoIP Configuration Interface to view the Firmware Update Settings page. t n e m e g a n a M e d a r g p U e r a w m r i F
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 63 Parameters Description Firmware Update Settings Firmware update server address This is the IP address of server where the firmware is located. Currently, only 32-
bit is supported (i.e. IPv4 type <aaa.bbb.ccc.ddd>) Firmware path The firmware is found at the \<Server>\<FwuPath>\BeatUs\ directory found in the FTP or TFTP server. Note: Either / or \ can be used The <Server> is the root directory of the server created by the administrator and should NOT be specified. The <FwuPath> is a folder within the <Server> that contains the BeatUs directory. This MUST be specified. By default the \BeatUs is hard-coded into the firmware. Therefore it should not be specified in the firmware path. Example of firmware path is \HQ_Office, \South_Office, or \FwuPath, etc. in that manner. Update Gateways/Handsets Required Version This is 8-bit value. The firmware filename is BeatUsSw_v00XX.fwu. The administrator has to enter a numerical value XX or 00XX, where XX is a positive integer. STEP 10 Click on Firmware Update URL. On the Firmware Update Settings page enter the relevant parameters as described in the table above. Next, Click on Save button to keep the modified parameters into the base station. t n e m e g a n a M e d a r g p U e r a w m r i F SME VoIP System Guide, Version 1.5 Proprietary and Confidential 64
r e t p a h C 10.6 Base Station(s) Firmware Upgrade STEP 11 On the Firmware Update Settings page > scroll down to the Update Gateways section >
Enter the relevant firmware version (e.g. 11) of the base station to upgrade or to downgrade. It is possible to upgrade a single base station and/or several base stations > the admin should choose right the radio button. STEP 12 Still on the same Update Gateway section > choose Start update button > select OK button from the dialog window to start the update/downgrade procedure. The relevant base station(s) will automatically reboot and retrieve the firmware specified from the server and update itself accordingly. All on-going voice calls are dropped from the base station(s) immediately the firmware update procedure starts. t n e m e g a n a M e d a r g p U e r a w m r i F SME VoIP System Guide, Version 1.5 Proprietary and Confidential 65
r e t p a h C 10.7 Handset (s) Firmware Upgrade STEP 13 Scroll down to Update handsets section on the Firmware Update Settings page > Enter the relevant handset firmware (e.g. 11) to upgrade or downgrade > press Start update button > Click on OK button from the dialog window to initialize the process of updating all handsets in the private network. As at the time of documenting this chapter, two different handset hardware module exist i.e. UXP1240H HW ver 00 and the native hardware UXP1240H. 10.8 Verification of Firmware Upgrade STEP 14 From the Handset Menu navigate to Settings > Scroll down to Status this will list information regarding Base station and Handset firmware versions. SME VoIP System Guide, Version 1.5 Proprietary and Confidential t n e m e g a n a M e d a r g p U e r a w m r i F 66
r e t p a h C STEP 15 Now the download should be initiated and it should be stated in the log window of the TFTP server:
STEP 16 During the download, the Wireshark shows the download as shown below:
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r e t p a h C 10.9 Reboot the Base station(s) STEP 17 In principle the base station(s) should reboot automatically when the Start update button is selected > to begin the firmware update procedure. If for some unknown reasons the base station does restart, then the admin must manually reboot the base station so the firmware update process can begin in the base station. Make sure the URL is shown on the page before rebooting the base station, e.g. tftp://192.68.50.1/FwuPath Click OK button from the dialog window. A successful restart of the base stations will lead to a display of the page: Gateway has been reset. The firmware update is now in progress. STEP 18 Wait about 3-5 minutes, Reboot the base/gateway. The base/gateway will now be updated (base LED will flash). The software version number on the start page should be changed to the new version number. The message Base FWU ended with exit code -2101 is shown in the debug log and the new firmware will be running after a restart of the base/gateway. SME VoIP System Guide, Version 1.5 Proprietary and Confidential 68
r e t p a h C t n e m e g a n a M e d a r g p U e r a w m r i F t n e m e g a n a M e d a r g p U e r a w m r i F SME VoIP System Guide, Version 1.5 Proprietary and Confidential 69
r e t p a h C 11 Registration Management - Handset In this chapter we briefly describe how to add server and register handsets in the SME VoIP Network. 11.1 Hardware required This section describes what hardware is needed to execute the guideline. You need the following to perform a registration of a handset to a base station (gateway):
1 x base station 1 x handset 1 x Ethernet cable (PoE) 1 x PC connected to the LAN, VLAN or WAN Plug the Ethernet cable into the base station and connect the other end of the cable to your network (LAN, VLAN or WAN ). The base station is powered via the Ethernet cable (PoE). PC must be connected to the same network as the base station. If your network doesnt support PoE (Power over Ethernet), the a PoE adaptor must be used. t e s d n a H
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r e t p a h C 70 SME VoIP System Guide, Version 1.5 Proprietary and Confidential 11.2 Software required equipment This section describes what software is needed. You will need the following to access the web configuration interface on the base station:
Any standard browser e.g. Windows IE installed on the PC 11.3 Add server This section describes how to add a SIP server to the base station (gateway). STEP 1 Open browser on the computer and type in the IP address of the base. Press Enter to access the base. If you dont know the IP address, then write ipdect<MAC address>. The MAC address is written on the base/gateway. If the MAC-address format does not work then use the network protocol analyser (e.g. wireshark) to retrieve the IP address of Base station. Enter the IP address to browser. STEP 2 Enter the Username and Password when prompted. The default username and password is admin. t e s d n a H
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r e t p a h C 71 SME VoIP System Guide, Version 1.5 Proprietary and Confidential STEP 3 A SIP server must be configured on the base before handsets can be registered to the base. To add a SIP server, select sheet Server and click the link Add server. STEP 4 Fill out the displayed form and click Save. In the shown example we configure a SIP server located on the IP address 192.168.50.77 and no Outbound proxy is used. Set NAT adaption if the base is behind NAT . It is also possible to type in the URLs direct link, i.e. like sipphone.com NOTE:
The new server is displayed to the left when it is successfully saved. t e s d n a H
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 72 11.4 Register handset to base This section describes how to register the wireless handset to the base station. NOTE:
Minimum one server must be registered to the base (system), otherwise a handset cannot be registered to the system. Please see chapter Add server. STEP 1 Open Windows Internet Explore on the computer and type in the ipdect<MAC address> or IP address of the base. Press Enter to access the base. STEP 2 Type in the Username and Password when prompted. The default username and password is admin. t e s d n a H
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r e t p a h C 73 SME VoIP System Guide, Version 1.5 Proprietary and Confidential STEP 3 Select Extensions URL and click Add extension link STEP 4 Fill out the form and click Save. In the example below we add the extension 3020 and this SIP account got the same number as Authentication User Name and Password. The Server 1 is selected by default as server for this extension. STEP 5 Set a Check mark on the extension which shall be assigned to the handset you want to register and click Register handset (s). The base is now open (ready state) for handset registrations for 5 minutes. t e s d n a H
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 74 STEP 6 Start the registration procedure on the handset by following step a to d below. a) Select main menu Connectivity b) Select menu Register c) Type in the AC code and press OK to start the registration. The default AC code is 0000. SME VoIP System Guide, Version 1.5 Proprietary and Confidential t e s d n a H
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r e t p a h C 75 d) After a while the handset is registered, and the idle display is shown. NOTE:
The unique handset IPEI is displayed on sheet Extensions when the handset is successfully registered. The web page must be manually updated by pressing F5 to see that the handset is registered; otherwise the handset IPEI (International Portable Equipment Identity) isnt displayed on the web page. We illustrate how extensions page will be when you register several handsets. SME VoIP System Guide, Version 1.5 Proprietary and Confidential t e s d n a H
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r e t p a h C 76 12 VLAN Setup Management In this chapter we describe how to setup a typical VLAN in the network. 12.1 Introduction In this chapter, we describe how to setup VLAN to typical network. There are three main stages involved in this procedure:
1) Configure a VLAN Aware Switch to a specific (un)tagged VLAN ID, so the SME system can process untagged frames forwarded to it. 2) Setup the Time Server (NTP Server) and other relevant network servers. 3) Configure the HTTP server in relevant Base stations to access the features in the PBX or SME system. VLAN allows administrators to separate logical network connectivity from physical connectivity analogous to traditional LAN which is limited by its physical connectivity. Normally, users in a LAN belong to a single broadcast domain and communicate with each other at the Data Link Layer or Layer 2. LANs are segmented into smaller units for each IP subnets and here communication between subnets is possible at the Network Layer or Layer 3, using IP routers. A VLAN can be described as a single physical network that can be logically divided into discrete LANs that can operate independently of each other. An Illustration of using VLANs to create independent broadcast domains across switches is shown below:
The figure above highlights several key differences between traditional LANs and VLANs. All switches are interconnected to each other. However, there are three different VLANs or broadcast domains on the network. Physical isolation is not required to define broadcast domains. If the figure was a traditional LAN without VLAN-aware switches, all stations would belong to one broadcast domain. All switch ports can communicate with one another at the Data Link Layer, if they become t n e m e g a n a M p u t e S N A L V
r e t p a h C members of the same VLAN. SME VoIP System Guide, Version 1.5 Proprietary and Confidential 77 The physical location of an end station does not define its LAN boundary. 1. An end station can be physically moved from one switch port to another without losing its view of the network. That is, the set of stations it can communicate with at the Data Link Layer remains the same, provided that its VLAN membership is also migrated from port to port. 2. By reconfiguring the VLAN membership of the switch port an end station is attached to, you can change the network view of the end station easily, without requiring a physical move from port to port. 12.2 Backbone/ VLAN Aware Switches To implement a VLAN in your network, you must use VLAN-aware switches. Before we continue, let consider two rules to remember regarding the functioning of a regular LAN switch:
1. When the switch receives a broadcast or multicast frame from a port, it floods (or broadcasts) the frame to all other ports on the switch. 2. When the switch receives a unicast frame, it forwards it only to the port to which it is addressed. A VLAN-aware switch changes the above two rules as follows:
1. When the switch receives a broadcast or multicast frame from a port, it floods the frame to only those ports that belong to the same VLAN as the frame. 2. When a switch receives a unicast frame, it forwards it to the port to which it is addressed, only if the port belongs to the same VLAN as the frame. 3. A unique number called the VLAN ID identifies each VLAN. Which VLAN Does a Frame Belong To?
The previous section notes that a frame can belong to a VLAN. The next question ishow is this association made?
A VLAN-aware switch can make the association based on various attributes of the type of frame, destination of MAC address, IP address, TCP port, Network Layer protocol, and so on. SME VoIP System Guide, Version 1.5 Proprietary and Confidential t n e m e g a n a M p u t e S N A L V 78
r e t p a h C An illustration of IEEE 802.1Q VLAN tag in Ethernet frame is as follows:
12.3 How VLAN Switch Work: VLAN Tagging VLAN functionality can be implemented via explicit frame tagging by switches and end stations. Network switches and end stations that know about VLANs are said to be VLAN aware. Network switches and end stations that can interpret VLAN tags are said to be VLAN tag aware. VLAN-tag-
aware switches and end stations add VLAN tags to standard Ethernet framesa process called explicit tagging. In explicit tagging, the end station or switch determines the VLAN membership of a frame and inserts a VLAN tag in the frame header (see figure above for VLAN tagging), so that downstream link partners can examine just the tag to determine the VLAN membership. 12.4 Implementation Cases Common types of usage scenarios for VLANs on typical VLAN switches: port-based VLANs, protocol-based VLANs, and IP subnet-based VLANs. Before figuring out which usage scenario suits your needs, you must understand what each type of usage scenario implies. Port-based VLAN: All frames transmitted by a NIC are tagged using only one VLAN ID. The NIC does not transmit or receive any untagged frames. All protocols and applications use this virtual interfaces virtual PPA to transmit data traffic. Therefore all frames transmitted by that NIC port are tagged with the VLAN ID of that Virtual Interface. Protocol-based VLAN: The NIC assigns a unique VLAN ID for each Layer 3 protocol (such as IPv4, IPv6, IPX, and so on). Therefore, the VLAN ID of outbound frames is different for each protocol. An inbound frame is dropped if the protocol and VLAN ID do not match. IP subnet-based VLAN: The NIC assigns a unique VLAN ID for each IP subnet it belongs to. Therefore, the VLAN ID of outbound frames is different for different destination subnets. An inbound frame is dropped if the IP subnet and VLAN ID do not match. t n e m e g a n a M p u t e S N A L V
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 79 12.5 Base station Setup After the admin have setup the Backbone switch, next is to configure the Base stations via HTTP interface. STEP 1 Connect the Base station to a private network via standard Ethernet cable (CAT-5). STEP 2 Open any standard browser and enter the address:
http://ipdect<MAC-Address-Base-Station> or http://<IP-address of base station>
This will retrieve the HTTP Web Server page from the base station. STEP 3 On the Login page, enter your authenticating credentials (the username and password is admin by default unless it is changed). Click OK button. STEP 4 Once you have authenticated, the browser will display front end of the SME Configuration Interface. The front end will show relevant information of the base station. STEP 5 Create the relevant SIP server information in the system. Each service provider/customer should refer SIP server vendor on how to setup SIP servers. t n e m e g a n a M p u t e S N A L V SME VoIP System Guide, Version 1.5 Proprietary and Confidential 80
r e t p a h C 12.6 Configure Time Server STEP 6 Navigate to the Time settings and configure it. Scroll on the left column and click on Time url link to Open the Time Settings Page. Enter the relevant parameters on this page and press the Save button. SME VoIP System Guide, Version 1.5 Proprietary and Confidential t n e m e g a n a M p u t e S N A L V 81
r e t p a h C 12.7 VLAN Setup: Base station STEP 7 Navigate to the Network url > On the network page enter the relevant settings in the VLAN section > VLAN Id should be the same as those configured into the backbone. SME VoIP System Guide, Version 1.5 Proprietary and Confidential t n e m e g a n a M p u t e S N A L V 82
r e t p a h C 13 Multi-cell Setup & Management This chapter seeks to describe how to install, add and synchronize one or multiple base stations to the network. There are two main procedures involved:
1) Proper placement of the base stations (which is called network dimensioning). The present chapter does not address this issue. Refer to Chapter 4 for details. 2) Creating and adding base station profiles to the network via the SME Configuration Tool (to form a multi-cell system). This chapter describes the second procedure. This chapter is valid for Base station firmware version 00.49 and above. 13.1 SME Configuration Interface RTX have offered HTTP interface in base station firmware that can be used as HTTP Web Server. The SME Configuration Interface can be retrieved from this HTTP Web Server in each Base station. The HTTP Web Server is enabled in the base station by default. This procedure is valid for Base station firmware version 00.34 and above. The system administrator must update the relevant Base station(s) to the latest firmware before proceeding to the next section. Refer to Chapter 10. 13.2 Adding Base stations via SME Configuration Interface Here are the recommended steps to add Base stations to network:
STEP 1 Connect the Base station to a private network via standard Ethernet cable (CAT-5). STEP 2 Open any standard browser and enter the address:
http://ipdect<MAC-Address-Base-Station>
e.g. http://ipdect00087B00AA10. This will retrieve the HTTP Web Server page from the base station with hardware address 00087B00AA10. SME VoIP System Guide, Version 1.5 Proprietary and Confidential t n e m e g a n a M
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r e t p a h C 83 STEP 3 Skip this step if you were successful with STEP 2. If the MAC-address method as described above does not work, then use a standard network protocol analyzer (e.g. Wireshark) to eavesdrop the IP address allocated to the base unit by the DHCP server. You can download Wireshark follow its documentation to know how it is done. Enter the IP address into the address bar of the browser to retrieve base station HTTP Web Server Page. STEP 4 On the Login page, enter your authenticating credentials (i.e. username and password is default admin). Click OK button. STEP 5 Once you have authenticated, the browser will display front end of the SME Configuration Interface. The front end will show relevant information of the base station (in this case MAC-
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 84 13.2.1 Time Server Setup STEP 6 Navigate to the Time settings and configure it. Scroll on the left column and click on Time url link to Open the Time Settings Page. Enter the relevant parameters on this page and press the Save button. Make sure there is contact to the Time server otherwise the Multi-cell feature will not work. You can verify whether the Time server is reachable by rebooting the base station and making sure the correct Time Server IP address is still in place. t n e m e g a n a M
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r e t p a h C 85 A successful reboot will reset the time in the base station. Ensure the Time Server IP address is correct. 13.2.2 SIP Server (or PBX Server) Setup STEP 7 Create the relevant SIP server (or PBX Server) information in the system. Each service provider/customer should refer SIP server vendor on how to setup SIP servers. Click the link Server at the left hand column of home page, you can add your SIP server for base station use. Next, from the Server page, click on the Add Server URL and enter the relevant SIP server information (an example is shown below). Choose No on NAT adaption parameter if NAT function of the SIP aware router is not enabled. Enter the relevant parameters based on the description in the table below. Select Save button. t n e m e g a n a M
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 86 13.2.3 Multi-cell Setup STEP 8 Click on Multi Cell url link in the SME VoIP Configuration to view the current Multi cell settings status of the current base station. Most brand new base stations have Multi cell system feature disabled by default. t n e m e g a n a M
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r e t p a h C 87 STEP 9 Next, the system administrator needs to create and Enable Multi Settings profile for the current base station. On the Multi Cell settings Page, choose Enable option from the drop down menu of the Multi cell system parameter. Enable the Multi cell debug option if the system administrator wants some Multi-cell related logs to be catalogued by the system. STEP 10 On the same Multi Cell Settings page > Enter the relevant values for System chain ID and Synchronization time (s) respectively. The System chain ID is a geographically unique DECT cell identity allocated to bridge several base stations together in a chain. An example is 2275. The Synchronization time (s) parameter is defined as window/period of time in seconds a specific base station synchronises to the master base station unit (by default 60). Synchronising the slaves to the master updates them with the latest network information. t n e m e g a n a M
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r e t p a h C 88 Click on Save button to keep modified changes of multi cell settings into the base station. The Multi Cell data synchronization ONLY works when the relevant Time Server is set in the system before Server/Subscriber profile is added or created. Refer to STEP 6. IMPORTANT:
Base stations must be rebooted after the time server has been set. STEP 11 Repeat STEP 1-10 as explained above for each base stations. IMPORTANT:
It takes up to 5 minutes (synchronization time) to add a new base station to a Multi Cell System. t n e m e g a n a M
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 89 13.3 Synchronizing the Base stations STEP 12 On each SME VoIP Configuration interface for the base station(s) navigate to the Home/Status page and Click the Reboot button. This will trigger Are you sure you want to reboot gateway? window. Click OK button on this window. A successful restart of the base stations will lead to a display of the page: Gateway has been reset. STEP 13 Navigate back to the Multi cell settings page by clicking Multi-cell url link at the left column. The revised Multi cell settings page shows the relevant base stations synchronized together. By default, the system uses the first registered base station as the master base unit. t n e m e g a n a M
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r e t p a h C 90 STEP 14 On the Multi-cell settings page, scroll to the DECT system settings and Enable or Disable the Auto configure DECT sync option source tree (See description in the table below). The DECT system RFPI parameter is computed by the system (Its often greyed in a multi-cell system configuration). STEP 15 Scroll to the SIP system settings section, configure and save the parameter based on the description below:
Parameter
## of SIP accounts before distributed load SIP Support for multiple registrations per Description The maximum number of handsets or SIP end nodes that are permitted to perform location registration on a specific Base unit before load is distributed to other base units. Note: A maximum of 8 simultaneous calls can be routed through each Base units in a multi-cell setup. Permitted Input: Positive Integers (e.g. 6) Enable this option so it is possible to use same extension (i.e. SIP Account) on multiple phones (SIP end nodes). These phones will ring simultaneously for all incoming calls. When a phone (from a SIP account group) initiates a handover t n e m e g a n a M
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 91 account from Base X to Base Y, this phone will de-register from Base X, and register to Base Y automatically. Note: Choose Yes when the SIP server supports this feature otherwise choose No if the Sip server does not support this feature. Permitted Input: Yes, No STEP 16 Next, on the MAC-units in chains section, you can manually configure the synchronisation source tree of the multi-cell system. Multi-cell settings page, scroll to the DECT system settings and Enable or Disable the Auto configure DECT sync option source tree (See description in the table below). The DECT system RFPI parameter is computed by the system (Its often grayed in a multi-cell system) 13.4 Summary of Procedure Creating a Chain We enumerate the short version of how to add 3 base stations units in a multi-cell setup. This can be applied for up to 40 number of base units. The procedure below is valid ONLY for base station firmware version 00.36 and above. This procedure is divided into four (4) main stages. Apply this procedure if all base unit are straight from production. 13.5 Stage 1 Skip this stage if relevant base stations are already in the network. a) Add 3 base stations i.e. RFP1, RFP2, RFP3 > Disable the Multi cell system option and Save b) RFP1, RFP2, RFP3: Reboot from the HTTP SME Configuration Main Page c) RFP1, RFP2, RFP3: Default by pressing reset button 12-sec. 13.6 Stage 2 Choosing 1st base unit i.e. RFP1 as Master a) RFP1: Define Time server and Save from the Time page b) RFP1: Reboot t n e m e g a n a M
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 92 c) RFP1: Press Add server and define SIP server IP and Save from the Servers page d) RFP1: Multi cell system = enable and Save from the Multi-cell page e) RFP1: Reboot (Verify the message:
SYNCMGR: This base is ready to be master in a Chain From the debug logs) 13.7 Stage 3 Choose another base unit,RFP2 as Slave1 a) RFP2: Multi cell system = enable and Save b) RFP2: Reboot (Verify from Debug log SYNCMGR: This base is ready to join into another Chain) c) RFP1, RFP2: Wait 2min for stable Master-Slave chain (check for the message:
SYNCMGR: Socket#10 creation success from the debug log) d) After a successful restart, on each base [RFP1 and RFP2] Multi-cell page you will find the other base connected and synchronized (the IP status shows This Unit or Connected) to the system as illustrated below. You can register an arbitrary HS to the extension and verify whether its successful from the Home page and Handset UI t n e m e g a n a M
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r e t p a h C 93 13.8 Stage 4 Choose the 3rd base unit, RFP3 as Slave2 e) RFP3: Multi cell system = enable and Save f) RFP3: Reboot (Verify Debug log SYNCMGR: This base is ready to join into another Chain) g) RFP1, RFP3: Wait 2min for stable Master-Slave chain (SYNCMGR: Socket#10 creation success) h) RFP3: Check mark ID2/RPN08 and select dropdown 1 RPN: 04 and Save i) RFP3: Reboot (Confirm from the Debug logthe message:
SYNCMGR: Socket#8 creation success
) Multi-cell chain of 3 base stations has been created successfully. Next step involves adding extensions to the system. t n e m e g a n a M
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 94 13.9 Practical Configuration of Multi-cell System In chapter 3 we described different scenarios where the SME VoIP system can be deployed. In this chapter we describe what exactly to configure in the SME VoIP Configuration Interface ensure these scenarios really work. 13.9.1 Case ##1: Isolated Buildings The optimal configuration for isolated buildings is standalone base stations setting. In this setting, you must:
STEP 1 Using the figure below as illustration, log into the Configuration Interface of each base station. STEP 2 STEP 3 On the main page of the configuration interface, click Network URL > disable the Multi-cell Configure the Time Server, SIP Server, Extensions as described in the previous chapters. parameter of each base station > Save and Reboot each base to complete the Case ##1 setup. Disable Multi Cell option of Base Stations t n e m e g a n a M
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r e t p a h C 95 13.9.2 Case ##2: Location with co-located partners To illustrate this setup, two slave base stations are synchronised to one master base in the two office blocks. It is not necessary to deploy a dedicated Base unit at the car park area because it is likely no telephony traffic or call will be placed at the area. Here is diagram to illustrate Case ##2. The procedure:
STEP 1 STEP 2 On the Network page of each base define the DECT sync source settings as illustrated in the Follow the steps described in sections 13.4 to 13.8 table below. STEP 3 Save and reboot each base to complete case ##2 setup Multi Cell Page of Base Stations Recommended settings of MAC-units in Chain section of page (Other different settings exist):
RPN Ver MAC Addr IP Addr IP Status DECT sync source DECT Property 00 04 08 XX XX XX XX:XX:XX XXX.XXX Connected 1: RPN:04 XX:XX:XX XXX.XXX Connected 1: RPN:04 Master XX:XX:XX XXX.XXX Connected 1: RPN:04 t n e m e g a n a M
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r e t p a h C 96 13.9.3 Case ##3: Large to Medium Sized Enterprises In this scenario, we have five slave bases synchronised to one master base. The master base is located in office block B while the slave bases are spread across the whole enterprise. No base station is deployed in the lift because it has high attenuation properties that will drastically reduce radio signals. The procedure:
STEP 1 STEP 2 On the Network page of each base define the DECT sync source settings as illustrated in the Follow the steps described in sections 13.4 to 13.8 table below. STEP 3 Save and reboot each base to complete case ## 3 setup Multi Cell Page of Base Stations Recommended settings of MAC-units in Chain section of page (Other valid setting exists):
NOTE:
The number of chains cannot exceed 6 levels. RPN Ver MAC Addr IP Addr IP Status DECT sync source DECT Property 00 04 08 0C 10 14 XX XX XX XX XX XX XX:XX:XX XXX.XXX Connected 1: RPN:04 XX:XX:XX XXX.XXX Connected 2: RPN:08 XX:XX:XX XXX.XXX Connected 3: RPN:0C XX:XX:XX XXX.XXX Connected 4: RPN:10 XX:XX:XX XXX.XXX Connected 4: RPN:10 Master XX:XX:XX XXX.XXX Connected 4: RPN:10 t n e m e g a n a M
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 97 13.9.4 Case ##4: Large Enterprises at Different Locations In this scenario, multi-cell systems are deployed at different locations; geographically separated from each other. Each location has a master base station with more than one slave base synchronise to it. The procedure:
STEP 1 STEP 2 On the Network page of each base define the DECT sync source settings as illustrated in the Follow the steps described in sections 13.4 to 13.8 table below. STEP 3 Save and reboot each base to complete case ## 4 setup Multi Cell Page of Base Stations Recommended settings of MAC-units in Chain section of page (Other valid setting exists):
DECT sync Ver MAC Addr IP Addr IP Status RPN source 00 04 08 0C XX XX XX XX XX:XX:XX XXX.XXX Connected 0: RPN:00 XX:XX:XX XXX.XXX Connected 0: RPN:00 XX:XX:XX XXX.XXX Connected 1: RPN:04 XX:XX:XX XXX.XXX Connected 3: RPN:0C DECT Property Master for HQ Master for Offices t n e m e g a n a M
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 98 10 14 18 1C 20 XX XX XX XX XX XX:XX:XX XXX.XXX Connected 3: RPN:0C XX:XX:XX XXX.XXX Connected 4: RPN:10 XX:XX:XX XXX.XXX Connected 6: RPN:18 XX:XX:XX XXX.XXX Connected 6: RPN:18 XX:XX:XX XXX.XXX Connected 7: RPN:1C Master for Enterprises t n e m e g a n a M
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r e t p a h C 99 SME VoIP System Guide, Version 1.5 Proprietary and Confidential 14 Functionality Overview So far we have setup our SME VoIP system. Next, in this chapter we describe what features and functionalities are available in the system. The SME VOIP system supports all traditional and advanced features of most telephony networks. In addition, 3rd party components handle features like voice mail, call diversion, conference calls, etc. A brief description of SME VOIP network functionalities are:
Outgoing/incoming voice call management: The SME VOIP system can provide multiple priority user classes. Further, about 3 repeaters can be linked to a Base-station depending on customer requirements and assuming that the DECT RPN values are geographically unique. Internal/external handover: User locations are reported to SIP Server in order to provide differentiated services and tariff management. Within a DECT traffic area, established calls can seamlessly be handover between Base-stations using connection handover procedures. External handover are supported for terminals within roaming regions. Mobility: The network supports seamless mobility where handset subscription information controls allowed mobility. A user terminal can be assigned 2 mobility options:
- Mobility within in single DECT traffic area.
- Mobility within a group of DECT traffic areas. Registration procedure can be activated in one or multiple cells. Security: The RTX SME VOIP system also supports robust security functionalities for Base-stations. Most security1 functionality is intrinsically woven into the SME VOIP network structure so that network connections can be encrypted and terminal authentication can be performed. 14.1 System Feature List This section gives a summary of some essential functionality within the wireless IP network. Components System Features Speech Coding 10 channels ADPCM G.726 on air interface2 10 channels of G729a/b on IP interface3, 2 10 channels of G711 on IP interface2 Support of mixed types of Codecs in one Base Station In-band Tones Dial tone Busy tone Error tone Call waiting tone Messages waiting tone Ring Back tone 1 With active security 4 channels is supported 2 In a multicell configuration 8 channels is supported 3 G729a requires an additional hardware module. SME VoIP System Guide, Version 1.5 Proprietary and Confidential w e i v r e v O y t i l a n o i t c n u F
r e t p a h C 100 Components System Features Radio Access Mechanism Bearer Handover and Connection Handover:
Intra Cell Inter Cell RFP (Own/Other) Repeater, Repeater Repeater Busy indication and support Connection re-establishment Emergency Calls4: Inside or outside roaming areas SIP support REGISTER, INVITE, and TERMINATE sessions Session Description Protocol (SDP), HTTP authentication Support 200 DECT instances (depending on SME VOIP configuration) locating SIP servers Support for re-INVITE Support for fail-over SIP proxy Message Waiting Indication Internal Synchronization Support for 302 response between UA SIP Server Internal Synchronization lock, timing and transmission Management Features Assignment of Base-stations Logging calls and internal events, and tracking use of resources Logging system faults TFTP server for software upgrade. WEB interface for remote management of network devices Remote debugging of network devices, including log features 14.2 Detail Feature Description CODECs G.711 PCM A-law & U-law Uncompressed voice G.722 G.726 G.729 Silence suppression ( No) Allows HD sound for the handset
(ADPCM, 32 Kbps) A/AB (including VAD, CN generation ) G.729.1 (ehem. G.729 EV) Note: Only with additional module, this is a extra option that requires a board connector mounted in Gateway. Per default not mounted. SDP: Session Description Protocol Uniform Resource Identifiers (URI): Generic Syntax In-Band DTMF/Out of band DTMF support The Naming Authority Pointer (NAPTR) DNS Resource Record The SIP INFO method SIP 2.0 Reliability of Provisional Responses in the Session Initiation Protocol (PRACK) Locating SIP Servers (DNS SRV, redundant server support) SIP RFC2327 RFC2396 RFC2833 RFC2915 RFC2976 RFC3261 RFC3262 RFC3263 4 Emergency call is not possible if the Network connection is not working or in case of power failure. SME VoIP System Guide, Version 1.5 Proprietary and Confidential w e i v r e v O y t i l a n o i t c n u F
r e t p a h C 101 RFC3264 RFC3265 RFC3326 RFC3311 RFC3325 RFC3326 RFC3515 RFC3550 RFC3581 RFC3842 RFC3891 RFC3892 RFC3960 RFC4475 SIPS In-band DTMF SRTP Web server Other features Quality of service IP quality Import of phonebook Automatic DST Tone Scheme Ethernet features VLAN DHCP Support Static IP TLS TFTP HTTP HTTPS TCP/IP/UDP SNTP VPN Quality of service DHCP option DNS srv Offer/Answer Model with SDP Specific Event Notification The Reason Header Field for the Session Initiation Protocol The Session Initiation Protocol UPDATE Method P-Asserted Identity The Reason Header Field for the Session Initiation Protocol (SIP) REFER: Call Transfer RTP: A Transport Protocol for Real-Time Application Rport Message Waiting Indication Replace header support The Session Initiation Protocol (SIP) Referred-By Mechanism Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP) Session Initiation Protocol (SIP) Torture Test Messages Will limit number of active calls pr. base when enabled. Embedded web server HTTP/HTTPS Easy configuration of the phone, remote configuration via Management Interface Reasonable customization of the Web Interface to customer branding Password protection Status information on webpage. Minimum:
HW/FW version Serial number + MAC address Connected handsets with serial number SIP-status Connected DECT handsets Remote firmware update (HTTPS/TFTP) Type of Service (ToS) including DiffServ Tagging, and QoS per IEEE 802.1p/q NetEQ by GIPS (License applies) WiFi friendly Warning Network outage, VoIP service outage Packet Loss Concealment support Sequence Error Handling Adaptive Jitter Buffer support Import of phonebooks in csv format (Office 2003) Country Depend Tone Scheme VLAN (802.1p/q) For secure connections For configuration download. For configuration download. For secure configuration download. For internet clock synchronization Add-on in the future Type of Service (ToS) including DiffServ Tagging, and QoS per IEEE 802.1p/q 66 and 160 w e i v r e v O y t i l a n o i t c n u F
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 102 DECT DECT CAP CAT-IQ v1.0 General Telephony Handset Support VoIP Accounts Simultaneous Calls Call log Phone Book DND Call Forward Connectionless handover, enhanced location registration Wideband Speech 10 simultaneous handsets supported (single cell) 30 VoIP accounts per base (maximum 40 bases per installation) Maximum 200 handsets per installation 4 Wideband calls (g.722) or 10 single cell, 8 multi cell narrowband calls (PCMA, PCMU, G.726) or mixed wideband and narrowband. 50 mixed between Incoming, outgoing, missed calls Common Phonebook with up to 200 entries Do Not Disturb All No Answer Busy SME VoIP System Guide, Version 1.5 Proprietary and Confidential w e i v r e v O y t i l a n o i t c n u F
r e t p a h C 103 15 Network Operations 15.1 Introduction In this chapter, we will provide an overview of the operation of the network during system start-up, location registration and speech calls including illustration of different call scenarios. 15.2 System Start Up When a Base station Unit is powered up, it achieves IP address from DHCP server and time from the Time-
Server. Optionally the base retrieves its configuration from a file on the TFTP server. This configuration file describes used network and cluster configuration parameters (optional and not needed). The SME VoIP network has successfully started up. 15.3 Terminal Attachment When a DECT Terminal (also called handset or SIP node) is turned on or moved into the coverage area of a Base-station it has to get attached to the network. When more Base-stations are available, the Terminal selects the one with best RF signal. This procedure, called Location Registration, always keeps the network informed about where a Terminal is located and enables it receive or originate calls. This procedure also authenticates the Terminal and checks the validity of the associated subscription. 15.4 Outgoing Calls Outgoing calls are initiated by the Terminal. It selects the Base-station with best RF signal and establishes a radio communication link to Base-station. DECT call control messages are exchanged between Terminal, Base station and other servers. This server forwards the outgoing call as SIP messages to the external SIP Server. The RTP stream is established between the involved Base-station (and the Media Gateway for PSTN calls). If the call is between two Terminals the media stream may be routed directly between the two involved Base-stations depending on the SIP Server routing strategy. 15.5 Incoming Calls Incoming calls are initiated by SIP INVITE messages from the SIP Server to the Base unit; inviting it to participate in an incoming session. The system sends paging messages to all the Base-station where the Terminal last performed a Location Registration. When the paging is received the Terminal establishes a radio communication link to the best available Base-station and sends a response back to DECT controller. DECT call control messages are exchanged and the Terminal starts ringing. When the user answers the call, a connect message is sent to the IP DECT controller that completes the incoming call by sending 200 OK back to the SIP Server and establishes an RTP media stream between Base-station (and Media Gateway from PSTN line). For internal calls the media stream may be routed directly between the involved Base-
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 104 15.6 Handover 15.6.1 RTP Stream Remains at Initial Base Station When the call is set up, the handset is located at base station 1. Thus, the DECT communication takes place between the handset and station 1, and the SIP signalling as well as the RTP stream takes place between base station 1 and the SIP server/media gateway. The figure below illustrates this application:
Stage 1: Before handover the handset is located at BS 1. After handover, the handset is located at base station 2, and hence the DECT communication goes on between the handset and base station 2. However, to avoid disruption of the audio, the RTP stream is relayed via the initial base station, since a transfer of the RTP stream to another base station may cause the media gateway (or whatever the remote endpoint is) to re-initialize the RTP stream with a small disruption of the RTP stream as consequence. Thus, from the point of view of the remote endpoint, the RTP stream is not affected by the handover, and since the call control also remains at base station 1 the SIP signalling is also unaffected, as shown below:
Stage 2: After handover to BS 2, the HS is located at BS 2, and the RTP stream is relayed via BS 1. Since the call control and hence the SIP User Agent remains at the initial base station, the SIP registration is also unaffected by the handover. SME VoIP System Guide, Version 1.5 Proprietary and Confidential 105 s n o i t a r e p O k r o w t e N
r e t p a h C SIP Server/Media GatewayBase station 1DECT ConnectionRTP StreamSIP SignalingBase station 2Base station n If the handset makes yet another handover, the RTP stream will still be relayed via the base station at which, the call was established (here base station 1). This is illustrated as follows:
After handover to BS n the handset is located at BS n, and the RTP stream is relayed via BS 1. 15.7 Roaming By roaming means the handset moves its SIP and DECT registration from one base station to another base station. Roaming can only be initiated from idle. Roaming does not immediately result in a new SIP registration, because this may cause a lot of unnecessary signalling. Therefore, the handset will not perform a new DECT Location Registration until it has resided on the same base station for a defined period of time. Since the SIP registration is initiated by the completion of the Location Registration, a new SIP registration will also not be done until this procedure has completed on a new base station. Thus, a handset must stay on the same base station as given in the rules stated below, before a new SIP registration will be made. Timing Criteria for Location Registration; or roaming will be initiated when:
1. Handsets lose contact to first base unit due to reset/power off/heavy DECT traffic. 2. After 5 minutes (configuration is possible) but before 5+2 minutes a. The plus maximum 2 minutes will occur when service connection traffic is signaled at the same time as location should happen. In this case the location registration procedure will be delayed. If an incoming call arrives while the handset has moved to another base station (base station 2) but still not performed a new Location Registration, the SIP call will arrive at the initial base station (base station 1), but the RTP stream will be set up between base station 2 and the remote endpoint (refer to figure below). Alternatively, in the case of an outgoing call, the SIP call will be established from the initial base station, and the RTP stream will be set up between base station 2 and the remote endpoint. s n o i t a r e p O k r o w t e N
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 106 An illustration of Handset moving to another base station, but call control is still handled by the initial base station. s n o i t a r e p O k r o w t e N
r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 107 SIP Server/Media GatewayBase station 1Base station 2Base station nRTP StreamDECT ConnectionSIP ControlAudio Control 16 Operation Setup Bases/Handsets/SIP Sever In this chapter, we describe the operation of the base unit and handsets during power up/down. Next we describe the signal flow graphs for some selected operations of the system. 16.1 Power Up Handset The handset is still off after the battery has been put into the Handset. The user has to Long Key Press the Red-key (On-hook key) to power up the handset. If the handset is registered, it will start searching for a base and if the handset is in range of a base, then it will location register to the base. The handset will display Unregistered if the handset isnt registered. If the handset is not registered, then the user has to enable the registration mode on the base and afterwards register the handset by selecting Connectivity Register <type in Access code> *OK+ on the MMI of handset. The handset supports connectionless handover between bases and inter-cell handover between a base and repeaters. Base Station Power is supplied to the base (gateway) via PoE. The base will start operation when connected to the LAN or WAN. The wireless handset must be registered to the base and a SIP proxy must be configured on the base (refer to Chapter(s) 11,13 for details). The registration mode must be enabled to make it possible to register a handset to the base. The registration mode is enabled on the base by accessing the web-interface on the base. This is done by typing in the IP address of base in the browser on a computer connected to the same network as the base. The user will be prompted for a user name and Password to access the web interface on the base. The user selects an extension via the web interface and click Register. The base will afterwards be open for registrations for 5 minutes. Up to 40 bases can be linked together and synchronisation between the bases is done via the air and the network. The base supports up to 10 SIP-proxies and 30 handsets. 16.2 Power Down Handset If the Handset loses battery or the battery level is less than a threshold that is in the EEPROM, it goes to power down mode. A battery warning will be given before the handset power down. Base Station One of the bases will be a master base and another base will work as fall back base in case of power failure on the master base. All handsets located on a base will be un-located in case of power down of the base. In case of power failure on one of the bases, the user can move to another base area and the handset will then locate on this base and be workable on the new base. r e v e S P I S
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 108 16.3 Call Operations In this section we describe call procedures in the SME VoIP system. We will narrow our description to call transfer and call conference. Please view accompanied document for detail description of Handset operations [3]. Some Definitions There are three actors in a given transfer event, each playing one of the following roles:
Transferee: the party being transferred to the Transfer Target. Transferor: the party initiating the transfer Transfer Target: the new party being introduced into a call with the Transferee. The following roles are used to describe transfer requirements and scenarios:
1. Originator - wishes to place a call to the Recipient. This actor is the source of the first INVITE in a session, to a Facilitator or a Screener. 2. Facilitator - receives a call or out-of-band request from the Originator, establishes a call to the Recipient through the Screener, and connects the Originator to the Recipient. 3. Screener - receives a call ultimately intended for the Recipient and transfers the calling party to the Recipient if appropriate. 4. Recipient - the party the Originator is ultimately connected to. Call Transfer - Requirements Any party in a call session is able to transfer any other party in that session at any point in that session. The Transferor and the Transferee are not removed from a session as part of a transfer transaction. This requirement is needed so e.g. ring-back on transfer failure will not be lost. The Transferor is aware of whether or not the transfer was successful 16.3.1 Initiating Calls Enter Number Press Green Button to start dialling Originating Party SME VoIP System Guide, Version 1.5 Proprietary and Confidential 109 r e v e S P I S
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r e t p a h C Destination Party The destination party must press the Green button available on its handset to accept the incoming call or reject to disallow the call. 16.3.2 Call Holding Press the Hold option at the left while call session is in progress or Connected, Press Retrieve option to re-connect the call placed on hold SME VoIP System Guide, Version 1.5 Proprietary and Confidential 110 r e v e S P I S
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r e t p a h C 16.3.3 Call Transfer (Blind) While Connected, press the Hold option, to put the call session on hold (refer to section 16.3.3 Connected screen shots). Enter the transfer destination number Next, Press the Transfer option to transfer call session from Originator to the Target Transfer. The Facilitator handset performs transfer procedure and returns to Idle mode (On hook mode) Facilitator 16.3.4 Call Bridging (Attended Transfer) While Connected, press the Hold option, to put the call session on hold (refer to section 16.3.3 Connected screen shots). Enter the transfer destination number and press Green button to connect/establish call session between Facilitator and the destination party (i.e. Line 2). Next, Press the Transfer option to transfer call session from Originator (i.e. Line 1) to destination party i.e. Line 2 Handset performs transfer procedure and returns to Idle mode (On hook mode) i.e. the facilitator handset. SME VoIP System Guide, Version 1.5 Proprietary and Confidential 111 r e v e S P I S
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r e t p a h C 16.3.5 Call Conference (Conference) While Connected, press the Hold option, to put the call session on hold (refer to section 16.3.3 Connected screen shots). Enter the transfer destination number Press the Green button to connect/establish call session between Facilitator and Line 2 i.e. the destination party. Next, Press the Conf. option to establish conference call session between all dialled parties i.e. Originator, Facilitator and Destination party. SME VoIP System Guide, Version 1.5 Proprietary and Confidential r e v e S P I S
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r e t p a h C 112 17 Handset - Service Menu Management We seek to describe how to use the Service Menu feature available only to vendors and developers of the handset. The document will also describe some options available in the Service Menu. 17.1 Service Menu Site Survey Mode This is normally reserved and used to reveal features not used/seen by the end-user. By means of a special key sequence a special service menu can be accessed. This special menu enables some special feature like Master reset, Site survey mode, Handset logs, Status, IPEI and Demo mode. To access the service menu, follow these steps:
STEP 1 Click on Menu from the Handset > Type *SERVICE* or *7378423* from the keypad to display the Service Menu STEP 2 On the Service Menu scroll down to the Site survey mode > Enable the Site survey mode to switch from Off to On. This sets the handset in a state to iteratively scan other handsets around it and/or chained to the same base station(s). Handsets in site survey mode can display up to 5 other handsets with the strongest signal strength. NOTE: If you set the handset to wideband mode, it will not possible to change back SME VoIP System Guide, Version 1.5 Proprietary and Confidential t n e m e g a n a M u n e M e c i v r e S
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r e t p a h C 113 17.2 Service Menu Parameter Definitions The Service Menu is not for end users only installers and suppliers. 17.2.1 Master Reset This feature allows the user to clear any pending errors or events and bring a Handset to normal condition and initial state in a controlled manner. Valid inputs: Yes or No Screen shots 17.2.2 Site Survey Mode In Site Survey Mode the handset MMI shows the RFP (including slave RFP) to which the handset is locked to and the corresponding RSSI. Valid Inputs: On or Off Line1:
Line2:
Line3:
Parameter/Line Description 28 20 03 56 84 78 FP: 4 RPN
-dBm FE PP: 1 RPN
-dBm (Signal Strength) FE PP:XX FP:XX The line contains the list of base stations identified by the handset or RPNs of the RFP-table entries in respect to the RSSI-values below. Up to 5 RPNs can be displayed. The Radio Fixed Part Number (RPN) is an 8-bit DECT cell identity allocated by the installer. The allocated RPN within a cluster must be geographically unique. This indicates the actual field strength information (RSSI) for the base station the current handset is locked to and additional RFPs stored in the RFP-table. RSSI unit is -dBm. The RSSI value of the RFP-table is updated in a 250ms cycle. This causes an update of any entry in a table with 3 RFPs every 3 * 250ms =
750ms. If the RFP could not be synchronized for an RSSI-update, the RSSI value is decreased. This implies that an entry will be deleted after a while, if e.g. the dummy-bearer position has changed or the RFP is unreachable. Indicates the number of sync/CRC errors (frame-errors) within the last update cycle. This information is only valid for the existing link to the current handset located to the relevant base station. The PP value is the number of detected Sync/CRC error(s) within the last 100 t n e m e g a n a M u n e M e c i v r e S
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 114 17.2.3 HS Logs receiving frames (per sec.). The FP value is the number of received Q1/Q2 bit information within the last 100 receiving frames (per sec.). This information is interpreted as Sync and CRC errors on the base station receiving side. The HS log is a debugging feature that allows the user to retrieve low level interesting messages from the handset. HS Logs Log error Description These are debug error logs retrieved from the PP log file. The last log retrieved is formatted into:
Error: This error code. Line: Location within the software code which triggered this error. Address: Register bank and/address from which error occurred. This reports of the function/exception handler that run after an erroneous state in handset operation. Next assert ID: Immediate exception handler ID to be run when a specific error occurs. Assert address: Register bank and/address where an exception handler is executed as a result of the error which occurred. Log Assert Screenshots t n e m e g a n a M u n e M e c i v r e S
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r e t p a h C SME VoIP System Guide, Version 1.5 Proprietary and Confidential 115 17.2.4 Status This provides the present condition of the handset and the base station it is location registered to. Some of the information available in this mode is described in the table below (this information is updated during location request update from the DECT system). Parameter Description SW Version Base station: Current firmware installed on the current Base station the Handset is location registered to. Handset: firmware presently installed in the handset. Format: Version Date Stamp (e.g.: 00.21 01-07-2010 00:00) HW Version Current hardware module used in base station and handset. DECT Band Operating frequency of system. DECT band includes UA, EU, LTAM, SA, N.A options. IP Address IP address of base station and handset MAC Address HW address of base station. System Name Name describing the SME network. Typically a string of 16 bytes. Battery Level Current handset battery energy position 17.2.5 IPEI The IPEI (International Portable Equipment Identity) is a unique identification of portable part (handset) and DECT Repeater. The IPEI is formatted and displayed in HEX, OCT, and OCT nomenclatures. t n e m e g a n a M u n e M e c i v r e S
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r e t p a h C 17.2.6 Demo mode The demo mode uses animations to show the user which functions the DECT phone offers (with/without connection to the SIP server). t n e m e g a n a M u n e M e c i v r e S
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r e t p a h C Appendix t n e m e g a n a M u n e M e c i v r e S
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r e t p a h C 18 Appendix A Handset A x i d n e p p A
r e t p a h C 119 SME VoIP System Guide, Version 1.5 Proprietary and Confidential Base Station Web interface A x i d n e p p A
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This product uses the FCC Data API but is not endorsed or certified by the FCC