Basic VoIP

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Configuring Basic Voice over IP and

Fax over IP Solutions

September 2007

Document No. DC-4272-10

Gilat Satellite Networks Ltd.

This document contains information proprietary to Gilat Satellite Networks Ltd. and may not be reproduced in whole or in part without the express written consent of Gilat Satellite Networks Ltd. The disclosure by Gilat Satellite Networks Ltd. of information contained herein does not constitute any license or authorization to use or disclose the information, ideas or concepts presented. The contents of

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Contents

1. Introduction ...1

1.1 General Overview...1

1.2 SkyEdge VoIP Network...1

1.3 SkyEdge VoIP Network Components...3

1.3.1 VoIP Gateway ...3

1.3.2 VoIP Gatekeeper (GK) Router/SIP Proxy ...5

1.3.3 Provisioning (TFTP) Server ...6

1.3.4 Remote VoIP Gateway (ATA)...6

1.4 SkyEdge VoIP Solution Description ...7

1.4.1 Voice over IP in SkyEdge Mesh Systems ...8

1.4.2 Quality of Service...8

1.5 SkyEdge VoIP Calculations ...8

1.5.1 Calculation Algorithm ...9

1.5.2 G.729 Codec Example ...10

1.5.3 G.723 Codec Example ...11

1.6 Basic VoIP Network Call Flow ...11

1.7 Approved VoIP Components...14

1.8 Requirements ...14

1.9 System Limitations ...14

2. SkyEdge Basic VoIP Main Features...16

2.1 H323 Implementation in SkyEdge ...16

2.1.1 Overview...16

2.1.2 Operation Modes...17

2.1.3 Direct Mode ...17

2.1.4 Routed Mode ...18

2.2 SIP Implementation in SkyEdge...19

2.2.1 Overview...19

2.2.2 SIP Proxy...21

2.3 Compressed RTP (cRTP) Implementation in SkyEdge...21

2.3.1 Configuration Notes ...23

2.4 Fax over IP...23

3. Cisco ATA Basic Configuration ...24

3.1 Overview ...24

3.2 Establishing Connection between the VSAT/PC and the ATA ...24

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4. H323 Configuration on the SkyEdge NMS ... 30

4.1 H323 Direct Mode Configuration ... 30

4.1.1 Procedure Overview ... 30

4.1.2 Configuring DPS Parameters... 31

4.1.3 Configuring TCP Timers in the VSAT Data Template... 33

4.1.4 Configuring DA Parameters in the VSAT Data Template... 35

4.1.5 Configuring VoIP Filters for H323 Direct Mode ... 43

4.1.6 Configuring Port Priorities in the VSAT Data Template ... 45

4.1.7 Saving and Committing VSAT Template Configuration ... 50

4.1.8 Configuring VSAT Unique Parameters... 50

4.2 H323 Routed Mode Configuration ... 56

4.2.1 Procedure Overview ... 56

4.2.2 Configuring VoIP Filters for the H323 Routed Mode... 57

5. H323 Configuration on the Cisco ATA ... 60

5.1 Accessing the Web Interface and Checking the ATA Version... 60

5.2 Configuring the Cisco ATA for H323 Using the ATA Web Interface ... 61

6. SIP Proxy Configuration on the SkyEdge NMS ... 68

6.1 Procedure Overview... 68

6.2 Configuring SIP Proxy VoIP Filters in the VSAT Data Template ... 68

6.3 Configuring SIP Port Priorities in the VSAT Data Template ... 71

7. SIP Configuration on the Cisco ATA... 75

7.1 Accessing the Web Interface and Checking the ATA Version... 75

7.2 Configuring the Cisco ATA for SIP via ATA Web Interface ... 76

8. Compressed RTP Configuration on the SkyEdge NMS... 83

8.1 Procedure Overview... 83

8.2 Enabling cRTP at the DPS ... 83

8.3 Enabling cRTP Support at the VSATs ... 84

9. Fax Support Configuration on the SkyEdge NMS ... 87

9.1 Procedure Overview... 87

9.2 Configuring Fax Support in a System with Dynamic DA ... 87

9.3 Configuring Fax Support in a System with Legacy DA... 90

10. Appendix A - Remote Gateway Configuration and Basic Commands... 94

10.1 Remote Gateway Configuration ... 94

10.1.1Voice Configuration Menu... 95

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10.2 Remote Gateway Commands ...99

10.2.1Refreshing the Cisco ATA Configuration ...99

10.2.2Resetting the Cisco ATA ...100

10.2.3Resetting the Cisco ATA to Factory Default Values...100

11. Appendix B - SkyEdge Basic VoIP System Debugging ...101

11.1 General System Debugging ...101

11.1.1Using a Sniffer Application ...101

11.2 VoIP Gateway & VoIP Gatekeeper Commands...103

11.2.1Logging on to the VoIP Gateway/Gatekeeper Command Line Interface...103

11.2.2VoIP Gateway Maintenance Commands ...104

11.2.3VoIP Gatekeeper Maintenance Commands ...108

12. Appendix C- Upgrading the Cisco ATA Software Version...111

12.1 ATA Upgrade Kit/Folder...111

12.2 Upgrading ATA ...112

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Figures

Figure 1: SkyEdge VoIP Network Architecture ... 2

Figure 2: VoIP Gateway and Gatekeeper (Cisco 2611 XM Modular Multiservice Router) . 4 Figure 3: One and Two Port E1 Multi-Flex Trunk Voice/Wan Interface Cards ... 4

Figure 4: Remote Gateway (Cisco ATA) and Telephone (Front View)... 6

Figure 5: Remote Gateway (Rear View)... 6

Figure 6: SkyEdge VoIP Call Establishment ... 13

Figure 7: Data Flow in H323 Direct Mode ... 18

Figure 8: Data Flow in H323 Routed Mode ... 19

Figure 9: SIP Data Flow... 21

Figure 10: RTP and cRTP Flow ... 22

Figure 11: Cisco ATA Connections ... 25

Figure 12: Cisco ATA Function Button ... 26

Figure 13: Cisco ATA Rear Panel ... 27

Figure 14: Cisco ATA Function Button ... 28

Figure 15: DPS Configuration ... 31

Figure 16: DPS TCP Parameters ... 32

Figure 17: VSAT TCP Profile Parameters ... 34

Figure 18: VSAT Templates... 36

Figure 19: VSAT Data Template ... 36

Figure 20: Data Template Dynamic Parameters... 37

Figure 21: VSAT Data Template – RT Applications... 37

Figure 22: VSAT Templates... 38

Figure 23: VSAT Data Template ... 39

Figure 24: VSAT Data Template – Legacy ADA... 39

Figure 25: VSAT Data Template – ADA ROT Parameters... 41

Figure 26: Legacy ADA - RT Applications Parameters ... 42

Figure 27: VoIP Filter Instances... 43

Figure 28: New VoIP Filter... 44

Figure 29: IP Classifier Template Table Instances ... 45

Figure 30: IP Classifier Entries ... 46

Figure 31: Configuring a New Entry in the IP Classifier... 46

Figure 32: IP Classifier Table Configured ... 48

Figure 33: CS1-CS7 Class Descriptors... 49

Figure 34: VSAT Data Parameters... 52

Figure 35: VSAT Licenses ... 53

Figure 36: VSAT IP Prioritization ... 54

Figure 37: VSAT Prioritization Parameters ... 55

Figure 38: VoIP Filter Instances... 57

Figure 39: New VoIP Filter for the H323 Routed Mode ... 58

Figure 40: H323 ATA Configuration Screen (Upper Part)... 60

Figure 41: ATA Configuration Screen for H323 ... 61

Figure 42: ATA Version and Configuration Information ... 61

Figure 43: ATA H323 Configuration ... 62

Figure 44: Applying Configuration Changes... 67

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Figure 46: VoIP Filter Instances ...69

Figure 47: Adding a New VoIP Filter ...70

Figure 48: Adding a New SIP VoIP Filter...70

Figure 49: IP Classifier Template Table Instances ...72

Figure 50: IP Classifier Entries...72

Figure 51: Configuring a New Entry in the IP Classifier ...73

Figure 52: IP Classifier Table Configured ...74

Figure 53: SIP ATA Configuration Screen (Upper Part) ...75

Figure 54: Cisco ATA SIP Configuration...76

Figure 55: ATA Version and Configuration Information...76

Figure 56: ATA Configuration for SIP ...77

Figure 57: Applying Cisco ATA Configuration Changes ...82

Figure 58: Cisco ATA Configured Successfully...82

Figure 59: DPS Data IP Parameters...83

Figure 60: DPS cRTP Parameters...84

Figure 61: VSAT Data Template cRTP Parameters ...85

Figure 62: Data Template Dynamic Parameters ...88

Figure 63: Dynamic DA Parameters ...89

Figure 64: Dynamic DA – Extra Bandwidth Reservation Parameters ...89

Figure 65: VSAT Data Template – Legacy ADA ...91

Figure 66: Legacy ADA - RT Applications Parameters ...91

Figure 67: Cisco ATA Web Configuration Menu (Partial View)...98

Figure 68: Sample Sniffer Capture ...103

Figure 69: sh9voice9dsp command ...105

Figure 70: sh9voice9call command...106

Figure 71: sh9h3239gateway9cause-codes command ...107

Figure 72: sh9gatekeeper9calls command...109

Figure 73: sh9gatekeeper9endpoints command ...110

Figure 74: ATA Software Upgrade Kit/Folder...111

Figure 75: Upgrade Commands File Content...112

Figure 76: Copying Upgrade Commands...113

Figure 77: Upgrading ATA...114

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Tables

Table 1: Voice Configuration Menu Commands ... 26

Table 2: Voice Configuration Menu Commands ... 28

Table 3: Voice Configuration Menu Commands (cont’d) ... 29

Table 4: H323 IP Classifier Instances ... 47

Table 5: ATA Configuration for H323 ... 63

Table 6: SIP IP Classifier Instances... 73

Table 7: ATA Configuration for SIP... 78

Table 8: Voice Menu Basic IP Connectivity Parameters... 95

Table 9: Voice Menu Alphanumeric Characters ... 97

Table 10: VoIP Gateway Show Commands... 104

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About This Manual

This section describes the objectives, audience, document layout and conventions of the SkyEdge Voice over IP Network Configuration manual.

Objectives

This manual provides a description of the SkyEdge Voice over IP network and gives you instructions on how to configure the VoIP feature in the SkyEdge networks.

How to Use This Manual

This manual is to be used as a reference guide by trained personnel. It is not recommended for use by personnel who have not previously attended a training course.

Audience

This manual is designed for operations personnel who have been trained in the operation of the SkyEdge NMS.

Organization

The table below contains a list of the chapters in the manual, the chapter titles and a short description of the material contained in each chapter.

Chapter Chapter Title Description

Chapter 1 Introduction Provides an introduction to the SkyEdge VoIP system.

Chapter 2 SkyEdge Basic VoIP Main Features

Explains the main features of the SkyEdge Basic VoIP solution. Chapter 3 Cisco ATA Basic

Configuration

Provides detailed instructions on how to establish connection with Cisco ATA.

Chapter 4 H323 Configuration on the SkyEdge NMS

Explains how to configure the SkyEdge NMS in order to support H323.

Chapter 5 H323 Configuration on the Cisco ATA

Provides detailed instructions on how to configure Cisco ATA to support the SkyEdge H323 solution.

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Chapter Chapter Title Description Chapter 6 SIP Proxy Configuration on

the SkyEdge NMS

Explains how to configure the SkyEdge NMS in order to support SIP.

Chapter 7 SIP Configuration on the Cisco ATA

Provides detailed instructions on how to configure Cisco ATA to support the SkyEdge SIP solution.

Chapter 8 Compressed RTP Configuration on the SkyEdge NMS

Explains how to configure cRTP on the SkyEdge NMS.

Chapter 9 Fax Support Configuration on the SkyEdge NMS

Provides detailed instructions on how to configure SkyEdge NMS to support the Fax over IP solution.

Chapter 10 Appendix A – Remote Gateway Configuration and Basic Commands

Explains Cisco ATA configuration options and lists some of the most useful commands.

Chapter 11 Appendix B – SkyEdge Basic VoIP System Debugging

Provides some debugging

information, including Gateway and Gatekeeper commands and Sniffer usage.

Chapter 12 Appendix C - Upgrading the Cisco ATA Software Version

Explains how to configure the Cisco ATA software version.

Chapter 13 Appendix D - Glossary Provides definitions for the technical terms used in this manual.

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Conventions

This manual uses the following conventions to convey instructions and information:

Convention Description

Boldface font Commands and keywords.

Italic font The result of an instruction or command.

Screen font Information to be typed into a form or dialog box.

9 Indicates a space in a CLI command.

WARNING

This warning symbol means danger. It is used to describe a situation that can cause bodily injury. Before working on any equipment, be aware of the hazards involved with electrical circuitry and how to prevent accidents.

CAUTION

This symbol means reader be careful. In this situation, damage may be caused to equipment or data may be lost.

NOTE

This symbol means reader take note. Notes contain helpful suggestions and explanations.

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1. Introduction

This document describes SkyEdge Basic VoIP implementation in SkyEdge versions 4.2 and higher.

This section describes the following:

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General Overview

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SkyEdge VoIP Network

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SkyEdge VoIP Network Components

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SkyEdge VoIP Solution Description

ƒ

SkyEdge VoIP Calculations

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Basic VoIP Network Call Flow

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Approved VoIP Components

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System Limitations

1.1 General Overview

VoIP (voice over IP - that is, voice delivered using the Internet Protocol) is a term used in IP telephony for a set of facilities for managing the delivery of voice information using the Internet Protocol. In general, this means sending voice information in digital form in discrete packets rather than in the traditional circuit-committed protocols of the public switched telephone network (PSTN).

1.2 SkyEdge VoIP Network

The SkyEdge VoIP solution includes VoIP phone and Residential Gateway (RGW). The telephone handset is used in its standard manner and is connected to the RGW. The RGW (Cisco ATA) provides the interface between the analog telephone set and/or Fax and VoIP. VSAT communicates with the hub via a full duplex satellite channel, transferring VoIP packets to and from the hub.

The VoIP packets destination can be one of the following:

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Residential Gateway (another VSAT)

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Hub Gateway (VoIP Gateway) – converts the VoIP packets back to legacy digital voice signal (PCM), carried over E1 lines to the public switch of the local PSTN, thereby enabling ordinary telephone communications with any other telephone subscriber in the world.

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The Hub can communicate with an ISP connection to provide VoIP communication over the Internet.

In the SkyEdge Basic VoIP solution, the VoIP data (signaling and voice media) is treated as a regular application and VoIP data runs on the same data channel (DA) as other kinds of applications, such as FTP, and so on. In this mode, the VoIP traffic is transparent to the satellite environment.

The SkyEdge Basic VoIP solution supports resource allocation to SIP and H323 signaling type. RTP compression for both signaling types is also supported. The same VSAT can support the H323 and SIP signaling.

Figure 1 shows a schematic view of the SkyEdge Basic VoIP network.

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A SkyEdge VoIP network consists of the following major components:

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A master earth station and control facility or hub.

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A Network Management System to control the system.

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A number of VSATs, located at the customer’s remote sites. These transmit either on Ku, C or extended C-band satellite channels to provide the transmission medium interconnecting the hub and the VSATs. Each VSAT has a Residential Gateway (e.g., Cisco ATA) connected to it, enabling VoIP. SkyEdge Pro and SkyEdge IP are the two VSAT types that can be used for the Basic VoIP solution.

1.3 SkyEdge VoIP Network Components

The standard SkyEdge VoIP Network contains the following components:

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At the Hub: − VoIP Gateway − VoIP Gatekeeper

− Provisioning server (TFTP)

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At the VSAT site:

− Remote VoIP Gateway (Cisco ATA-186) − (Analog) Telephone device

1.3.1 VoIP Gateway

The VoIP Gateway (GW) is installed at the hub and connects to the DPS via LAN. On the Inbound, the VoIP Gateway receives IP data packets from the Remote VoIP GW and forwards them to the PSTN or PBX. On the Outbound, the VoIP GW

receives the voice data from the PSTN/PBX and forwards it to the Remote VoIP GW. In a standard SkyEdge VoIP network, Cisco 2611XM Modular Multiservice Router is used as a VoIP Gateway. Cisco 2611 XM router supports 30 simultaneous calls.

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Figure 2: VoIP Gateway and Gatekeeper (Cisco 2611 XM Modular Multiservice Router)

In networks with more VoIP traffic the following routers can be used as VoIP Gateways:

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Cisco 3725 router to support 60 simultaneous calls

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Cisco 3745 router to support 120 simultaneous calls

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Cisco 5400 router to support 240 simultaneous calls

Cisco 2600 series routers support three basic voice Interface card types:

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E1 - An E1 line can carry 30 digitized voice channels

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FXO---Foreign Exchange Office interface. The FXO interface is an RJ-11 connector that allows a connection to be directed at the PSTN central office (or to a standard PBX interface, if the local telecommunications authority permits). This interface is of value for off-premise extension applications.

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FXS---The Foreign Exchange Station interface. This interface is an RJ-11 connector that allows connection for basic telephone equipment, keysets, PBXs, and supplies ring, voltage, and dial tone.

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NOTE

The VoIP Gateways are delivered pre-configured to the customer premises. Their configuration is beyond the scope of this manual.

For more information about the devices that have been tested and approved for use as VoIP Gateways, refer to Section 1.7, page 14.

1.3.2 VoIP Gatekeeper (GK) Router/SIP Proxy

The Gatekeeper/SIP Proxy is used for management of the Voice over IP traffic in the SkyEdge networks.

The Gatekeeper performs the following functions:

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Translates phone numbers into IP addresses and vice versa.

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Assigns resources to calls

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Monitors the operation of Remote Gateways

In a standard SkyEdge H323 networks, Cisco 2611XM Modular Multiservice Access Router is used as a Gatekeeper (Figure 2).

NOTE

The VoIP Gatekeeper is delivered pre-configured to the customer premises. Its configuration is beyond the scope of this manual.

For more information about the devices that have been tested and approved for use as VoIP Gatekeepers, refer to Section 1.7, page 14.

In SkyEdge SIP VoIP Networks, SIP Proxy server can be installed on any machine that answers the following minimum requirements:

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Pentium IV

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0.5 G RAM memory

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Standard VGA card

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Minimum of 20 G Byte hard disk space

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1.3.3 Provisioning (TFTP) Server

The Provisioning TFTP server is an optional component. The TFTP server is

installed at the Hub and is used for automatic configuration of the Remote Gateways (Cisco ATA 186).

1.3.4 Remote VoIP Gateway (ATA)

The Remote VoIP GW (Connection End Point) is installed at the remote VSAT sites and connects the VSAT and the analog telephone device. On the Inbound, the

Remote VoIP Gateway converts voice data into IP data packets and forwards it to the VoIP Gateway. On the Outbound, the Remote VoIP GW translates the IP data into analog data and forwards it to the telephone device.

Cisco ATA (Analog Telephone Adapter) 186 serves as a Remote VoIP Gateway in the SkyEdge VoIP networks. The Remote Gateway is installed at remote sites, it connects to the analog telephone via RJ-11 connector and to the 4-port VSAT via RJ-45 (LAN connection).

Figure 4: Remote Gateway (Cisco ATA) and Telephone (Front View)

Figure 5: Remote Gateway (Rear View)

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Cisco ATA 186 has the following characteristics:

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(A) - Two standard telephone voice ports (RJ-11 FXS)

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(B) One 10Base-T Ethernet port (RJ-45)

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(C) 12V power connector

For more information about the devices that have been tested and approved for use as Remote Gateways, refer to Section 1.7, page 14.

1.4 SkyEdge VoIP Solution Description

This section lists the main features of the SkyEdge VoIP solution:

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The connection end points (Cisco ATA) communicate with each other through Gatekeeper (H323) or SIP Proxy (SIP).

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Since VoIP is a CBR (Constant Bit Rate) application, the H323 and SIP support includes an algorithm that requests for APDA (Automatic PDA allocation) triggered by configurable set of characteristics of the TCP (H.323) or UDP (SIP) connections being opened (e.g. destination IP address, Port number, etc.) This trigger can generate a PDA request for the number of slots required for the active VoIP calls.

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Unlike most other triggers, the VoIP trigger does not have a fixed number of slots associated with it. Instead, calculation of the combined number of slots required for the current number of active VoIP calls (plus Data in case Dynamic DA is disabled) is required every time there is a change in the number of active VoIP calls.

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Ability to have an adaptive PDA slots allocation according to the number of VoIP sessions.

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Ability to close an H.323/SIP connection for which there is not enough bandwidth or for which the allowed amount of configured VoIP sessions has been exceeded.

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Return a busy tone to the caller by sending a RST message to the TCP SYN message (H323) or blocking the INVITE messages (SIP).

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Since VoIP runs simultaneously with other applications, the VoIP traffic will get priority over other traffic running in the inbound (Inbound QOS) and in the outbound (Allot).

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Ability to limit the total number of VoIP session (SIP and H323) per VSAT

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Once VoIP is working with Dynamic DA, the Dynamic DA mechanism allocates PDA slot up to the configured MIR (Maximum information rate) value.

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The allocation will be based on the data rate required by the VSAT not including the VoIP traffic (the VoIP Traffic is not part of the Dynamic DA calculations)

1.4.1 Voice over IP in SkyEdge Mesh Systems

In SkyEdge Mesh networks the following support for the Basic VoIP is available:

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SIP is fully supported.

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H323 Direct mode is supported.

1.4.2 Quality of Service

The use of QoS in a SkyEdge Basic VoIP network is of the utmost importance. Voice packets must form a continuous stream, with minimal fluctuation in delay, jitter, and close-to-zero packet loss. These are essential requirements for streaming. Any significant deviation from the nominal delay will cause a short break in the normal conversation, resulting in an unpleasant user experience. Lost packets will be

converted to a continuous, yet unclear speech, because the algorithm at the receiving side will try to guess what was missing.

In SkyEdge, in order to support QoS and prevent unwanted interferences, the hub automatically allocates enough bandwidth upon call establishment. Furthermore, priority is given to the internal router of the VSAT hub station to streaming packets over any other packets. More bandwidth is later allocated if a second call is

established over the same Remote Gateway, and the bandwidth is later released upon call termination.

On the Outbound link, Allot NetEnforcer is required to provide QoS services.

1.5 SkyEdge VoIP Calculations

The following adjustments must be made in a SkyEdge network to ensure proper operation of the Basic VoIP feature:

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A VoIP packet is sent every multi slot. The multi slot duration and VoIP packet interval must be synchronized.

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1.5.1 Calculation Algorithm

Variables:

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Time slot duration

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Multi slot

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S = bytes per voice frame

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T = voice frame per time

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MI – bytes per time slot. Codecs:

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For G.723 (6.3) codec:

− Every voice sample is 24 bytes, voice frame every 30 ms S=24, T = 30

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For G.729 codec:

− Every voice sample is 10 bytes, voice frame every 10 ms S=10, T = 10

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For G.711 codec:

− Every voice sample is 80 bytes, voice frame every 10 ms S=80, T = 10

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For G.728 codec:

− Every voice sample is 10 bytes, voice frame every 5 ms S=10, T = 5

The algorithm:

Multi slot duration = Time slot duration * Multi slot

For every codec:

For i=1 to N (i = Number of Voice frames per packet, N=1, 2, 3, 4, 5, 6…..) {X= Multi slot duration / T * I Packet size = 54 + S * I}

The following conditions should be met: 1. T * I <= 180 ms

2. X - will be as close as it can be to 1 ( 0.9 < X < 1.1) 3. Packet size < (MI – 8)

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If these conditions are not met, the codec is not valid. Example

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Time slot duration – 7.9967 ms

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Multi slot – 15 slots

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MI – bytes per time slot – 250 bytes

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Example for G.723 (6.3) codec:

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S = bytes per voice frame – 24 bytes

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T = voice frame per time – 30ms

Multi slot duration = 15 * 7.9967 = 119.95 ms.

I = 1, X= 3.998, Packet size = 78: condition 2 doesn’t meet. I = 2, X= 1.999, Packet size = 102: condition 2 doesn’t meet. I = 3, X= 1.333, Packet size = 126: condition 2 doesn’t meet. I = 4, X= 0.999, Packet size = 150: all conditions are meet: 1. 120<180

2. 0.9 < 0.999 < 1.1 3. 150<242

4. 150/242. Not equal to 1 but does not meet the condition.

With codec g.723, we will send a VoIP packet of 4 frames, length of 150 bytes every 120 ms.

1.5.2 G.729 Codec Example

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The G729 codec has a voice sample of 10 bytes every 10ms.

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The system operates in a multi slot duration of 120/180 ms (multi slot duration = time slot duration * multi slot).

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To work in 120 ms, there must be 12 samples (120/10 = 12) in every VoIP packet.

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Since every sample is 10 bytes, the packet payload will be 120 bytes (12*10) and the packet size will be 174 bytes (120+54).

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The slot MI must be big enough to carry 174 bytes: in a non-compressed RTP, the MI should be at least (since there are BB +LAPU headers) the packet size + 16 bytes – this means: MI>= 174 + 16 =190.

1.5.3 G.723 Codec Example

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The G.723 codec has a voice sample of 24 bytes every 30ms.

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The system operates in a multi slot duration of 120/180 ms. (multi slot duration = time slot duration * multi slot)

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To operate in 120 ms, there must be 4 samples in every VoIP packet 4 (120/30 = 4).

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Since every sample is 24 bytes the packet payload will be 96 bytes (24*4) and the packet size will be 150 bytes (96+54).

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The slot MI must be big enough to carry 150 bytes: in a non compressed RTP environment, the MI should be at least (since we are having BB +LAPU headers) the packet size + 16 bytes; this means MI>= 150 + 16 =166.

1.6 Basic VoIP Network Call Flow

This section describes the call data flow for a VSAT to PSTN/PBX call and VSAT-to-VSAT calls:

1. At the remote site, the user picks up the phone (off-hook), a local dial tone is generated locally by the Remote GW (Cisco ATA).

2. The user dials the number and presses the pound key [#] to indicate that the number is complete. If the pound key is not pressed, the number is sent to the Gatekeeper after a timeout of 3 seconds.

3. The Remote GW sends the number to the VoIP Gatekeeper. 4. The VoIP Gatekeeper performs the following:

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Transforms the phone number into an IP address – if no matching IP address is found for the requested number, a congestion tone is sent to the Remote GW and telephone device.

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Checks whether there are enough resources (bandwidth) to make the phone call - if there are not enough resources available, a congestion tone is sent to the Remote GW and telephone device.

5. If the above conditions are satisfied, the VoIP Gatekeeper sends the IP address of the destination (VoIP Gateway) to the Remote GW.

6. The Remote GW (Cisco ATA) opens a VoIP session with the VoIP Gateway at the hub.

7. If the destination is busy, the VoIP session is closed, a busy tone is sent to the Remote GW and the phone device.

8. If the destination is free to accept the call, the VoIP session continues. A diagram of the call establishment is shown in Figure 6.

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The user picks up the phone at the Remote site. The Phone is off hook

The dial tone is sounded. The dial tone is generated locally at the ATA

(Remote GW)

The user dials the number and presses the pound key or waits for a

timeout of 3 seconds

The ATA sends the number to the Gatekeeper

Gatekeeper finds a matching IP address to

the dialed number

No

Gatekeeper allocates resources to the call

The requested destination is free to

accept the call Yes

Yes

Yes

No

No ATA tries to open a VoIP session

with the requested destination Gatekeeper sends the IP address of

the requested destination (GW) to the ATA

ATA continues the VoIP session

The VoIP session is terminated A congestion tone is sent to the

telephone device

A busy tone is sent to the telephone device

The call is terminated The call is terminated

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1.7 Approved VoIP Components

The following components were tested and approved:

ƒ

Gateway

− Cisco 2600*/3600*/2800/3800 with FXO/FXS/E1 PRI / MFC-R2 − Addpac 2520 – E&M

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Gatekeeper

− Cisco 2600*/2800/3600*/3800 and AirSpan SSW

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CPE (Connection End Point/Remote Gateway) − Cisco ATA 186, AddpaC 200b.

NOTE

The following components have been declared end of life: Cisco 2600 and Cisco 3600.

1.8 Requirements

SkyEdge version 4.2 and higher.

1.9 System Limitations

This section lists system limitations that occur due to the use of Basic VoIP in a network:

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During a VoIP session, all applications run in DA – the data cannot be sent over DA and GA concurrently, thus during a VoIP session that requires the use of DA all other applications must run in DA.

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Fax over IP Transmissions requires extra bandwidth of 30 kbps – in

SkyEdge there are no separate identification mechanisms for VoIP and fax calls, and therefore the same bandwidth is allocated to voice and fax calls.

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Fax over IP - is not supported in SkyEdge Mesh networks.

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VoIP packets must be defined as absolute priority – VSATs send VoIP packets in the first opportunity in the available DA slot.

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VSAT can not stream EF packets with other packets – to avoid wasting system resources, configure the MI to be as close to the Compressed VoIP packet.

− For example, if the MI is 1000 bytes , and the RTP packet is 100 bytes , 900 bytes are not used

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VLAN tagging – this feature cannot be implemented in a SkyEdge network with Basic VoIP.

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cRTP (Compressed RTP) - is supported only for audio transmissions.

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cRTP - is a lossy compression; as a result some fields are synthesized.

ƒ

SIP – SIP is supported for standard port 5060 only.

ƒ

(In SkyEdge version 4.2) - Every VoIP session requires one PDA slot. Basic VoIP runs over DA, an extra PDA slot must be maintained for data and VSAT management traffic. This problem was solved in version 5.

For information about limitations in the SkyEdge Mesh networks, refer to Section 1.4.1, page 8.

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2. SkyEdge Basic VoIP Main Features

This section describes the following:

ƒ

H323 Implementation in SkyEdge

ƒ

SIP Implementation in SkyEdge

ƒ

Compressed RTP (cRTP) Implementation in SkyEdge

ƒ

Fax over IP

2.1 H323 Implementation in SkyEdge 2.1.1 Overview

H.323 is an OSI layer 5 protocol. The H323 network consists of the following protocols:

ƒ

H.225 Registration, Admission and Status - is a protocol between endpoints (terminals, gatekeeper and gateway) of the H.323 network. It is used to perform registration, admission control, bandwidth changes, status updates and disengage procedures between endpoints and gatekeeper. This is a signaling channel that is opened prior to the establishment of any other channels.

ƒ

H.225 Call Signaling - is used to establish a connection between two H.323 endpoints. This channel is opened between two H.323 terminals or between a terminal and the gatekeeper.

ƒ

H.245 Control Signaling - is used to exchange end-to-end control messages governing the operation of the H.323 endpoint. The control messages carry information related to:

− Capabilities exchange.

− Opening and closing of logical channels used to carry media streams. − Flow-control messages.

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ƒ

RCTP – Real-Time Transport Control Protocol - is a counterpart of RTP that provides control services. This protocol provides feedback on the quality of the data distribution; it carries a transport-level identifier for the RTP source (canonical name) that is used by receivers to synchronize audio and video.

ƒ

RTP – Real-Time Transport Protocol - provides end-to-end delivery services of real-time audio and video. It is based on the UDP for transport functionality. RTP provides payload identification, sequence numbering, time-stamping and delivery monitoring.

2.1.2 Operation Modes

Gatekeepers have two modes of operation - Direct mode and Routed mode. The routed mode is more commonly used.

In the Routed mode, the Gatekeeper performs address translation and provides endpoints with the transport address for the call signaling channel destination. In the Direct mode, the Gatekeeper provides the endpoints with the address of the destination endpoint and directs them to the call-signaling channel so that all messages can be exchanged directly between the two endpoints without Gatekeeper involvement.

In SkyEdge, H323 can be implemented in the following two modes:

ƒ

Direct Mode

ƒ

Routed Mode

2.1.3 Direct Mode

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Figure 7: Data Flow in H323 Direct Mode

ƒ

In the H323 Direct mode, the admission request and admission confirmation is sent from the End point (H323 Terminal) to the Gatekeeper.

ƒ

After the Gatekeeper’s confirmation is received, the two End Points connect directly without any mediator.

ƒ

The TCP connection is established between the two End Points.

ƒ

The RTP (VoIP media) is transmitted directly between the two End Points.

2.1.4 Routed Mode

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Figure 8: Data Flow in H323 Routed Mode

ƒ

In the H323 Routed mode, all signaling runs through the Gatekeeper.

ƒ

The TCP connection is established between the End Points and the Gatekeeper.

ƒ

The RTP runs directly between the End Points and does not go through Gatekeeper.

2.2 SIP Implementation in SkyEdge 2.2.1 Overview

SIP is an OSI 5-layer protocol. It is independent of lower layers protocols

SIP supports five facets of establishing and terminating multimedia communications:

ƒ

User location: determination of the end system to be used for communication.

ƒ

User capabilities: determination of the media and media parameters to be used.

ƒ

User availability: determination of the willingness of the called party to engage in communications.

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ƒ

Call setup: “ringing”, establishment of call parameters at both called and calling party.

ƒ

Call handling: including transfer and termination of calls.

Like any protocol SIP components communicate by exchanging SIP messages. SIP messages are built from Methods and responses. There are 7 different Methods:

ƒ

INVITE – Initiates a call by inviting user to participate in session.

ƒ

ACK - Confirms that the client has received a final response to an INVITE request.

ƒ

BYE - Indicates termination of the call.

ƒ

CANCEL - Cancels a pending request.

ƒ

REGISTER – Registers the user agent.

ƒ

OPTIONS – Used to query the capabilities of a server.

ƒ

INFO – Used to carry out-of-bound information, such as DTMF digits. Establishing communication using SIP usually occurs in six steps:

1. Registering, initiating and locating the user.

2. Determine the media to use – involves delivering a description of the session that the user is invited to.

3. Determine the willingness of the called party to communicate – the called party must send a response message to indicate willingness to communicate – accept or reject.

4. Call setup.

5. Call modification or handling – example, call transfer (optional). 6. Call termination.

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Figure 9: SIP Data Flow

Each time a user turns on the SIP user client (SIP IP Phone, PC, or other SIP device), the client registers with the proxy/registration server. Registration can also occur when the SIP user client needs to inform the proxy/registration server of its location. The registration information is periodically refreshed and each user client must re-register with the proxy/registration server. Typically the proxy/registration server will forward this information to be saved in the location/redirect server.

2.2.2 SIP Proxy

SIP proxies are elements that route SIP requests to user agent servers and SIP

responses to user agent clients. A request may traverse several proxies on its way to a UAS. Each will make routing decisions, modifying the request before forwarding it to the next element. Responses will route through the same set of proxies traversed by the request in the reverse order.

2.3 Compressed RTP (cRTP) Implementation in SkyEdge

NOTE

„ Compressed RTP is supported for both SIP and H323.

„ Compressed RTP is supported in SkyEdge star and mesh networks.

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Starting from SkyEdge version 4.2, the RTP packets can be compressed on the Inbound and Outbound. The compression is based on the RFC 2508 and RFC 3545. 32 cRTP sessions are supported per VSAT.

The VSAT and DPS open a tunnel per RTP session and only information that is necessary for RTP is transferred. The RTP header (54 Bytes: 14 Ethernet, 20 IP, 8 UDP, 12 RTP) is compressed to 6 bytes and total of 28 bytes (6 bytes cRTP, 8 bytes Backbone, and 14 bytes LAPU +L3).

The cRTP packet size over the Satellite:

Payload + Compressed header + BB header +LAPU header = Payload + 28 Bandwidth utilization without CRTP was increased from:

57% (96/166) to 72% (180/250)

Bandwidth utilization with CRTP was increased from: 77% (96/124) to = 87% (180/208).

The DPS/VSAT looks for two RTP packets (according to UDP ports and other RTP characteristics) with the same session number (SSRC). The RTP stream is monitored according to the SSRC. An audio RTP stream is recognized. The compression continues until a timeout occurs indicating the call had stopped.

Figure 10 shows the RTP and cRTP flow.

Figure 10: RTP and cRTP Flow

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ƒ

The VSAT/DPS recognizes a new RTP packet and adds to the Uncompressed RTP header the tunnel ID (5 bits) and sends it to DPS/VSAT.

ƒ

The VSAT/DPS receives the RTP packet with the tunnel ID and saves all relevant RTP headers fields.

ƒ

The VSAT/DPS sends ACK message to the originator of the packet DPS/VSAT with the tunnel ID.

ƒ

From the moment the VSAT/DPS receives the ACK message, it sends the RTP message without the unchanged RTP header fields but with the tunnel ID.

ƒ

If the CC or CSRC fields have been changed, the entire RTP packet is sent.

2.3.1 Configuration Notes

ƒ

Backbone fragment size must be the same as the cRTP packet.

ƒ

The first RTP packet is bigger than the rest.

ƒ

In case the Backbone fragment size is the same as the compressed RTP packet, the first RTP packets are fragmented. The jitter occurs at the beginning of the session.

2.4 Fax over IP

The SkyEdge Basic VoIP solution supports fax over IP transmissions. A fax machine is connected to the Residential Gateway (Cisco ATA). Any fax machines can be used in SkyEdge networks as long as they match fax transmission standards.

NOTE

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3. Cisco ATA Basic Configuration 3.1 Overview

Cisco ATA 186 is employed in the SkyEdge VoIP networks as a Remote Gateway. Cisco ATA 186 can be configured for use with H.323 or SIP using one of the following methods:

ƒ

Using a TFTP server - This method allows you to set up a unique Cisco ATA configuration file or a configuration file that is common to all Cisco ATAs.

ƒ

Manual configuration:

− Voice configuration menu - This is the method that must be used if the process of establishing IP connectivity for the Cisco ATA requires changing the default network configuration settings. You also can use the Voice configuration menu to review all IP connectivity settings. For more information, refer to Section 10.1.1, page 95.

− Web-based configuration - To use this method, the Cisco ATA must first obtain IP connectivity, either through the use of a DHCP server or by using the Voice configuration menu to statically configure IP addresses. For more information, refer to Section 0, page 97.

3.2 Establishing Connection between the VSAT/PC and the ATA

The connection between the VSAT or PC and the Cisco ATA 186 can be established in one of the two ways:

ƒ

Enabling the Cisco ATA DHCP via the Voice Configuration Menu OR

ƒ

Configuring the ATA Static IP Address

3.2.1 Enabling the Cisco ATA DHCP via the Voice Configuration Menu

NOTE

This section describes how to enable DHCP at the ATA.

For information on how to configure the ATA static IP address, subnet mask and default gateway, refer to Section 3.2.2, page 27.

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To enable DHCP at the Cisco ATA via the Voice configuration menu:

1. Check that the DHCP is enabled for the VSAT or PC to which the Cisco ATA unit will be connected. If Cisco ATA is connected to the VSAT, VSAT must be configured as a DHCP Server.

− For a detailed procedure on how to configure a VSAT as a DCHP Server, refer to Section 2.2 of the Advanced IP Features Configuration in SkyEdge Version 5 guide (DC-4458-10).

− For a detailed procedure on how to enable DHCP at the PC, refer to

Section 2.4 of the Advanced IP Features Configuration in SkyEdge Version 5 guide (DC-4458-10).

2. Connect an analog touch-tone phone (RJ-11 telephone line) to the port labeled Phone 1 on the back of the Cisco ATA.

Figure 11: Cisco ATA Connections

NOTE

To configure the Cisco ATA, the telephone device must be connected to the port labeled Phone 1.

3. Plug the AC power adaptor into an electrical outlet. Plug the power cord into the rear panel of the Cisco ATA 186 unit.

4. Lift the handset and press the Function button located on the top of the Cisco ATA.

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Figure 12: Cisco ATA Function Button

5. Reset the ATA to its factory settings as follows:

− On the telephone keypad, dial the following digits 322873738 (FACTRESET) and then the pound key [#].

− Press the star key [*] to save or the pound key [#] to exit.

6. To enable DHCP, press the following telephone keys: 20[#]1[#].

Result: The Voice Configuration Menu repeats the entered value and announces the following menu.

Table 1 lists the main Voice configuration commands as they are announced by the ATA after it is restored to the factory settings.

Table 1: Voice Configuration Menu Commands

Voice Menu Number Function

1[#] Change the entered value.

2[#] Review the entered value.

3[#] Save the entered value.

4[#] Return to the previous value.

# Exit to the Main configuration window.

7. To save the entered value (DHCP enabled), press the following keys: [3][#]. 8. Connect one end of a 10-BaseT Ethernet cable to the VSAT or PC. Connect the

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9. Wait for a few seconds and press 21[#].

Result: The ATA IP address that was received via DHCP is announced.

NOTE

If the Function button blinks slowly, the Cisco ATA cannot find the DHCP server. Check the Ethernet connections and make sure the DHCP server is available, e.g., the VSAT is online.

10. Write down the announced address.

3.2.2 Configuring the ATA Static IP Address

NOTE

This section describes how to configure the ATA static IP address, subnet mask and default gateway. These parameters should be configured only if the DHCP is disabled at the VSAT.

For information on how to enable DHCP at the ATA, refer to Section 3.2.1, page 24.

To configure the ATA Static IP address: 1. Connect an analog touch-tone phone

(RJ-11 telephone line) to the port labeled Phone 1 on the back of the Cisco ATA.

Figure 13: Cisco ATA Rear Panel

2. Plug the AC power adaptor into an electrical outlet. Plug the power cord into the rear panel of the Cisco ATA 186 unit.

3. Lift the handset and press the Function button located on the top of the Cisco ATA.

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Figure 14: Cisco ATA Function Button

4. Reset the ATA to its factory settings as follows:

− On the telephone keypad, dial the following digits 322873738 (FACTRESET) and then the pound key [#].

− Press the star key [*] to save or the pound key [#] to exit. 5. To disable DHCP, press the following telephone keys: 20[#]0[#].

Result: The Voice Configuration Menu repeats the entered value and announces the following menu.

Table 2 lists the main Voice configuration commands as they are announced by the ATA after it is restored to the factory settings.

Table 2: Voice Configuration Menu Commands

Voice Menu Number Function

1[#] Change the entered value.

2[#] Review the entered value.

3[#] Save the entered value.

4[#] Return to the previous value.

# Exit to the Main configuration window.

6. To save the entered value (DHCP disabled), press the following keys: [3][#]. 7. Using the telephone keypad, enter the voice menu code for the relevant

parameter or command and press the pound key [#].

8. To configure the ATA static IP address (for example, 111.222.33.44), press the following telephone keys: [1][ # ][1][1][1][*][2][2][2][*][3][3][*][4][4][#], where

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Result: The Voice Configuration Menu repeats the entered value and announces the commands as described in Table 2.

Table 3 lists additional voice configuration menu commands.

Table 3: Voice Configuration Menu Commands (cont’d)

Voice Menu Number Function

1[#] Configure the Cisco ATA static IP address.

2[#] Configure the Cisco ATA default gateway (static route).

10[#] Configure the Cisco ATA subnet mask.

20[#] DHCP—Set value to 0 to disable the use of a DHCP server; set value to 1 to enable DHCP.

21[#] Review the IP address of the Cisco ATA.

22[#] Review the default router for the Cisco ATA to use. 23[#] Review subnet mask of the Cisco ATA.

[*] Set a delimiter (dot) in the numeric values.

9. Press [3][#] to save the ATA static IP address.

10. To configure the ATA Subnet mask (for example, 255.255.255.0), press the following keys: [1][0][#][2][5][5][*][2][5][5][*][2][5][5][*][0][#].

11. Press [3][#] to save the Subnet mask.

12. To configure the ATA Default Gateway (for example, 111.222.33.10), press the following keys: [2][#][1][1][1][*][2][2][2][*][3][3][*][1][0][#].

13. Press [3][#] to save the ATA Default Gateway.

14. Press [21][#] to review the ATA IP address and write down the ATA static IP address.

15. Press [23][#] to review the Subnet mask and write down the Subnet mask address.

16. Press [22 ][#] to review the Default Gateway and write down the Default Gateway address.

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4. H323 Configuration on the SkyEdge NMS

This section describes the following:

ƒ

H323 Direct Mode Configuration

ƒ

H323 Routed Mode Configuration

4.1 H323 Direct Mode Configuration 4.1.1 Procedure Overview

To configure H323 Direct Mode parameters in the SkyEdge system, perform the following:

1. Configure/review DPS TCP parameters and timers as described in Section 4.1.2, page 31.

2. Configure TCP parameters and timers in the VSAT Data template as described in Section 4.1.3, page 33.

3. Depending on the DA mechanism used in your system, perform either of the following:

− Configure the general VoIP and Dynamic DA parameters in the networks that use the Dynamic DA mechanism. For more information, refer to Section 4.1.4.1, page 35.

OR

− Configure the general VoIP and Automatic DA parameters in the networks that use the Legacy DA mechanism. For more information, refer to Section 4.1.4.2, page 38.

4. Configure VSAT Data template VoIP Filters for the H323 Direct Mode as described in Section 4.1.5, page 43.

5. Configure port priorities in the VSAT Data template as described in Section 4.1.6, page 45.

6. Save and commit VSAT template configuration as described in Section 4.1.7 page 50.

7. Configure the licenses and port priorities of the VSATs that will be using H323 Direct Mode. For more information, refer to Section 4.1.8.1, page 51.

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4.1.2 Configuring DPS Parameters

This section describes how to configure DPS TCP timers that enable support of the keep-alive messages during a VoIP session. The keep-alive timers must be

configured at the hub site, at the DPS and at the VSAT site. For information on how to configure VSAT TCP timers, refer to Section 4.1.3, page 33.

To configure the DPS VoIP parameters:

1. In the Hub View window, double-click the DPS icon.

Result: The DPS Configuration window is displayed.

Figure 15: DPS Configuration

2. In the DPS Configuration window, select ConfigurationŒPortsŒDataŒTCP

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Figure 16: DPS TCP Parameters

3. Under General, configure the following DPS TCP profile parameters:

ƒ

Set the TCP Spoofing parameter to Enable.

ƒ

Set the TCP Connectivity parameter to END TO END

ƒ

Set the TCP Connection Keep Alive timer to Enable. 4. Under Timers, configure the following parameters:

ƒ

Set the User Timer to 10 seconds.

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NOTE

The DPS and VSAT Idle Timers must be smaller than the keep-alive timer of the Gatekeeper.

If the TCP Connection Keep Alive parameter at the DPS is enabled, keep-alive messages will be sent to the TCP peer on the user/application LAN port.

After the connection (towards the user/application connected network) is established, the User Timer is used to monitor connection inactivity. When triggered and no activity was detected, keep-alive will be sent. If no activity has been detected for 5 times the connection will be terminated. When triggered, the Idle (Inactivity) Timer will send a keep-alive segment at the specified frequency and will retrigger the retransmission and user timers.

5. Validate, save and commit the DPS configuration changes.

6. Right-click the DPS icon and select Commands

Œ

Reboot to reboot a non-redundant DPS or select Commands

Œ

Reboot Active & Standby.

Result: The DPS is rebooted.

7. Verify that the DPS completes its reboot sequence and goes online.

4.1.3 Configuring TCP Timers in the VSAT Data Template

NOTE

This section describes how to configure TCP timers in the VSAT Data template. These timers enable support of the keep-alive messages during a VoIP session. The keep-alive timers must be configured at the hub site, at the DPS and at the VSAT site. For information on how to configure DPS TCP timers, refer to Section 4.1.2, page 31.

To configure TCP timers in the VSAT Data template:

1. Open the VSAT Data template configuration window as described in Section 4.1.4.1, steps 1- 5.

2. In the left pane of the VSAT Data template configuration window, click PortsŒEthernetŒTCP Profile.

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Figure 17: VSAT TCP Profile Parameters

NOTE

The values of the VSAT TCP timers must match these of the DPS (Section 4.1.2, page 31).

3. Configure the following VSAT TCP Profile parameters:

ƒ

Set the TCP Spoofing parameter to Enabled.

ƒ

Set the TCP Connection keep alive parameter to Enable.

ƒ

Set the Local SynACK parameter to Disable.

ƒ

Set the User Timer parameter to 10 sec.

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NOTE

The DPS and VSAT Idle Timers must be smaller than the keep-alive timer of the Gatekeeper.

If the TCP Connection Keep Alive parameter at the VSAT is enabled, keep-alive messages will be sent to the TCP peer on the user/application LAN port.

After the connection (towards the user/application connected network) is established, the User Timer is used to monitor connection inactivity. When triggered and no activity was detected, keep-alive will be sent. If no activity has been detected for 5 times the connection will be terminated. When triggered, the Idle (Inactivity) Timer will send a keep-alive segment at the specified frequency and will retrigger the retransmission and user timers.

4.1.4 Configuring DA Parameters in the VSAT Data Template

Depending on the DA mechanism used in the network perform either of the following:

ƒ

Configuring VoIP and Dynamic DA Parameters OR

ƒ

Configuring VoIP and Legacy ADA Parameters in the VSAT Data Template

4.1.4.1 Configuring VoIP and Dynamic DA Parameters

To configure VoIP and Dynamic DA parameters in the VSAT Data template: 1. Click the Templates button.

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Figure 18: VSAT Templates

2. In the VSAT Software column, select the relevant software version, 3. In the VSAT Remote Processes column, select Data

4. In the Template Name column, select the relevant Data template. 5. Double-click the selected template.

Result: The selected Data template configuration window is displayed.

Figure 19: VSAT Data Template

6. In the left pane of the VSAT Data configuration window click Data

Œ

ADA.

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7. Set the ADA Operation Scheme parameter to Enhanced DA.

Result: The Enhanced (Dynamic) DA parameters are displayed.

Figure 20: Data Template Dynamic Parameters

8. In the Advanced Configuration section, set the

LAPU Tx mode while in DA parameter to Unnumbered. This parameter specifies the LAPU access mode during the VSAT ADA transmission of numbered or unnumbered traffic.

9. In the left pane of the VSAT Data Configuration window, click DataŒADAŒRT Applications.

Result: The RT Applications parameters are displayed.

Figure 21: VSAT Data Template – RT Applications

10. Under RT Applications, set the Max Num of Calls parameter to the maximum number of concurrent calls per VSAT (31). This numbers refers to the total number of concurrent calls per VSAT including SIP and H323.

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4.1.4.2 Configuring VoIP and Legacy ADA Parameters in the VSAT Data Template

To configure VoIP and Legacy ADA parameters in the VSAT Data template: 1. Click the Templates button.

Result: The VSAT Templates Configuration window is displayed.

Figure 22: VSAT Templates

2. In the VSAT Software column, select the relevant software version, 3. In the VSAT Remote Processes column, select Data

4. In the Template Name column, select the relevant Data template. 5. Double-click the selected template.

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Figure 23: VSAT Data Template

6. In the left pane of the VSAT Data configuration window click Data

Œ

ADA.

Result: The ADA parameters are displayed on the right.

7. Under General, set the ADA Operation Scheme parameter to Legacy ADA.

Result: The Legacy ADA parameters are displayed.

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ƒ

The ADA Optimum Slots and ADA Minimum Slots are not applicable to the VoIP ADA trigger. The maximum number of DA slots that can be assigned to a VSAT derives from the Super Slot Size parameter defined in the VSAT Access template. If the hub cannot allocate the minimum number of DA slots, the VSAT will not switch to DA. It will remain in the RA/GA mode and will transmit a new DA request after the DA Retry Timer expires (see below).

ƒ

Verify that the Fixed PDA Slots and Fixed PDA Rate parameters are set to 0.

ƒ

Set the DA Retry Time parameter to 30 seconds. This parameter specifies the time (in seconds) the VSAT will wait before retransmitting a request for DA allocation after a previous request was denied by the HSP.

ƒ

Verify that the ADA fairness timer is set to 0. This timer is used to set a time interval during which the HSP will allow the VSAT to transmit in DA using the original (maximum) number of allocated slots. When using VoIP, this parameter should be set to 0.

9. In the Advanced Configuration section, set the

LAPU Tx mode while in DA parameter to Unnumbered. This parameter specifies the LAPU access mode during the VSAT ADA transmission of numbered or unnumbered traffic.

10. In the left pane of the VSAT Data configuration window click Data

Œ

ADA

Œ

ROT threshold.

Result: The ROT threshold parameters are displayed.

11. Click the ROT tab.

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Figure 25: VSAT Data Template – ADA ROT Parameters

12. Under Operation Mode, configure the following parameters:

ƒ

Ignore EF Traffic –

− If the Ignore EF Traffic parameter is set to TRUE, the ROT trigger will ignore the EF class traffic (VoIP) when calculating the VSAT traffic. This will enable the use of the ROT trigger for sending more DA requests over active VoIP triggers.

− If the Ignore EF Traffic parameter is set to FALSE, the ROT Trigger will include the EF traffic (VoIP) when calculating the VSAT traffic.

The rest of the parameters on the ROT tab are not relevant to VoIP configuration. 13. Click DataŒADAŒRT Applications.

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Figure 26: Legacy ADA - RT Applications Parameters

14. Configure the RT Applications parameters using the following guidelines:

ƒ

In the Max Num of Calls field, enter the maximum number of concurrent calls per VSAT (31). This numbers refers to the total number of concurrent calls per VSAT including SIP and H323.

ƒ

In the Default Data Bandwidth field, specify the bandwidth (in Kbps) that will be guaranteed to the Data applications running simultaneously with the VoIP traffic.

− To support fax applications, set the Default Data Bandwidth to 30 Kbps. − The Default Data Bandwidth is used for calculating the number of PDA

slots needed for the Data applications running simultaneously with the VoIP traffic.

− When calculating the number of PDA slots for the VSAT, the required number of slots for this Data bit rate is added to the required number of slots for the active VoIP calls.

ƒ

Configure the Default Data Mandatory parameter as follows:

− If the Default Data Mandatory parameter is set to Yes, new VoIP calls will be allowed only if there are enough PDA slots to run the required Data bit rate simultaneously with the active VoIP calls. If there are not enough PDA slots for the Data applications, new VoIP calls will be discarded.

− If this parameter is set to No, the VSAT will try to acquire enough PDA slots to run the required Data bit rate. The new VoIP calls will be allowed even if the Data requirements are not met.

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NOTE

The Default Data Bandwidth is used for calculating the number of PDA slots needed for the Data applications running simultaneously with the VoIP traffic.

When calculating the number of PDA slots for the VSAT, the required number of slots for this Data bit rate is added to the required number of slots for the active VoIP calls.

4.1.5 Configuring VoIP Filters for H323 Direct Mode

To configure VoIP filters for the H323 Direct mode:

1. Open the VSAT Data template configuration window as described in Section 4.1.4.1, steps 1- 5.

2. In the left pane of the VSAT Data template, click RT Applications

Œ

VoIP Filter Instances.

Result: The VoIP Filter Instances table is displayed.

Figure 27: VoIP Filter Instances

3. If there are no preconfigured filter instances, right-click the VoIP TCP Filter Instances table on the right and select Add VoIP Filters.

Result: The new VoIP filter is added to the table.

4. Right-click the filter and select Display row in new window.

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Figure 28: New VoIP Filter

5. Review the configuration parameters and modify if necessary to support TCP H.323 Direct mode:

ƒ

In the Filter Enabled/Disabled field, select Enabled.

ƒ

In the VoIP Packets Interval field, specify the time in milliseconds between two successive VoIP packets in each call associated with the selected TCP filter. This parameter is used The VoIP Packets Interval parameter is used for calculation of the PDA (partial DA allocation) and must correspond to the relevant

parameters in the ATA configuration.

ƒ

In the Bytes Per VoIP Packet field, specify the number of bytes per VoIP packet in each call associated with the selected TCP filter. The Bytes Per VoIP Packet parameter is used for calculation of the PDA (partial DA allocation) and must correspond to the relevant parameters in the ATA configuration.

ƒ

Set the IP Address and IP mask parameters to 0.

ƒ

Set Port Number to 1720.

ƒ

Set the Signaling protocol type to to TCP H.323. NOTE

When configuring Codec G723, 6 samples per packet, set the VoIP Packets Interval parameter to 180 mSeconds and the Bytes Per VoIP Packet parameter to 198 bytes.

For Codec G723, 4 samples per packet, set VoIP Packets Interval to 120 mSeconds and Bytes Per VoIP Packet to 150 bytes.

For Codec G729, 12 samples per packet, set VoIP Packets Interval to

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6. Configure the second VoIP TCP filter as described in steps 3 through 5.

7. Set Port Number of the second VoIP TCP filter instance to 1721. All other parameters must be configured as described in this section.

8. Save the changes.

4.1.6 Configuring Port Priorities in the VSAT Data Template

To configure port priorities in the VSAT Data template:

1. Open the VSAT Data template configuration window as described in Section 4.1.4.1, steps 1- 5.

2. In the left pane of the VSAT Data configuration window, click

PortsŒEthernetŒIP PrioritizationŒIP Classifier Template Table Instances

Result: The VSAT IP Classifier Template Table Instances are displayed.

Figure 29: IP Classifier Template Table Instances

3. Right-click the IP Classifier Template table and select Add Multiple IP Classifier Template Table.

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4. Enter the number of IP Classifier entries (8) to be created and click OK. Total of nine entries must be configured in the IP Classifier table: one entry is predefined and eight more must be added.

Result: New IP Classifier entries are added.

Figure 30: IP Classifier Entries

5. Right-click the first entry and select Display Row in new window.

Result: The selected row is redisplayed in the window format.

Figure 31: Configuring a New Entry in the IP Classifier

6. Enter the Name for the entry. 7. Set the Active parameter to Yes.

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8. Set the Protocol parameter to UDP.

9. Set Source Port Start and Source Port Stop parameters to 1719. 10. Set the Operation parameter to CS5.

11. Save the changes.

12. Configure the rest of the IP Classifier instances (total 9) as shown in Table 4 and in Configuration Notes below:

Table 4: H323 IP Classifier Instances

Instance Name Active Protocol TOS Source Port Start Source Port Stop Destin. Port Start Destin. Port Stop Operation 1 Yes UDP 0 1719 1719 CS5 2 Yes UDP 0 1719 1719 CS5 3 Yes TCP 0 1720 1721 CS5 4 Yes TCP 0 1720 1721 CS5 5 Yes UDP 0 1739 1739 CS5 6 Yes UDP 0 1739 1739 CS5 7 Yes UDP 184 EF 8 Yes UDP 0 16384 16386 EF 9 Yes UDP 0 16384 16386 EF Configuration Notes:

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Lines 1 through 6 indicate the H323 Signaling process.

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Lines 7-9 indicate the RTP configuration.

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Each port must be configured twice: first as a source port and then as a destination port range.

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Port 1719 must be configured as two entries in the table: as a source port range and then as a destination port range. The Operation level of port 1719 must be set to CS5.

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Port 1739 must be configured as two entries in the table: as a source port range and then as a destination port range. The Operation of port 1739 must be set to

Figure

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References

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