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 this document are subject to change without prior notice.

Transcript of Basic VoIP

Page 1: Basic VoIP

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 this document are subject to change without prior notice.

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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

3.2.1 Enabling the Cisco ATA DHCP via the Voice Configuration Menu ...............24 3.2.2 Configuring the ATA Static IP Address ........................................................27

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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

10.1.1 Voice Configuration Menu........................................................................... 95 10.1.2 Web-Based Configuration........................................................................... 97

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10.2 Remote Gateway Commands ................................................................................99 10.2.1 Refreshing the Cisco ATA Configuration .....................................................99 10.2.2 Resetting the Cisco ATA ...........................................................................100 10.2.3 Resetting the Cisco ATA to Factory Default Values ...................................100

11. Appendix B - SkyEdge Basic VoIP System Debugging ...................................101 11.1 General System Debugging .................................................................................101

11.1.1 Using a Sniffer Application ........................................................................101 11.2 VoIP Gateway & VoIP Gatekeeper Commands....................................................103

11.2.1 Logging on to the VoIP Gateway/Gatekeeper Command Line Interface.....103 11.2.2 VoIP Gateway Maintenance Commands ...................................................104 11.2.3 VoIP 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

13. Appendix D - Glossary ......................................................................................116

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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 Figure 45: Cisco ATA Configured successfully ............................................................. 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: sh voice dsp command .............................................................................105 Figure 70: sh voice call command..............................................................................106 Figure 71: sh h323 gateway cause-codes command ................................................107 Figure 72: sh gatekeeper calls command...................................................................109 Figure 73: sh gatekeeper endpoints 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 Figure 78: ATA Upgrade Command.............................................................................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 Table 11: VoIP Gatekeeper Commands....................................................................... 108

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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.

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:

General Overview

SkyEdge VoIP Network

SkyEdge VoIP Network Components

SkyEdge VoIP Solution Description

SkyEdge VoIP Calculations

Basic VoIP Network Call Flow

Approved VoIP Components

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:

Residential Gateway (another VSAT)

Hub VoIP phone

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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.

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.

Figure 1: SkyEdge VoIP Network Architecture

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A SkyEdge VoIP network consists of the following major components:

A master earth station and control facility or hub.

A Network Management System to control the system.

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:

At the Hub:

− VoIP Gateway

− VoIP Gatekeeper

− Provisioning server (TFTP)

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:

Cisco 3725 router to support 60 simultaneous calls

Cisco 3745 router to support 120 simultaneous calls

Cisco 5400 router to support 240 simultaneous calls

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

E1 - An E1 line can carry 30 digitized voice channels

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.

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.

Figure 3: One and Two Port E1 Multi-Flex Trunk Voice/Wan Interface Cards

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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:

Translates phone numbers into IP addresses and vice versa.

Assigns resources to calls

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:

Pentium IV

0.5 G RAM memory

Standard VGA card

Minimum of 20 G Byte hard disk space

One Ethernet Port of 10/100

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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)

A B C

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Cisco ATA 186 has the following characteristics:

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

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

(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:

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

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.

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.

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

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.

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

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).

The priority of the voice media (RTP) will be EF – absolute priority while the VoIP signaling will have CS5 priority.

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

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

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:

SIP is fully supported.

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:

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

Use this algorithm to calculate the codec for VoIP requirements.

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1.5.1 Calculation Algorithm

Variables:

Time slot duration

Multi slot

S = bytes per voice frame

T = voice frame per time

MI – bytes per time slot.

Codecs:

For G.723 (6.3) codec:

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

For G.729 codec:

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

For G.711 codec:

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

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)

4. Packet size/(MI – 8) = as closer as it can be to 1 (Not obligatory)

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

Example

Time slot duration – 7.9967 ms

Multi slot – 15 slots

MI – bytes per time slot – 250 bytes

Example for G.723 (6.3) codec:

S = bytes per voice frame – 24 bytes

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

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

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

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).

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

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

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

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

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).

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:

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 theRemote site.

The Phone is off hook

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

(Remote GW)

The user dials the number andpresses the pound key or waits for a

timeout of 3 seconds

The ATA sends the number to theGatekeeper

Gatekeeper finds amatching IP address to

the dialed numberNo

Gatekeeper allocatesresources to the call

The requesteddestination is free to

accept the call

Yes

Yes

Yes

No

No

ATA tries to open a VoIP sessionwith the requested destination

Gatekeeper sends the IP address ofthe requested destination (GW)

to the ATA

ATA continues the VoIP session

The VoIP session is terminated

A congestion tone is sent to thetelephone device

A busy tone is sent to the telephonedevice

The call is terminated

The call is terminated

Figure 6: SkyEdge VoIP Call Establishment

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1.7 Approved VoIP Components

The following components were tested and approved:

Gateway

− Cisco 2600*/3600*/2800/3800 with FXO/FXS/E1 PRI / MFC-R2

− Addpac 2520 – E&M

Gatekeeper

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

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:

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.

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.

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

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

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

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

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

SIP – SIP is supported for standard port 5060 only.

(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:

H323 Implementation in SkyEdge

SIP Implementation in SkyEdge

Compressed RTP (cRTP) Implementation in SkyEdge

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:

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.

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.

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.

− General commands and indications.

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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.

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:

Direct Mode

Routed Mode

2.1.3 Direct Mode

Figure 7 shows data flow in the network operating in the H323 Direct mode.

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Figure 7: Data Flow in H323 Direct Mode

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

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

The TCP connection is established between the two End Points.

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

2.1.4 Routed Mode

Figure 8 shows data flow in the network operating in the H323 Routed mode.

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Figure 8: Data Flow in H323 Routed Mode

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

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

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:

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

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

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

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

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:

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

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

BYE - Indicates termination of the call.

CANCEL - Cancels a pending request.

REGISTER – Registers the user agent.

OPTIONS – Used to query the capabilities of a server.

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.

Compressed RTP is supported only for audio transmissions.

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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

According to Figure 10, the RTP and cRTP data flow is as follows:

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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.

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

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

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.

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

2.3.1 Configuration Notes

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

The first RTP packet is bigger than the rest.

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

The fax machines should be set to the highest speed.

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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:

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.

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:

Enabling the Cisco ATA DHCP via the Voice Configuration Menu

OR

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 other end of the Ethernet cable to the ATA RJ-45 input port.

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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 the star key [*] indicates a delimiter.

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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.

17. Connect one end of a 10-BaseT Ethernet cable to the VSAT or PC. Connect the other end of the Ethernet cable to the ATA RJ-45 input port.

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4. H323 Configuration on the SkyEdge NMS

This section describes the following:

H323 Direct Mode Configuration

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 Configuration Ports Data TCP

Result: The DPS TCP parameters are displayed.

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Figure 16: DPS TCP Parameters

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

Set the TCP Spoofing parameter to Enable.

Set the TCP Connectivity parameter to END TO END

Set the TCP Connection Keep Alive timer to Enable.

4. Under Timers, configure the following parameters:

Set the User Timer to 10 seconds.

Set the Idle Timer to 30 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 Reboot to reboot a non-redundant DPS or select Commands 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 Ports Ethernet TCP Profile.

Result: VSAT TCP Profile parameters are displayed.

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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:

Set the TCP Spoofing parameter to Enabled.

Set the TCP Connection keep alive parameter to Enable.

Set the Local SynACK parameter to Disable.

Set the User Timer parameter to 10 sec.

Set the Idle Timer parameter to 30 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:

Configuring VoIP and Dynamic DA Parameters

OR

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.

Result: The VSAT Templates Configuration window is displayed.

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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 ADA.

Result: The ADA parameters are displayed on the right.

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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 Data ADA RT 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.

11. Validate and save VSAT Data template configuration.

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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.

Result: The selected Data template configuration window is displayed.

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Figure 23: VSAT Data Template

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

Figure 24: VSAT Data Template – Legacy ADA

8. In the ADA section, configure the parameters as described below:

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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).

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

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.

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 ADA ROT threshold.

Result: The ROT threshold parameters are displayed.

11. Click the ROT tab.

Result: The Rate of Traffic parameters are displayed.

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Figure 25: VSAT Data Template – ADA ROT Parameters

12. Under Operation Mode, configure the following parameters:

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 Data ADA RT Applications.

Result: The RT Application parameters are displayed on the right.

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Figure 26: Legacy ADA - RT Applications Parameters

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

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.

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.

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 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.

Result: The new VoIP filter is redisplayed. The new filter is added with default values which usually should not be modified.

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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:

In the Filter Enabled/Disabled field, select Enabled.

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.

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.

Set the IP Address and IP mask parameters to 0.

Set Port Number to 1720.

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 120 mSeconds and Bytes Per VoIP Packet to 174 bytes.

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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 Ports Ethernet IP Prioritization IP 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.

Result: The Number of Elements window is displayed.

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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:

Lines 1 through 6 indicate the H323 Signaling process.

Lines 7-9 indicate the RTP configuration.

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

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.

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 CS5.

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Ports 1720 and 1721 can be configured as source and destination port ranges. The Operation levels of ports 1720 and 1721 must be set to CS5.

The VSAT forwards a packet sent with TOS of 184 at the highest operation level – EF.

The ATA must be configured to send RTP packets with TOS byte set to EF.

Ports 16384 and 16386 can be configured as source and destination port ranges.

The Operation level of ports 16384 and 16386 must be set to EF – the highest priority. The Operation levels of all other ports should be set to CS5.

The QoS configuration presented in Table 4, above is for Fast Connect mode. In case of working in Slow Connect mode, add two entries for port 1740.

Configure the following ports for signaling: 1719, 1720, 1721, and1739.

Configure the following ports for Voice traffic: 16384 and 16386.

UDP ports: 1719, 1739, 16384, and 16386

TCP ports: 1720, 1721

Figure 32 shows an example of the configured IP Classifier table.

Figure 32: IP Classifier Table Configured

13. In the Data template, select Data Ports Ethernet IP Prioritization CS1-CS7 Traffic Class Descriptions Instances

Result: The CS1-CS7 Traffic Class Descriptions table is displayed.

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Figure 33: CS1-CS7 Class Descriptors

14. Configure the CS5 (5th table row) traffic class parameters as follows:

In the 5th row, set the Queuing Algorithm parameter to Priority.

Calculate the Maximal Bit Rate for the CS5 priority using the following formula:

Maximum Number of Calls * 8 = Maximal Bit Rate

The Maximum Number of Calls is configured in the Data ADA VoIP section (see Section 4.1.4.1, step 10 and Section 4.1.4.2, step 14 ). Currently the default maximum number of VoIP calls per VSAT is 2. So according to this formula, the Maximal Bit Rate for CS5 is 16:

2 (Maximum Number of Calls) *8 = 16

The Weight and Flow Priority parameters are not in use.

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NOTE

In networks using Voice over IP solutions, VoIP is assigned a CS5 traffic class.

If in such networks there are applications assigned higher traffic classes, such as CS6 and CS7 and there will be not enough bandwidth, the VoIP calls will not be established.

Generally, CS6 and CS7 traffic classes should be assigned to mission-critical applications only.

4.1.7 Saving and Committing VSAT Template Configuration

To save and commit VSAT Template configuration:

1. Click Save.

2. Click Commit.

Result: The Commit and Reset VSAT confirmation window is displayed.

NOTE

For the changes to take effect, all VSATs must be reset.

If more changes must be made to VSAT configuration, it is recommended to make all the changes and then to reset a VSAT.

3. Perform either of the following:

Select Yes, to reset VSATs associated with the selected template immediately.

Select No, to commit configuration changes without resetting the VSATs.

Result: The VSAT Data template status changes to Committed.

4. If you selected to reset the VSATs with the Commit command (see step 3 above), verify that all VSATs associated with the modified template are reset successfully and go online.

4.1.8 Configuring VSAT Unique Parameters

This section describes how to configure parameters located in the VSAT Configuration window.

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This procedure must be performed for each VSAT that will be using VoIP. Users that have good knowledge of the SkyEdge NMS can use the Edit VSATs utility to modify all VSATs at the same time.

NOTE

To activate the Edit VSATs utility:

In the VSAT Manager window, right-click the VSATs to be modified and select Edit VSATs utility.

Do not use the Edit VSATs utility on more than 500 VSATs at the same time.

Use the Edit VSATs utility to modify the relevant parameters and reset the VSATs.

For more information about the Edit VSATs utility, refer to Part III of the SkyEdge NMS Operator’s Manual for Version 15.0 (DC-4230-10).

4.1.8.1 Configuring VSAT Licenses and Port Priorities

To configure VSAT licenses and port priorities:

1. Click the VSATs View button.

Result: The VSATs View window is displayed.

2. Double-click the relevant VSAT icon.

Result: The VSAT Configuration window is displayed.

3. Click the Data tab.

Result: The VSAT Data parameters are displayed.

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Figure 34: VSAT Data Parameters

4. In the Data Template field, select the relevant Data template or verify that the correct Data template is selected. This is the Data template that was configured with VoIP parameters as described in Sections 4.1.2 through 4.1.7.

5. In the Data template, select Data License Management

Result: VSAT License Management parameters are displayed.

6. Verify that the VSAT Enhanced IP license is activated for the selected VSAT.

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Figure 35: VSAT Licenses

7. In the Data template, select Data Ports Ethernet IP Prioritization

Result: VSAT IP Prioritization parameters are displayed.

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Figure 36: VSAT IP Prioritization

8. Set the IP Classifier Tables Order parameter to Template then Specific.

− The IP Classifier Tables Order parameter defines the order at which VSAT Template and VSAT specific prioritization tables are checked. The following options are available: first template and then VSAT specific parameters, first VSAT and then template, only VSAT, or only template.

− If this parameter is set to Template, then Specific, VSAT template prioritization table will be checked first and only afterwards VSAT specific table will be checked.

9. Set the Default Traffic Class parameter to any value ranging from CS1 to CS4. The Default Traffic Class parameter defines the priority that will be applied to the VSAT traffic that does not fit any of the rules defined in the VSAT template and specific parameters.

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Figure 37: VSAT Prioritization Parameters

NOTE

For the VSAT configuration changes to take effect VSAT configuration must be saved, committed and the VSAT must be reset.

When several changes must be made in the VSAT configuration, it is recommended first to make all the changes and then to commit and reset the VSAT. This will reduce the number of times the VSAT must be reset for the configuration changes to take effect.

10. Validate and save VSAT configuration changes.

11. Click the Commit button.

Result: The VSAT Reset confirmation window is displayed.

Click Yes, to commit the VSAT configuration and reset the VSAT

Click No to commit the VSAT configuration without resetting the VSAT.

Result: The VSAT status is changed to Committed.

12. If you selected to reset the VSAT immediately, verify that the VSAT is reset successfully and goes online.

13. You can reset the VSAT later on by right-clicking the VSAT icon and selecting Commands Access Reset.

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4.2 H323 Routed Mode Configuration

NOTE

The H323 Routed mode is not supported by Cisco routers containing Gatekeeper IOS.

The H323 VoIP over Mesh is not supported in the H323 Routed mode.

4.2.1 Procedure Overview

To configure H323 Routed 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 Routed Mode as described in Section 4.2.2, page 57.

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 Routed Mode. For more information, refer to Section 4.1.8.1, page 51.

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4.2.2 Configuring VoIP Filters for the H323 Routed Mode

NOTE

Perform this procedure after defining DPS and VSAT TCP parameters and configuring general VoIP and Dynamic/Legacy DA parameters as described in Sections 4.1.2 - 4.1.4.

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 VoIP Filter Instances.

Result: The VoIP Filter Instances table is displayed.

Figure 38: 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.

Result: The new VoIP filter is redisplayed. The new filter is added with default values which usually should not be modified.

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Figure 39: New VoIP Filter for the H323 Routed Mode

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

In the Filter Enabled/Disabled field, select Enabled.

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.

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.

In the IP Address and IP mask parameters, define the Gatekeeper’s IP address and subnet mask.

Set Port Number to 1720.

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

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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 120 mSeconds and Bytes Per VoIP Packet to 174 bytes.

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.

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

10. Save and commit the template configuration changes as described in Section 4.1.7, page 50.

11. Configure licenses and priorities of the VSATs that will be working the H323 Routed mode as described in Section 4.1.8.1, page 51.

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5. H323 Configuration on the Cisco ATA

NOTE

For information on how to configure the Cisco ATA for SIP, refer to Section 7, page 75.

This section describes the following procedures:

Accessing the Web Interface and Checking the ATA Version

Configuring the Cisco ATA for H323 Using the ATA Web Interface

5.1 Accessing the Web Interface and Checking the ATA Version

To access the Web interface and check the ATA version:

1. Verify that the VSAT/PC and the Cisco ATA are visible to each other by issuing a ping from the VSAT/PC to the Cisco ATA.

2. Open the Web browser and verify that no proxy settings are configured.

3. Type the URL of the ATA configuration page: http://<ipaddress>/dev; where <ipaddress> is the ATA IP address.

Use the ATA IP address that was obtained through DHCP as described in step 10 of Section 3.2, page 24.

OR

Use the Static ATA IP address as described in step 14 of Section 3.2.2, page 27.

Result: The ATA 186 (H323) Configuration screen is displayed.

Figure 40: H323 ATA Configuration Screen (Upper Part)

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Figure 41: ATA Configuration Screen for H323

4. Scroll the bottom of the ATA Configuration screen. The ATA version details are displayed on the left.

Figure 42: ATA Version and Configuration Information

5. Check the ATA Build number. The ATA Build number should be 041104T. If the ATA Build number does not match this number, upgrade the ATA version as described in Appendix C- Upgrading the Cisco ATA Software Version, page 111.

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

NOTE

The use of H323 supports VoIP sessions between a VSAT/ATA and a PSTN and VoIP sessions between two VSATs/ATAs – Point-to-Point.

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NOTE

For information on how to configure the Cisco ATA for SIP, refer to Section 5, page 60.

To configure the Cisco ATA for H323:

1. Access the ATA 186 (H323) Web Configuration interface as described in steps 1 through 3 of Section 5.1, page 60.

Result: The Cisco ATA 186 Configuration Screen is displayed.

Figure 43: ATA H323 Configuration

2. Configure the relevant parameters as shown in Table 5, below. The description in this table is based on the Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (H323).

NOTE

This guide describes Cisco ATA parameters that are relevant for Gilat H323 configuration. For full description of the parameters and information about other ATA configuration parameters refer to the Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (H323).

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Table 5: ATA Configuration for H323

Parameter Name Description Values Default/ Recommended Value

Parameters for Configuration Method and Encryption These parameters are not relevant to Gilat Configuration

UIPassword This parameter is not relevant to Gilat configuration.

N/A N/A

UseTftp This parameter is not relevant to Gilat configuration.

N/A N/A

Tftp URL This parameter is not relevant to Gilat configuration.

N/A N/A

CfgInterval This parameter is not relevant to Gilat configuration.

N/A N/A

EncryptKey This parameter is not relevant to Gilat configuration.

N/A N/A

EncryptKeyEx This parameter is not relevant to Gilat configuration.

N/A N/A

Network Parameters

Dhcp Enables/Disables the use of DHCP server.

Values: 1 = Use DHCP to obtain IP, route, netmask, DNS, NTP, TFTP, etc.

0 = Use hard-coded parameter values for IP, route, netmask, DNS, NTP, TFTP, etc.

StaticRoute Statically assigned default route. Used if DHCP is set to 0.

IP address When used, this parameter should be set to the VSAT/PC IP address.

StaticIP Statically assigned IP address. Used if DHCP is set to 0.

IP address

StaticNetMask Statically assigned subnetmask. Used if DHCP is set to 0.

Subnetmask

Account Information Parameters

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Parameter Name Description Values Default/ Recommended Value

UID0 Specify the telephone numbers for the Cisco ATA port: Phone 1.

If this parameter is set to 0, the port is disabled and no dial tone is sound.

UID0 is designated as Phone1 and is always defined.

Use UID0 for registration and authentication. If a Proxy server requires separate registration, use UID0 to specify the registration ID only. Use Login0 to specify the authentication ID.

Alphanumeric string of up to 31 characters.

PWD0 This parameter is not used in Gilat configuration.

N/A N/A

UID1 Indicates Cisco ATA port: Phone2.

UID1 is defined only when G711 codec is used.

In all other cases, this parameter should be set to 0.

Alphanumeric string of up to 31 characters.

PWD1 This parameter is not used in Gilat configuration.

N/A N/A

Gateway Specifies the H323 Gateway IP address. This is applicable to the networks that do not contain an H323 Gatekeeper.

In this case, all calls originating from the Cisco ATA will be directed to Gateway specified in this parameter.

IP address

UseLoginID Indicates which of the following will be used as a login ID:

UID0 and UID1

Or

LoginID0 and Login ID1

1 – use the specified LoginID

0 – do not use the specified LoginID.

If UseLoginID is set to 0, UID0 and UID1 are used for authentication.

0

LoginID0 This parameter is not used in Gilat configuration.

N/A N/A

LoginID1 This parameter is not used in Gilat configuration.

N/A N/A

Gatekeeper Parameters

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Parameter Name Description Values Default/ Recommended Value

GkOrProxy The IP address of the primary H323 Gatekeeper.

If set to 0, Gatekeeper-routed calls are disabled.

IP address

GkId Gatekeeper (H323) name. Alphanumeric string of up to 31 characters

GkTimeToLive Indicates the “time to live” value that is used when the Cisco ATA registers with the H.323 gatekeeper.

The registration is valid until the configured time expires.

Seconds

AltGk This parameter is not relevant to Gilat configuration.

N/A N/A

AltGkTimeOut This parameter is not relevant to Gilat configuration.

N/A N/A

Operating Parameters

MediaPort Specifies the RTP base port that the Cisco ATA uses to receive and transmit RTP media streams.

This port is defined in the VSAT IP Classifier on the SkyEdge NMS. For more information, refer to Section 4.1.6, page 45.

Port number 16384

AudioMode Enables/disables silence suppression.

To enable silence suppression, set to 0x00150015.

To disable silence suppression, set to 0x00100010.

0x00100010 – by default, the silence suppression is disabled.

LBRCodec (Low-bit-rate codec)

Indicates the low bit rate codec that can be used during codec negotiation.

Values:

0 – select G.723 as low-bit-rate codec.

3 – select G.729 as low bit rate codec.

6 – select G.726-32kbps as the low bit rate code.

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Parameter Name Description Values Default/ Recommended Value

RxCodec Indicates the Receiving audio codec preference.

Values:

0 – select G.723 as low-bit-rate codec; can be selected only if LBRCodec is 0.

3 – select G.729a as low bit rate codec; can be selected only if LBRCodec is 3.

1 – G.711A-law

2 – G.711-law

6 - G.726 – 32 kbps

TxCodec Indicates the Transmitting audio codec preference.

Values:

0 – select G.723 as low-bit-rate codec; can be selected only if LBRCodec is 0.

1 – G.711A

2 – G.711u

3 – select G.729a as low bit rate codec can be selected only if LBRCodec is 3.

NumTxFrames Indicates the number of frames per packet.

The frame size for each G.729 data packet is 10 ms.

The frame size for each G.723 data packet is 30 ms.

The frame size for G.711 is fixed at 20 ms per packet and is not configurable.

To obtain 180 ms of G.723 audio, set to 6.

To obtain 120 ms of G.723 audio, set to 4.

To obtain 180 ms of G.729 audio, set to 18.

To obtain 120 ms of G.729 audio, set to 12.

ConnectMode Enables/disables fast start procedure. (Connection mode of the protocol used).

To enable the fast start procedure, set 0x00060401.

To disable the fast start procedure, set 0x00060400.

By default, this parameter should be (enabled) - 0x00060401

3. Scroll to the bottom of the page and click apply.

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Figure 44: Applying Configuration Changes

Result: The Cisco ATA 186 Configured Successfully message is displayed.

Figure 45: Cisco ATA Configured successfully

4. For the new parameters to take effect, click the Click here to reload option.

5. Wait for the page to refresh. The “Click here to reload” message should disappear.

6. Verify the relevant parameters are updated.

7. You can now try to make a test phone call to the hub.

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6. SIP Proxy Configuration on the SkyEdge NMS

6.1 Procedure Overview

To configure SIP Proxy 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 SIP Proxy VoIP Filters in the VSAT Data template as described in Section 6.2, page 68.

5. Configure SIP port priorities in the VSAT Data template as described in Section 6.3, page 71.

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 SIP. For more information, refer to Section 4.1.8.1, page 51.

6.2 Configuring SIP Proxy VoIP Filters in the VSAT Data Template

NOTE

Perform this procedure after defining DPS and VSAT TCP parameters and configuring general VoIP and Dynamic/Legacy DA parameters as described in Sections 4.1.2 - 4.1.4, pages 31 - 35.

To configure SIP Proxy VoIP filters:

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1. Click the Templates button.

Result: The VSAT Templates Configuration window is displayed.

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.

6. In the left pane of the VSAT Data template, click Data ADA RT Applications VoIP Filter Instances.

Result: The VoIP Filter Instances table is displayed.

Figure 46: VoIP Filter Instances

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

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Figure 47: Adding a New VoIP Filter

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

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

Result: The new VoIP filter is redisplayed. The new filter is added with default values.

Figure 48: Adding a New SIP VoIP Filter

9. Review the SIP Proxy configuration parameters and modify if necessary as shown below.

10. In the Filter Enabled/Disabled field, select Enabled.

11. 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. 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.

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12. 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.

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 120 mSeconds and Bytes Per VoIP Packet to 174 bytes.

13. In the IP Address and IP mask fields, specify the Gatekeeper’s IP Address and subnet mask.

14. Set Port Number to 5060.

15. Set the Signaling protocol type to to UDP SIP.

16. Save the changes.

17. Configure port priorities in the VSAT Data template as described in the next section, see Section 6.3, page 71.

6.3 Configuring SIP 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 6.2, page 68, steps 1- 5.

2. In the left pane of the VSAT Data configuration window, click Ports Ethernet IP Prioritization IP Classifier Template Table Instances

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

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Figure 49: IP Classifier Template Table Instances

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

Result: The Number of Elements window is displayed.

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

Result: New IP Classifier entries are added.

Figure 50: 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.

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Figure 51: Configuring a New Entry in the IP Classifier

6. Enter the Name for the entry.

7. Set the Active parameter to Yes.

8. Set the Protocol parameter to UDP.

9. Set Source Port Start and Source Port Stop parameters to 5060.

10. Set the Operation parameter to EF.

11. Save the changes.

12. Configure the rest of the IP Classifier instances (total 5) as shown in Table 6 and in Configuration Notes below:

Table 6: SIP 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 5060 5060 EF

2 Yes UDP 0 5060 5060 EF

3 Yes UDP 184 EF

4 Yes UDP 0 16384 16386 EF

5 Yes UDP 0 16384 16386 EF

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Configuration Notes:

Lines 1-2 indicate the SIP Signaling configuration.

Lines 3-5 indicate the RTP configuration.

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

The VSAT forwards a packet sent with TOS of 184 at the highest operation level – EF.

The ATA must be configured to send RTP packets with TOS byte set to EF.

Ports 16384 and 16386 can be configured as source and destination port ranges.

The Operation level of ports 16384 and 16386 must be set to EF – the highest priority.

Configure the following port for SIP signaling: 5060 and these ports for Voice traffic: 16384 and 16386.

UDP ports: 5060,16384, and 16386

Figure 32 shows an example of the configured IP Classifier table.

Figure 52: IP Classifier Table Configured

13. Save the changes.

14. Save and commit the template configuration changes as described in Section 4.1.7, page 50.

15. Configure licenses and priorities of the VSATs that will be working the H323 Routed mode as described in Section 4.1.8.1, page 51.

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7. SIP Configuration on the Cisco ATA

This section contains the following procedures:

Accessing the Web Interface and Checking the ATA Version

Configuring the Cisco ATA for SIP via ATA Web Interface

7.1 Accessing the Web Interface and Checking the ATA Version

To access the Web interface and check the ATA version:

1. Verify that the PC and the Cisco ATA are visible to each other by issuing a ping from the PC to the Cisco ATA.

2. Open the Web browser and verify that no proxy settings are configured.

3. Type the URL of the ATA configuration page: http://<ipaddress>/dev. Where <ipaddress> is the ATA IP address.

− Use the ATA IP address that was obtained through DHCP as described in Section 3.2.1, page 24.

− Or

− Use the Static ATA IP address as described in Section 3.2.2, page 27.

Result: The ATA 186 (SIP) Configuration screen is displayed.

Figure 53: SIP ATA Configuration Screen (Upper Part)

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Figure 54: Cisco ATA SIP Configuration

4. Scroll the bottom of the ATA Configuration screen.

Result: The ATA version details are displayed on the left.

Figure 55: ATA Version and Configuration Information

5. Check the ATA Build number. The ATA Build number should be 040211A. If the ATA Build number does not match this number, upgrade the ATA version as described in Appendix C- Upgrading the Cisco ATA Software Version, page 111.

7.2 Configuring the Cisco ATA for SIP via ATA Web Interface

NOTE

The use of SIP supports VoIP sessions between a VSAT/ATA and a PSTN.

NOTE

For information on how to configure the Cisco ATA for H323, refer to Section 5, page 60.

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To configure Cisco ATA for SIP via ATA Web interface:

1. Access the ATA 186 (SIP) Web Configuration interface as described in Section 7.1, steps 1 through 3, page 75.

Result: Cisco ATA 186 Configuration Screen is displayed.

Figure 56: ATA Configuration for SIP

2. Configure the relevant parameters as shown in Table 7, below. The description in this table is based on the Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide for SIP (version 3.0).

NOTE

This guide describes Cisco ATA parameters that are relevant for Gilat SIP configuration. For full description of the parameters and information about other ATA configuration parameters refer to Chapter 5 of the Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide for SIP (version 3.0).

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Table 7: ATA Configuration for SIP

Parameter Name Description Values Default/ Recommended Value

Parameters for Configuration Method and Encryption These parameters are not relevant to Gilat Configuration

UIPassword This parameter is not relevant to Gilat configuration

N/A N/A

UseTftp This parameter is not relevant to Gilat configuration

N/A N/A

Tftp URL This parameter is not relevant to Gilat configuration

N/A N/A

CfgInterval This parameter is not relevant to Gilat configuration

N/A N/A

EncryptKey This parameter is not relevant to Gilat configuration

N/A N/A

EncryptKeyEx This parameter is not relevant to Gilat configuration

N/A N/A

Network Parameters

Dhcp Enables/Disables the use of DHCP server.

Values: 1 = Use DHCP to obtain IP, route, netmask, DNS, NTP, TFTP, etc.

0 = Use hard-coded parameter values for IP, route, netmask, DNS, NTP, TFTP, etc.

StaticRoute Statically assigned default route. Used if DHCP is set to 0.

IP Address When used, this parameter should be set to the VSAT/PC IP address

StaticIP Statically assigned IP address. Used if DHCP is set to 0.

IP Address

StaticNetMask Statically assigned subnet mask. Used if DHCP is set to 0.

Subnet mask

Account Information Parameters

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Parameter Name Description Values Default/ Recommended Value

UID0 Specify the telephone numbers for the Cisco ATA port: Phone 1.

If this parameter is set to 0, the port is disabled and no dial tone is sound.

UID0 is designated as Phone1 and is always defined.

Use UID0 for registration and authentication. If a Proxy server requires separate registration, use UID0 to specify the registration ID only. Use Login0 to specify the authentication ID.

Alphanumeric string of up to 31 characters.

PWD0 This parameter is not relevant to Gilat configuration

N/A N/A

UID1 Indicates Cisco ATA port: Phone2.

UID1 is defined only when G711 codec is used.

In all other cases, this parameter should be set to 0.

Alphanumeric string of up to 31 characters.

PWD1 This parameter is not relevant to Gilat configuration

N/A N/A

SIP Configuration Parameters

GkOrProxy SIP Proxy server IP address.

For a SIP proxy server, this can be an IP address with or without a port parameter such as: 123.123.110.45, 123.123.110.45.5060, or 123.123.110.45:5061, or a URL such as sip.cisco.com, or sip.ata.cisco.com:5061.

For an IP address, a '.' or ':' can be used to delimit a port parameter.

For a URL, a ':' must be used to indicate a port.

AltGk This parameter is not relevant to Gilat configuration

N/A N/A

GkTimeToLive This parameter is not relevant to Gilat configuration

N/A N/A

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Parameter Name Description Values Default/ Recommended Value

UseLoginID Indicates which of the following will be used as a login ID:

UID0 and UID1

Or

LoginID0 and Login ID1

1 – use the specified LoginID

0 – do not use the specified LoginID.

If UseLoginID is set to 0, UID0 and UID1 are used for authentication.

1

LoginID0 This parameter is not relevant to Gilat configuration

N/A

LoginID1 This parameter is not relevant to Gilat configuration

N/A

SIPPort Indicates the port through which the Cisco ATA listens for incoming SIP requests and sends outgoing SIP requests.

Integer from 1 to 65535

5060

SIPRegInterval Indicates a number of seconds between the Cisco ATA registration renewal with the SIP Proxy server.

To avoid registration expiration, the Cisco ATA renews the registration some time earlier than the specified timeout.

Seconds from 1 to 86400

SIPRegOn Enables/Disables the SIP registration.

When this flag is enabled, the Cisco ATA registers with the SIP Proxy server configured in the GkOrProxy parameter. The Cisco ATA registers within the interval specified by the SIPRegInterval parameter.

Audio Configuration Parameters

MediaPort Specifies the base port that the Cisco ATA uses to receive RTP media streams.

Integer from 1 to 65535

16384

AudioMode Enables/disables silence suppression.

To enable silence suppression, set to 0x00150015.

To disable silence suppression, set to 0x00100010.

0x00150015 – by default the silence suppression is enabled.

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Parameter Name Description Values Default/ Recommended Value

LBRCodec (Low-bit-rate codec)

Indicates the low bit rate codec that can be used during codec negotiation.

Values:

0 – select G.723 as low-bit-rate codec.

3 – select G.729a as low bit rate codec.

6 – select G.726-32kbps as the low bit rate code.

RxCodec Indicates the Receiving audio codec preference.

Values:

0 – select G.723 as low-bit-rate codec; can be selected only if LBRCodec is 0.

3 – select G.729a as low bit rate codec; can be selected only if LBRCodec is 3.

1 – G.711A-law

2 – G.711-law

6 - G.726 – 32 kbps

TxCodec Indicates the Transmitting audio codec preference.

Values:

0 – select G.723 as low-bit-rate codec; can be selected only if LBRCodec is 0.

1 – G.711A

2 – G.711u

3 – select G.729a as low bit rate codec can be selected only if LBRCodec is 3.

6 - G.726 – 32 kbps

NumTxFrames Indicates the number of frames per RTP packet that the Cisco ATA transmits.

The frame size for each G.729 data packet is 10 ms.

The frame size for each G.723 data packet is 30 ms.

The frame size for G.711 is fixed at 20 ms per packet and is not configurable.

To obtain 180 ms of G.723 audio, set to 6.

To obtain 120 ms of G.723 audio, set to 4.

To obtain 180 ms of G.729 audio, set to 18.

To obtain 120 ms of G.729 audio, set to 12.

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Parameter Name Description Values Default/ Recommended Value

ConnectMode Enables/disables fast start procedure. (Connection mode of the protocol used).

To enable the fast start procedure, set 0x00060401.

To disable the fast start procedure, set 0x00060400.

By default, this parameter should be (enabled) - 0x00060401

3. Scroll to the bottom of the page and click apply.

Figure 57: Applying Cisco ATA Configuration Changes

Result: The Cisco ATA 186 Configured Successfully message appears.

Figure 58: Cisco ATA Configured Successfully

4. For the new parameters to take effect, click the Click here to reload option

5. Wait for the page to refresh. The “Click here to reload” message should disappear.

6. Verify the relevant parameters are updated.

7. You can now try to make a test phone call to the hub.

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8. Compressed RTP Configuration on the SkyEdge NMS

8.1 Procedure Overview

To enable the cRTP support in a SkyEdge network, perform the following procedures:

Enabling cRTP at the DPS

Enabling cRTP Support at the VSATs

8.2 Enabling cRTP at the DPS

This section describes how to enable the Compressed RTP support on the Outbound link.

To enable cRTP at the DPS:

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

Result: The DPS Configuration window is displayed.

2. In the DPS Configuration window, select Configuration Ports Data IP

Result: The DPS IP parameters are displayed.

Figure 59: DPS Data IP Parameters

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3. Scroll to the right to the CRTP section.

Figure 60: DPS cRTP Parameters

4. In the CRTP section, set the CRTP enable parameter to Enabled. This parameter enables/disables the RTP header compression in a SkyEdge network.

5. Set the CRTP TOS filter to EF. This filter indicates that the cRTP will be applied only to the packets matching the specified DiffServ ToS. The cRTP will be applied only to the packets with the EF ToS.

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

7. Right-click the DPS icon and select Commands Reboot to reboot a non-redundant DPS or select Commands Reboot Active & Standby.

Result: The DPS is rebooted.

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

8.3 Enabling cRTP Support at the VSATs

This section describes how to enable the Compressed RTP support on the Inbound link. Compressed RTP is activated at the VSAT Data template.

To enable and configure cRTP at the VSATs:

1. Click the Templates button.

Result: The VSAT Templates Configuration window is displayed.

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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.

6. In the left pane of the VSAT Data Template configuration window, click Ports Ethernet IP Profile

Result: The VSAT Data Template cRTP parameters are displayed.

Figure 61: VSAT Data Template cRTP Parameters

7. Set the CRTP enable parameter to Enabled. This parameter enables/disables the RTP header compression in a SkyEdge network.

8. Set the CRTP TOS filter to EF. This filter indicates that the cRTP will be applied only to the packets matching the specified DiffServ ToS. By default, the cRTP will be applied only to the packets with the EF ToS. The value of the VSAT CRTP TOS filter parameter must match the DPS CRTP TOS filter.

9. Save the configuration changes in the Data template.

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10. Click Commit.

Result: The Commit and Reset VSAT confirmation window is displayed.

NOTE

For the changes to take effect, all VSATs must be reset.

If more changes must be made to VSAT configuration, it is recommended to make all the changes and then to reset a VSAT.

11. Perform either of the following:

Select Yes, to reset VSATs associated with the selected template immediately.

Select No, to commit configuration changes without resetting the VSATs.

Result: The VSAT Data template status changes to Committed.

12. If you selected to reset the VSATs with the Commit command (see step 11 above), verify that all VSATs associated with the modified template are reset successfully and go online.

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9. Fax Support Configuration on the SkyEdge NMS

9.1 Procedure Overview

This section describes how to configure the SkyEdge system to support the T38 protocol for fax applications.

This section contains the following procedures:

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

Configure the Fax application and Dynamic DA parameters in the networks that use the Dynamic DA mechanism. For more information, refer to Section .

OR

Configure the Fax application and Automatic DA parameters in the networks that use the Legacy DA mechanism. For more information, refer to Section .

9.2 Configuring Fax Support in a System with Dynamic DA

NOTE

This procedure describes how to modify the existing template in order to configure the fax feature in the SkyEdge network.

Depending on the network and the number of VSATs using the template, it might be recommended to create a new Data template that will be used for Voice over IP traffic and then apply it to the relevant VSATs.

For information on how to create a new Data template, refer to Part III of the SkyEdge NMS Operator’s Manual for Version 15.0 (DC-4230-10).

To configure fax support in a system with Dynamic DA:

1. Click the Templates button.

Result: The VSAT Templates Configuration window is displayed.

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.

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5. Double-click the selected template.

Result: The selected Data template configuration window is displayed.

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

Result: The ADA parameters are displayed on the right.

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

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

Figure 62: 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 Data ADA Dynamic DA.

Result: The Dynamic DA parameters are displayed.

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Figure 63: Dynamic DA Parameters

10. Under Extra Bandwidth Reservation, set the But not Less than a Fixed Extra of parameter to 30 Kbps. The value of 30 Kbps is configured in the system that supports one concurrent fax per single VSAT.

Figure 64: Dynamic DA – Extra Bandwidth Reservation Parameters

NOTE

The But not Less than a Fixed Etra of parameter specifies the minimum bandwidth that will be required by VSATs associated with this template. This parameter refers to the minimum extra bandwidth and this value is fixed.

This parameter specifies the minimum bandwidth that will be guaranteed to the Data applications running simultaneously with the VoIP traffic Configuring the But not Less than a Fixed Etra of parameter to the value of 30 Kbps and higher affects the system performance.

11. Save the changes in the VSAT Data template.

12. Click Commit.

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Result: The Commit and Reset VSAT confirmation window is displayed.

NOTE

For the changes to take effect, all VSATs must be reset.

If more changes must be made to VSAT configuration, it is recommended to make all the changes and then to reset a VSAT.

13. Perform either of the following:

Select Yes, to reset VSATs associated with the selected template immediately.

Select No, to commit configuration changes without resetting the VSATs.

Result: The VSAT Data template status changes to Committed.

14. If you selected to reset the VSATs with the Commit command (see step 3 above), verify that all VSATs associated with the modified template are reset successfully and go online.

9.3 Configuring Fax Support in a System with Legacy DA

To configure a SkyEdge system to support fax in a Legacy DA environment:

1. Click the Templates button.

Result: The VSAT Templates Configuration window is displayed.

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.

6. In the left pane of the VSAT Data configuration window click Data 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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Figure 65: VSAT Data Template – Legacy ADA

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. Click Data ADA RT Applications.

Result: The RT Application parameters are displayed on the right.

Figure 66: Legacy ADA - RT Applications Parameters

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

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11. 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.

12. 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.

13. 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.

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.

14. Click Save.

15. Click Commit.

Result: The Commit and Reset VSAT confirmation window is displayed.

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NOTE

For the changes to take effect, all VSATs must be reset.

If more changes must be made to VSAT configuration, it is recommended to make all the changes and then to reset a VSAT.

16. Perform either of the following:

Select Yes, to reset VSATs associated with the selected template immediately.

Select No, to commit configuration changes without resetting the VSATs.

Result: The VSAT Data template status changes to Committed.

17. If you selected to reset the VSATs with the Commit command (see step 3 above), verify that all VSATs associated with the modified template are reset successfully and go online.

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10. Appendix A - Remote Gateway Configuration and Basic Commands

This section includes the following:

Remote Gateway Configuration

Remote Gateway Commands

10.1 Remote Gateway Configuration

NOTE

This section is based on the Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (H323)

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:

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. The Cisco ATA can automatically download its latest configuration file from the TFTP server when the Cisco ATA powers up, is refreshed or reset, or when the specified TFTP query interval expires.

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.

− 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.

CAUTION

Parameters configured using the TFTP server overwrite any parameters configured by either Voice configuration menu or Web-based configuration.

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10.1.1 Voice Configuration Menu

The Voice configuration menu is mainly used to configure the IP address of the TFTP server.

NOTE

For a detailed procedure on how to configure the Cisco ATA IP address, subnet mask, and the default gateway, refer to Section 3.2.2, page 27.

To manually configure the Cisco ATA by using the Voice configuration menu and the telephone keypad, perform the following:

1. Connect a touch-tone phone to the port labeled Phone 1 on the back of the Cisco ATA.

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

Result: The opening Voice configuration menu prompt is initiated.

3. Using the telephone keypad, enter the voice menu code for the parameter that you want to configure or the command that you want to execute, and then press the pound key [#]. For a list of voice menu codes, see Table 8.

NOTE

If you are using the Voice configuration menu to statically configure the Cisco ATA IP address, you must disable DHCP by setting its value to 0.

Table 8 lists the Voice configuration menu options needed to configure basic IP connectivity for the Cisco ATA, after which the Cisco ATA Web configuration can be used to configure additional parameters.

Table 8: Voice Menu Basic IP Connectivity Parameters

Voice Menu Number Features 1 StaticIP - IP address of the Cisco ATA.

2 StaticRoute - Default gateway for the Cisco ATA to use.

10 StaticNetMask - Subnet mask of the Cisco ATA.

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.

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Voice Menu Number Features 23 Review subnet mask of the Cisco ATA.

4. Follow the voice prompts and enter the appropriate values, then press the pound key [#].

NOTE

Use the star key (*) to indicate a delimiter (dot). For example, to enter an IP address of 192.168.3.1, enter 192*168*3*1 on your telephone keypad.

5. The Voice configuration menu repeats the entered value, then prompts you to press one of the following keys:

[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 menu

6. After entering the desired value, press the pound key [#]. If you do not press the pound key [#], the system will reach time out automatically after 10 seconds.

7. Gilat strongly recommends setting a password. Use the Voice menu code 7387277 (SETPASS) to configure a password through the Voice configuration menu, after which you are prompted for the password whenever you attempt to change a parameter value.

8. After completing the configuration through the Voice configuration menu, press the pound key [#] to exit.

9. Hang up the telephone. The Cisco ATA configuration refreshes. The Function button fast-blinks when the refresh completes.

NOTE

Do not use the Voice configuration menu to attempt to change any values that have been configured by means of the TFTP configuration file method. Whenever the Cisco ATA refreshes, it downloads its ata<macaddress> configuration file or atadefault.cfg default configuration file from the TFTP server, and the values in either of these files will overwrite the values of any corresponding parameters configured with the Voice configuration menu.

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10.1.1.1 Configuring Alphanumeric Values

Some Voice configuration menu options require entering alphanumeric characters. After each alphanumeric entry the pound key [#] must be pressed. The Voice prompt announces when an alphanumeric entry must configured, otherwise numeric entries (0 to 9) should be configured.

Table 9 lists the keys on a telephone keypad and their respective alphanumeric characters.

Using Table 9 as a reference, enter the appropriate number key on the telephone keypad as many times as needed to select the number, letter, or symbol required. For example, to enter 58sQ, the following combination must be entered:

[5][#][8][#][7][7][7][7][7][#][7][7][7][7][7][7][#][#].

Table 9: Voice Menu Alphanumeric Characters

Key Alphanumeric Characters 1 1 ./_\@*space return +-!,?|~^#=$”‘’%<>[]:;{}()&

2 2 a b c A B C

3 3 d e f D E F

4 4 g h i G H I

5 5 j k l J K L

6 6 m n o M N O

7 7 p q r s P Q R S

8 8 t u v T U V

9 9 w x y z W X Y Z

0 0

* . (delimiter)

10.1.2 Web-Based Configuration

The Web-based configuration can be used to configure any of the ATA parameters, including the TFTP server IP address or URL and to issue commands. In the Web configuration, the parameters are grouped and color-coded according to their functions.

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

The Cisco ATA 186 Web configuration can be accessed from any graphics-capable browser, such as Microsoft Internet Explorer or Netscape. To open the Web configuration of Cisco ATA 186, type the following URL address: http://<ipaddress>/dev.

The Web configuration can be used to complete the Cisco ATA 186 configuration. The TFTP configuration file values always overwrite the corresponding values of the Web configuration.

NOTE

Whenever the Cisco ATA refreshes, it downloads its ata<macaddress> configuration file or atadefault.cfg default configuration file from the TFTP server, and the values in either of these files overwrite the values of any corresponding parameters configured using the Web-based configuration.

To set parameters using the Web configuration:

1. Verify that the PC and the Cisco ATA are visible to each other by issuing a ping from the PC to the Cisco ATA.

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2. Open the Web browser.

3. Enter the URL for the configuration page. The default URL for the Web configuration is: http://IP Address/dev. For example, the configuration page for a Cisco ATA with the IP address 192.116.89.201 is: http:// 192.116.89.201/dev.

Result: The Web Configuration page is displayed (Figure 67).

4. Configure the relevant parameters.

5. Scroll down to the bottom of the page and click apply to save the changes.

Result: The following message appears on the ATA Configuration page:

Cisco ATA 186 Configured Successfully

Restarting ATA Now

[Click here to reload]

6. For the new parameters to take effect, click to reload.

Result: The ATA contacts the TFTP server and starts downloading the relevant configuration file. During the download process the Function button led blinks. As the download process completes, the led stops blinking and the dial tone should be sounded.

7. At the end of the download process, the Web configuration is redisplayed.

8. Verify that the relevant parameters are updated.

9. Close the Web browser.

10.2 Remote Gateway Commands

This section describes the following procedures:

Refreshing the Cisco ATA Configuration

Resetting the Cisco ATA

Resetting the Cisco ATA to Factory Default Values

10.2.1 Refreshing the Cisco ATA Configuration

The refresh procedure updates the Cisco ATA configuration file.

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To refresh and display the most recent Cisco ATA configuration:

1. Open the Web browser.

2. Type the following command: http://<ipaddress>/refresh, where ipaddress is the IP address of the relevant Cisco ATA.

Result: The Web Configuration Main Menu of the specified ATA opens.

10.2.2 Resetting the Cisco ATA

NOTE

The reset procedure updates the Cisco ATA configuration file and powers down and powers up the Cisco ATA.

To reset the Cisco ATA:

1. Open the Web browser.

2. Type the following command http://<ipaddress>/reset, where ipaddress is the IP address of the Cisco ATA to be reset.

Result: The specified Cisco ATA is reset.

10.2.3 Resetting the Cisco ATA to Factory Default Values

CAUTION

This procedure describes how to reset the Cisco ATA to its factory default values. Do not perform this procedure unless instructed by the Gilat Technical Support.

To perform a factory reset:

1. Press the Function button on the Cisco ATA.

2. Press the digits 322873738 (FACTRESET) then press the pound key [#] on the telephone keypad.

3. Press 3 and then the pound key [#] on the telephone keypad to confirm that you want to reset the Cisco ATA, and then hang up the phone.

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11. Appendix B - SkyEdge Basic VoIP System Debugging

This section includes the following procedures:

General System Debugging

VoIP Gateway & VoIP Gatekeeper Commands

11.1 General System Debugging

This section lists general steps that must be performed if a problem is detected in a SkyEdge Basic VoIP network.

Perform these steps:

1. Check whether the system is configured to work with 120 ms or 180 ms (Packet Interval).

After establishing the size of the multislot, check whether the ATA and VSAT are configured appropriately. For detailed information, refer to Sections 4 - 9 in this manual.

Check configuration of the following ATA parameters: LBRCodec, RxCodec, TxCodec, and NumTxFrames. For more information, refer to Table 5, page 63 and Table 7, page 78.

2. Verify configuration of the rest of the ATA parameters that are relevant to Gilat configuration. For more information, refer to Table 5, page 63 and Table 7, page 78.

3. Use a Sniffer application to check the VoIP traffic as described in Section 11.1.1, page 101.

11.1.1 Using a Sniffer Application

NOTE

Please note that when using a Low Fly simulator, the delay is much lower than when working with the real satellite link.

This section provides general information about how to use Sniffer to examine VoIP traffic.

1. Connect the Sniffer application to port monitoring in between two ATAs/Gateways.

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2. In the Sniffer application, define the filters relevant for each protocol: SIP or H323.

3. Define the following Sniffer fields to examine the VoIP traffic (see Figure 68):

Source and Destination – information from these fields allows establishing whether the call is indeed initiated at the correct ATA, whether admission request has been sent and received by the relevant parties. If problems are detected at this level, check whether ATAs are configured properly, and are communicating with the adjacent VSATs.

Info – information from this field enables you to view call establishment in details, including call registration, admission, status, signaling, and call control signaling for each the protocols: H323 or SIP. Verify whether the call is established according to the relevant protocol configuration. If problems are detected at this level, review configuration of VSATs and ATAs.

Protocols – use this field to verify which protocol is used.

Timing – use this information to determine whether there is any abnormal delay between the request and answer messages. The call is disconnected if a delay exceeds predefined timers. If problems are detected at this level, check CPEs configuration.

Delta – verify that the delta value is symmetric. Delta indicates interval between the packets. When using Codec G.729, the Codec sample interval is 10 ms for each packet. Therefore, when the system works in 120ms, delta of 120ms for each transmission should be indicated by the Sniffer. If the delta value is asymmetric, it means that the system MI size does not fit the volume of the system transmissions. If the delta value is symmetric, it indicates the jitter is none. For more information, refer to Section 1.5, page8.

Length – this parameter is used for calculation of the packet size. The packet size is according to the Codec type used in the system. When the system works in Codec G.729, every sample is 10 K, therefore when working in 120ms, there must be 12 samples per packet, which is 120K (12x10). TCP header of 54K is added to this packet, resulting in 174K. In this case, the length should be 174K consistently. If the length is not according to the Codec type, it means that the ATA codec timing interval is not configured correctly.

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Figure 68: Sample Sniffer Capture

4. Create a capture of the Sniffer log and send it to Gilat for further analysis.

11.2 VoIP Gateway & VoIP Gatekeeper Commands

This chapter contains the following sections:

Logging on to the VoIP Gateway/Gatekeeper Command Line Interface

VoIP Gateway Maintenance Commands

VoIP Gatekeeper Maintenance Commands

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

To log on to the Gateway/Gatekeeper CLI:

1. On a Remote PC or ProComm terminal, open a Telnet connection to the Gateway/Gatekeeper.

Result: The Password prompt is displayed on the CLI.

2. Type $giLat$.

Result: The Gateway> or Gatekeeper> prompt appears.

3. At the > prompt, type en.

Result: The password prompt is displayed.

4. Type $giLat$.

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Result: The Gateway# or Gatekeeper# prompt appears.

5. The connection to the Gatekeeper is established and the statistic commands can be issued.

11.2.2 VoIP Gateway Maintenance Commands

This section lists the main VoIP Gateway statistic commands and their description. For examples of some of the commands, refer to Sections 11.2.2.1 through 11.2.2.3, pages 105 - 107.

Table 10: VoIP Gateway Show Commands

Command Description sh voice port Display E1 configuration

sh voice call Display E1 call status

sh voice dsp Display E1 DSP status

sh call active voice Display all active calls

sh call history voice Display calls stored in the voice history table

sh call resource voice stats dsp Display the voice DSP resources utilization statistics

sh dial-peer voice Display configuration of the Voice Encapsulated Peers (POTS or VoIP)

sh gateway Display the VoIP gateway status

sh h323 gateway cause-codes Display the disconnect cause codes and statistics

sh h323 gateway h225 Display the H.225 Gateway statistics

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11.2.2.1 SHOW VOICE DSP

Purpose

To display the Digital Signal Processes (DSP) status and configuration.

Syntax

sh voice dsp

Example

Figure 69: sh voice dsp command

Explanation

The sh voice dsp command provides the following information:

List of the Digital Signal Processes configured in the system. In this example, six DSPs are configured in the system. This means that six simultaneous calls are supported.

DSP version

Current state of the processes. In the example, two DSPs are in busy state, indicating that two calls are being made and four are in idle state. This indicates that two calls are currently in process.

To learn more about the currently active calls, use the sh voice call command (see Section 11.2.2.2, page 106).

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11.2.2.2 SHOW VOICE CALL

Purpose

To display the system call setup.

Syntax

sh voice call

Example

Figure 70: sh voice call command

Explanation

The sh voice call command provides the following information about currently active calls:

Call ID. In the example, there are two currently active calls. The ID of the first is 0x8026 and the ID of the second is 0x003A.

Call state. In the example, the first call is in the connect state and the second is in the initialization call state.

Call originator and call destination. The first call was originated by 1111 (callg) and the destination was 9055258081 (calld). The second call was originated by 3227 and its destination was 1200.

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11.2.2.3 SHOW H323 GATEWAY CAUSE-CODES

Purpose

To display the call disconnect cause codes and statistics.

Syntax

sh h323 gateway cause-codes

Example

Figure 71: sh h323 gateway cause-codes command

Explanation

The sh h323 gateway cause-codes command provides information about disconnect causes of calls made between the VoIP Gateway and another peer (VoIP Gatekeeper or Remote Gateway):

Disconnect cause code.

No route to destination – the requested destination was not found. Due to this cause, 13 calls were disconnected by the Gateway and one by its peer.

Normal call clearing – the calls were terminated by the end-users. Due to this cause, 17 calls were disconnected by the Gateway and 17 by its peer.

User busy – the requested destination was not available to accept the call. Due to this cause, 0 calls were disconnected by the Gateway and 2 by its peer.

Call rejected – not enough resources were allocated to the call. Due to this cause, 5 calls were disconnected by the Gateway and 0 calls by its peer.

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11.2.3 VoIP Gatekeeper Maintenance Commands

This section lists the main VoIP Gatekeeper statistic commands and their description. For information on how to issue a command, refer to Sections 11.2.3.1 and 11.2.3.2, pages 109 - 110.

Table 11: VoIP Gatekeeper Commands

Command Description sh gatekeeper calls Display current Gatekeeper call status

sh gatekeeper endpoints Display all endpoints registered with this Gatekeeper

sh gatekeeper gw-type-prefix Display Gateway Technology Prefix table

sh gatekeeper performance stats Display Gatekeeper performance data

sh gatekeeper status Display the Gatekeeper status

sh gatekeeper zone status Display all zone statuses known to this gatekeeper

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11.2.3.1 SH GATEKEEPER CALLS

Purpose

To display the current status of the Gatekeeper calls.

Syntax

sh gatekeeper calls

Example

Figure 72: sh gatekeeper calls command

Explanation

The sh gatekeeper calls command provides the following information:

Call ID, duration in seconds (age) and bandwidth (Kbps).

Call source: source phone number, IP address and Port number

Call destination, destination phone number, IP address and Port number

In the example, the first call (call ID 28-65535) was originated by the Remote Gateway (ATA10-1) with an IP address of 10.101.6.247 and a source port of 1720. The call destination was the VoIP Gateway with an IP address of 10.101.100.241 and a destination port of 1720.

At the time the command was issued, the first call duration was 57 seconds and its bandwidth – 64 (Kbps).

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11.2.3.2 SH GATEKEEPER ENDPOINTS

Purpose

To display information about the endpoints registered with this Gatekeeper.

Syntax

sh gatekeeper endpoints

Example

Figure 73: sh gatekeeper endpoints command

Explanation

The sh gatekeeper endpoints command provides the following information about each registered endpoint (peer):

IP address

Port

ID

Phone number

The command provides the total number of registered peers.

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12. Appendix C- Upgrading the Cisco ATA Software Version

This section describes how to upgrade the ATA software version. Separate upgrade files must be used to upgrade an ATA running SIP and H323.

12.1 ATA Upgrade Kit/Folder

The ATA upgrade kit or folder is supplied by the Gilat Technical Support. Figure 74 shows an example of the ATA software upgrade kit.

Figure 74: ATA Software Upgrade Kit/Folder

The ATA upgrade kit/folder (Figure 74) consists of the following files:

ATA SIP or H323 upgrade file - The format of the upgrade file name contains the ATA Firmware parameters, protocol name (SIP or H323) and the software version. The file extension is ZUP.

sata186us – an application used for upgrading the ATA.

Upgrade Commands txt file – this file contains commands used for upgrading the ATA (see Figure 75).

− The upgrade command syntax consists of the execution command, and ATA upgrade file, for example an upgrade command for ATA running H323 can be in the following format: sata186us.exe ATA030100H323041104T.zup –d1 -any2

Figure 75 shows an example of the Upgrade Commands txt file.

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Figure 75: Upgrade Commands File Content

12.2 Upgrading ATA

NOTE

Upgrading ATA requires remote site down time equal to the ATA reboot time.

During the ATA software version upgrade, the ATA IP address may be changed. If the Cisco ATA is configured with the Static IP address, the IP address will not be changed after the upgrade. If the Cisco ATA receives its IP address through DHCP, the IP address after the upgrade may change.

To upgrade an ATA to a new software version:

1. Verify that you have the correct ATA software upgrade kit. The kit is provided by Gilat Technical Support. The contents of the upgrade kit are as described in the bulleted list above and as shown in Section 12.1, page 111.

2. Establish connection between the VSAT/PC and the ATA as described in Section 3.2, page 24.

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

Result: The configuration menu enter menu number followed by the pound key, or press the pound key to exit message is announced.

4. Check the ATA IP address by pressing the [21][#] keys on the telephone keypad.

Result: The ATA IP address is announced.

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5. Verify that the PC/VSAT and ATA are visible to one another by issuing a ping command to the Cisco ATA IP address (an ATA IP address that was obtained through DHCP or the static IP address).

6. On the PC connected to the ATA, open the web browser and enter the Cisco ATA IP address in the following format: http://<Cisco ATA IP address>/dev.

Result: The ATA web configuration screen is displayed.

7. On the PC connected to the ATA, open the ATA Upgrade kit/folder.

8. Open the Upgrade Commands text file.

9. Click Start Run.

10. In the Run window, enter cmd.

Result: The Commands window is displayed.

11. At the command prompt, enter cd software.

12. Copy the relevant command from the Upgrade Commands file to the Commands window, for example:

− When upgrading an ATA running an H323 software version, copy sata186us.exe ATA030100H323041104T.zup –d1 -any2

− When upgrading an ATA running SIP software version, copy sata186us.exe ATA030100SIP040211A.zup –d1 -any2

Figure 76: Copying Upgrade Commands

Result: The ATA upgrade starts.

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Figure 77: Upgrading ATA

13. When the ATA upgrade code is displayed in the Commands window, press the ATA Function button and enter the upgrade code via the telephony keypad.

Figure 78: ATA Upgrade Command

14. Wait until the Upgrade Successful message is announced via the Voice configuration menu.

15. Reset the Cisco ATA by unplugging it from the electricity and re-plugging it.

16. Refresh your browser ATA web page.

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

18. Check the ATA IP address by pressing the [21][#] keys on the telephone keypad.

19. Verify that the PC/VSAT and ATA are visible to one another by issuing a ping command to the Cisco ATA IP address

20. Access ATA Web configuration page using the new IP address.

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13. Appendix D - Glossary

ABCD

Analog Continuous representation of voice, sound or video signals (which is not digital)

ATA Analog Telephone Adapter (Cisco)

BB Backbone protocol between the DPS and the VSAT (for IP-enabled networks).

Bit rate The number of bits that pass a given point in a network in a given amount of time, usually a second.

BoD Bandwidth On Demand (Dynamic DA) – Gilat’s access scheme.

BPS Bits per second

CBR Constant Bit Rate

Cell (1) The basic switching unit in cellular networks that carries and manages the continuity of the call by transferring it to the next call. (2) The basic switching element in ATM systems that carries data destined to reach the remote station.

Codec – Coder/Decoder

Electronically converts audio or video into digital codes and vice versa using techniques such as PCM and Delta Modulation. Also used for compression and decompression of digitized audio and video.

Compression Compression is the reduction in size of data in order to save space or transmission time. Compression is performed by a program that uses a formula or algorithm to determine how to compress or decompress data.

CPE Customer Premises Equipment

CRTP Compressed Real Time Protocol

DA Dedicated Access

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Dial Tone A signal from the telephone company that is sounded when a telephone is picked up. This signal indicates that a number can be dialed. In the event of very high load conditions, the signal may sometimes be delayed.

DPS Data Protocol Processor

DPSK Dual/Differential Phase Shift Keying Protocol

DSP Digital Signal Processing – a technique to process data from analog and digital sources by first converting to digital (if necessary) and then processing the data stream in accordance with a pre-defined program. Used to perform tasks such as data compression, decompression, and so on.

DTMF Dual Tone Multi-Frequency – describes “push button” or “touch-tone” dialing. Each digit is represented by a different frequency, and transmitted over a switched line to the local exchange where the tones are deciphered back into digits.

EFGH

E1 A European communication link, counterpart to the US T-1, that operates at a speed of 2.048 Mbps.

Echo The echo of speech heard in voice or data conversations.

Echo Cancellation Technique that isolates and filters unwanted signals caused by echoes from the main transmitted signal.

Echo Suppressor Reduces the effects of echoes in telephone connections. Works by turning off transmission in the reverse direction while one person is talking – in essence converting it to a one-way circuit.

Encryption Conversion of data into a form that cannot be decrypted without a key or a password. Decryption is the process of converting encrypted data back into its original form.

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Ethernet Ethernet is the most widely installed local area network (LAN) technology. An Ethernet LAN typically uses coaxial cable or special grades of twisted pair wires. The most commonly installed Ethernet systems are called 10BASE-T and provide transmission speeds up to 10 Mbps. Fast Ethernet or 100BASE-T provides transmission speeds up to 100 megabits per second and is typically used for LAN backbone systems, supporting workstations with 10BASE-T cards.

FTP File Transfer Protocol

GA Guaranteed Access – Gilat’s Access Scheme

Gatekeeper Also known as Call Agent or Network Administrator, its function is to identify, control, count and supervise the flow of traffic through the network gate, plus handling terminal and Gateway registration, address resolution, bandwidth control, admission control, and so on.

Gateway Processes data received through the line, such as protocol conversion, data security checks, or converting one e-mail system to another.

H.225 Is part of the H.323 standard – middleware that specifies a message set for call signaling registration and admissions, supporting call negotiations.

H.245 A control channel that controls messages governing the operation of an H.323 terminal. Includes capability exchanges, indications and commands.

H.323 A standard approved by the International Telecommunication Union (ITU) that defines a set of call control, channel setup and codec specifications for real time transmission of voice and video of various types of networks that do not offer guaranteed service or high quality of service. This standard defines the negotiation and adaptation layer for video and audio over packet switched networks.

HSP Hub Satellite Processor

Hub Central station of a network

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IJKL

IDU Indoor Unit

IF Intermediate Frequency

Inbound Signal transmitted from the VSAT to the hub

Inbound Band Group of the HSP Receivers with allocated bandwidth resources. The use of the Inbound Bands/Workgroup feature allows providing different services to different types of users over a single hub.

IP Internet Protocol

IP Address A 32-bit number that identifies each sender or receiver of information that is sent in packets across the Internet. The IP address space is divided into two parts with the left part identifying a particular network and the right part identifying a host on a network. Private IP addresses are used in private networks due to the shortage of publicly registerable IP addresses.

IP Priority Traffic of an IP address can be assigned a priority among traffic of other IP addresses.

IP Telephony A set of technologies that enables the transmission of voice, data and video over existing IP-based LANs, WANs and the Internet.

IRDP Internet Router Discovery Protocol – used for redundancy between routers or between VSATs and routers.

ISP Internet Service Provider

LAN Local Area Network

LAPU Link Access Protocol USAT – Gilat’s proprietary data link layer protocol used between the HSP and the VSATs.

LNB Low-Noise Block down-converter

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MNOP

MAC Address The unique serial number burned into Ethernet and Token Ring Network Interface Cards that identifies that network card from all others.

MGCP Media Gateway Control Protocol

MIR Maximum Information Rate

MPEG Digital video compression standards and file formats

MSK Minimum Shift Keying Protocol

NMS Network Management System

ODU Outdoor Unit

OMT Ortho Mode Transducer

Outbound Signal transmitted from the hub to the VSATs

PBX Private Branch Exchange – a private telephone network used within an enterprise that interconnects telephone extensions to each other, as well as to the outside telephone network.

PDA Partial Dedicated Access Frequency

POTS Plain Old Telephone Service – the standard telephone service. The basic telephone services with no added features. Supplies single line telephones with access to the public switched network. All POTS lines work on loop start signaling.

PPP Point-to-Point Protocol – provides dial-up access to the Internet

PSTN Public Switched Telephone Network – the world’s collection of interconnected voice-oriented public telephone networks, both commercial and government-owned. Today it is almost entirely digital, except for the final link from the central (local) telephone office to the user.

RJ-11 The most common telephone plug (jack) in the world – typically, it is a six conductor modular jack with four wires.

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RJ-45 An 8-pin connector over standard telephone wire.

QRST

QoS Quality of service

QPSK Quadrate Phase Shift Keying Protocol

RA Random Access

RFT Radio Frequency Terminal

RIP Routing Information Protocol – specifies how routers exchange the routing table information.

RPA Remote Protocol Agent

RTCP Real-Time Transport Control Protocol

RTP Real-Time Transport Protocol - Internet protocol for transmitting real-time data such as audio and video. RTP runs on top of the UDP protocol. Also supports content identification, timing reconstruction, loss detection and security.

Signaling A communication over a line between its two ends. The signaling rate over the line, which may be different from the data rate or clock, is limited by the line quality and bandwidth.

Signal Processing A combination of telephone call control and media processing. The term applies to either digital signals that are the result of sampling an analog source, or to a digital stream of data from the source itself. The procedure detects, analyzes and processes a digital bit stream into a different stream of data, usually for compression.

Silence Suppression While one person is talking on the line, the listener is silent – that means that approximately 50% of the circuit is silent, and can be filled with other transmissions – e.g. data, video, etc.

SIP Session Initiation Protocol – a simple signaling protocol for Internet Conferencing and Telephony.

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SNMP Simple Network Management Protocol

SLA Service Level Agreeement

T-1 A US standard 1.554 Mbps digital transmission link using two pairs of normal twisted wires. Often used for leased lines.

TCP Transmission Control Protocol

TDM Time Division Multiplex

TDMA Time Division Multiple Access

TFTP Trivial File Transfer Protocol

Time-slot A period of time during which certain activities are governed by specific regulators.

TVRO Television Receive Only

UVWX

UDP User Datagram Protocol

VDA Voice Dedicated Channel

VLAN Virtual LAN

VPN Virtual Private Network

VSAT Very Small Aperture Terminal

WAN Wide Area Network

Workgroup Group of VSATs with allocated hub resources. The use of the Inbound Bands/Workgroup feature allows providing different services to different types of users over a single hub.

Zone (1) In telephony - one of a series of specified areas, beyond the base rate of an exchange. (2) In LANs – the part of a LAN that is typically defined by a router.