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.
Basic VoIP and Fax over IP Configuration
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
Basic VoIP and Fax over IP

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Proprietary and Confidential
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

8.2 Enabling cRTP at the DPS .................................................................................... 83
8.3 Enabling cRTP Support at the VSATs ................................................................... 84
9. Fax Support Configuration on the SkyEdge NMS ............................................. 87

9.2 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: sh9voice9dsp command .............................................................................105
Figure 70: sh9voice9call command..............................................................................106
Figure 71: sh9h3239gateway9cause-codes command ................................................107
Figure 72: sh9gatekeeper9calls command...................................................................109
Figure 73: sh9gatekeeper9endpoints command ..........................................................110
Figure 74: ATA Software Upgrade Kit/Folder...............................................................111
Figure 75: Upgrade Commands File Content...............................................................112
Figure 76: Copying Upgrade Commands.....................................................................113
Figure 77: Upgrading ATA...........................................................................................114
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 Explains the main features of the
Features SkyEdge Basic VoIP solution.
Chapter 3 Cisco ATA Basic Provides detailed instructions on how
Configuration to establish connection with Cisco
ATA.
Chapter 4 H323 Configuration on the Explains how to configure the
SkyEdge NMS SkyEdge NMS in order to support
H323.
Chapter 5 H323 Configuration on the Provides detailed instructions on how
Cisco ATA to configure Cisco ATA to support
the SkyEdge H323 solution.

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Chapter Chapter Title Description
Chapter 6 SIP Proxy Configuration on Explains how to configure the

the SkyEdge NMS SkyEdge NMS in order to support
SIP.
Chapter 7 SIP Configuration on the Provides detailed instructions on how
Cisco ATA to configure Cisco ATA to support
the SkyEdge SIP solution.
Chapter 8 Compressed RTP Explains how to configure cRTP on
Configuration on the the SkyEdge NMS.
SkyEdge NMS
Chapter 9 Fax Support Configuration Provides detailed instructions on how
on the SkyEdge NMS to configure SkyEdge NMS to support
the Fax over IP solution.
Chapter 10 Appendix A – Remote Explains Cisco ATA configuration
Gateway Configuration and options and lists some of the most
Basic Commands useful commands.
Chapter 11 Appendix B – SkyEdge Provides some debugging
Basic VoIP System information, including Gateway and
Debugging Gatekeeper commands and Sniffer
usage.
Chapter 12 Appendix C - Upgrading Explains how to configure the Cisco
the Cisco ATA Software ATA software version.
Version
Chapter 13 Appendix D - Glossary Provides definitions for the technical
terms used in this manual.

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Conventions
This manual uses the following conventions to convey instructions and information:
Convention Description
Boldface font Commands and keywords.
Italic font The result of an instruction or command.
Screen font Information to be typed into a form or dialog
box.
9 Indicates a space in a CLI command.
WARNING
This warning symbol means danger. It is used to describe a
situation that can cause bodily injury. Before working on any
equipment, be aware of the hazards involved with electrical circuitry
and how to prevent accidents.
CAUTION
This symbol means reader be careful. In this situation, damage may be
caused to equipment or data may be lost.
NOTE
This symbol means reader take note. Notes contain helpful suggestions
and explanations.

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1. Introduction
This document describes SkyEdge Basic VoIP implementation in SkyEdge versions

4.2 and higher.
This section describes the following:
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 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.
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)
A B C
Figure 5: Remote Gateway (Rear View)

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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
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:
For G.711 codec:
For G.728 codec:
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 = 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.

Configuration 12
September, 2007
The user picks up the phone at the

Remote site.
The Phone is off hook
The dial tone is sounded. The dial

tone is generated locally at the ATA
(Remote GW)
The user dials the number and

The call is terminated
presses the pound key or waits for a
timeout of 3 seconds
The ATA sends the number to the

Gatekeeper
Gatekeeper finds a
A congestion tone is sent to the
matching IP address to No
telephone device
the dialed number
Yes
The call is terminated
Gatekeeper allocates
No
resources to the call
Yes
Gatekeeper sends the IP address of

the requested destination (GW) A busy tone is sent to the telephone
to the ATA device
ATA tries to open a VoIP session

with the requested destination
The requested
destination is free to No The VoIP session is terminated
accept the call
Yes
ATA continues the VoIP session
Figure 6: SkyEdge VoIP Call Establishment

Configuration 13
September, 2007
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.

Configuration 14
September, 2007
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.

Configuration 15
September, 2007
2. SkyEdge Basic VoIP Main Features
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.

Configuration 16
September, 2007
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.

Configuration 17
September, 2007
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.

Configuration 18
September, 2007
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.

Configuration 19
September, 2007
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.

Configuration 20
September, 2007
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.

Configuration 21
September, 2007
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:

Configuration 22
September, 2007
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.

Configuration 23
September, 2007
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.

Configuration 24
September, 2007
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.

Configuration 25
September, 2007
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.

Configuration 26
September, 2007
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
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.
ATA.

Configuration 27
September, 2007
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[#].
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

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.

Configuration 28
September, 2007
announces the commands as described in Table 2.
Table 3 lists additional voice configuration menu commands.
Table 3: Voice Configuration Menu Commands (cont’d)

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.

Configuration 29
September, 2007
4. H323 Configuration on the SkyEdge NMS
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
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
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.

Configuration 30
September, 2007
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

ConfigurationPortsDataTCP
Result: The DPS TCP parameters are displayed.

Configuration 31
September, 2007
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.

Configuration 32
September, 2007
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 CommandsReboot to reboot a non-

redundant DPS or select CommandsReboot 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
PortsEthernetTCP Profile.
Result: VSAT TCP Profile parameters are displayed.

Configuration 33
September, 2007
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.

Configuration 34
September, 2007
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.

Configuration 35
September, 2007
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 DataADA.
Result: The ADA parameters are displayed on the right.

Configuration 36
September, 2007
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

DataADART 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.

Configuration 37
September, 2007
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:
Figure 22: VSAT Templates

Configuration 38
September, 2007
Figure 23: VSAT Data Template
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:

Configuration 39
September, 2007
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.

10. In the left pane of the VSAT Data configuration window click
DataADAROT threshold.
Result: The ROT threshold parameters are displayed.
11. Click the ROT tab.
Result: The Rate of Traffic parameters are displayed.

Configuration 40
September, 2007
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 DataADART Applications.
Result: The RT Application parameters are displayed on the right.

Configuration 41
September, 2007
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.

Configuration 42
September, 2007
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
4.1.5 Configuring VoIP Filters for H323 Direct Mode
To configure VoIP filters for the H323 Direct mode:

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

RT ApplicationsVoIP 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.

Configuration 43
September, 2007
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.

Configuration 44
September, 2007
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:

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

PortsEthernetIP PrioritizationIP 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.

Configuration 45
September, 2007
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.

Configuration 46
September, 2007
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.
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 Source Destin. Destin. Port Operation
Port Port Port Start Stop
Start Stop
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.

Configuration 47
September, 2007
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 DataPortsEthernetIP Prioritization

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

Configuration 48
September, 2007
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 DataADAVoIP 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.

Configuration 49
September, 2007
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.

Configuration 50
September, 2007
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.

Configuration 51
September, 2007
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 DataLicense Management
Result: VSAT License Management parameters are displayed.
6. Verify that the VSAT Enhanced IP license is activated for the selected VSAT.

Configuration 52
September, 2007
Figure 35: VSAT Licenses
7. In the Data template, select DataPortsEthernetIP Prioritization
Result: VSAT IP Prioritization parameters are displayed.

Configuration 53
September, 2007
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.

Configuration 54
September, 2007
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
CommandsAccessReset.

Configuration 55
September, 2007
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.

following:

OR

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.

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.

Configuration 56
September, 2007
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:


RT ApplicationsVoIP Filter Instances.
default values which usually should not be modified.

Configuration 57
September, 2007
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 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.

Configuration 58
September, 2007
NOTE
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.
9. Configure port priorities in the VSAT Data template as described in

10. Save and commit the template configuration changes as described in

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.

Configuration 59
September, 2007
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)

Configuration 60
September, 2007
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.

Configuration 61
September, 2007
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).

Configuration 62
September, 2007
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 N/A N/A
Gilat configuration.
UseTftp This parameter is not relevant to N/A N/A
Tftp URL This parameter is not relevant to N/A N/A
CfgInterval This parameter is not relevant to N/A N/A
EncryptKey This parameter is not relevant to N/A N/A
EncryptKeyEx This parameter is not relevant to N/A N/A
Network Parameters
Dhcp Enables/Disables the use of DHCP Values: 1 = Use DHCP
server. 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. IP address When used, this
Used if DHCP is set to 0. parameter should
be set to the
VSAT/PC IP
address.
StaticIP Statically assigned IP address. IP address
Used if DHCP is set to 0.
StaticNetMask Statically assigned subnetmask. Subnetmask
Account Information Parameters

Configuration 63
September, 2007

Recommended
Value
UID0 Specify the telephone numbers for Alphanumeric string of
the Cisco ATA port: Phone 1. up to 31 characters.
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.
PWD0 This parameter is not used in Gilat N/A N/A
configuration.
UID1 Indicates Cisco ATA port: Phone2. Alphanumeric string of
up to 31 characters.
UID1 is defined only when G711
codec is used.
In all other cases, this parameter
should be set to 0.
PWD1 This parameter is not used in Gilat N/A N/A
configuration.
Gateway Specifies the H323 Gateway IP IP address
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.
UseLoginID Indicates which of the following will 1 – use the specified 0
be used as a login ID: LoginID
UID0 and UID1 0 – do not use the

specified LoginID.
Or
If UseLoginID is set to
LoginID0 and Login ID1 0, UID0 and UID1 are
used for
authentication.
LoginID0 This parameter is not used in Gilat N/A N/A
configuration.
LoginID1 This parameter is not used in Gilat N/A N/A
configuration.
Gatekeeper Parameters

Configuration 64
September, 2007

Recommended
Value
GkOrProxy The IP address of the primary IP address
H323 Gatekeeper.
If set to 0, Gatekeeper-routed calls
are disabled.
GkId Gatekeeper (H323) name. Alphanumeric string of
up to 31 characters
GkTimeToLive Indicates the “time to live” value Seconds
that is used when the Cisco ATA
registers with the H.323
gatekeeper.
The registration is valid until the
configured time expires.
AltGk This parameter is not relevant to N/A N/A
AltGkTimeOut This parameter is not relevant to N/A N/A
Operating Parameters
MediaPort Specifies the RTP base port that Port number 16384
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
AudioMode Enables/disables silence To enable silence 0x00100010 – by
suppression. suppression, set to default, the
0x00150015. silence
suppression is
To disable silence disabled.
suppression, set to
0x00100010.
LBRCodec Indicates the low bit rate codec Values:
(Low-bit-rate codec) that can be used during codec
negotiation. 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.

Configuration 65
September, 2007

Recommended
Value
RxCodec Indicates the Receiving audio Values:
codec preference.
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 Values:
codec preference.
be selected only if
LBRCodec is 0.
1 – G.711A
2 – G.711u
low bit rate codec can
be selected only if
LBRCodec is 3.
NumTxFrames Indicates the number of frames per To obtain 180 ms of
packet. G.723 audio, set to 6.
The frame size for each G.729 To obtain 120 ms of
data packet is 10 ms. G.723 audio, set to 4.
The frame size for each G.723 To obtain 180 ms of
data packet is 30 ms. G.729 audio, set to 18.
The frame size for G.711 is fixed at To obtain 120 ms of
20 ms per packet and is not G.729 audio, set to 12.
configurable.
ConnectMode Enables/disables fast start To enable the fast By default, this
procedure. (Connection mode of start procedure, set parameter should
the protocol used). 0x00060401. be (enabled) -
0x00060401
To disable the fast
start procedure, set
0x00060400.
3. Scroll to the bottom of the page and click apply.

Configuration 66
September, 2007
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.

Configuration 67
September, 2007
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


following:

OR

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


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:

Configuration 68
September, 2007

DataADART ApplicationsVoIP Filter Instances.

Configuration 69
September, 2007
Figure 47: Adding a New VoIP Filter
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.

Configuration 70
September, 2007
12. In the Bytes Per VoIP Packet field, specify the number of bytes per VoIP packet
NOTE
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.
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:

Section 6.2, page 68, steps 1- 5.
2. In the left pane of the VSAT Data configuration window, click

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

Configuration 71
September, 2007
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.

Configuration 72
September, 2007
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.
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 Source Destin. Destin. Port Operation
Port Port Port Start Stop
Start Stop
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

Configuration 73
September, 2007
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
14. Save and commit the template configuration changes as described in

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.

Configuration 74
September, 2007
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
− 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)

Configuration 75
September, 2007
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.

Configuration 76
September, 2007
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

Recommended
Value
Parameters for Configuration Method and Encryption
These parameters are not relevant to Gilat Configuration
UIPassword This parameter is not relevant to N/A N/A
Gilat configuration
UseTftp This parameter is not relevant to N/A N/A
Gilat configuration
Tftp URL This parameter is not relevant to N/A N/A
Gilat configuration
CfgInterval This parameter is not relevant to N/A N/A
Gilat configuration
EncryptKey This parameter is not relevant to N/A N/A
Gilat configuration
EncryptKeyEx This parameter is not relevant to N/A N/A
Gilat configuration
Network Parameters
Dhcp Enables/Disables the use of DHCP Values: 1 = Use DHCP
server. 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. IP Address When used, this
Used if DHCP is set to 0. parameter should
be set to the
VSAT/PC IP
address
StaticIP Statically assigned IP address. IP Address
StaticNetMask Statically assigned subnet mask. Subnet mask
Account Information Parameters

Configuration 78
September, 2007

Recommended
Value
UID0 Specify the telephone numbers for Alphanumeric string of
the Cisco ATA port: Phone 1. up to 31 characters.
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.
PWD0 This parameter is not relevant to N/A N/A
Gilat configuration
UID1 Indicates Cisco ATA port: Phone2. Alphanumeric string of
up to 31 characters.
UID1 is defined only when G711
codec is used.
In all other cases, this parameter
should be set to 0.
PWD1 This parameter is not relevant to N/A N/A
Gilat configuration
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 N/A N/A
Gilat configuration
GkTimeToLive This parameter is not relevant to N/A N/A
Gilat configuration

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Recommended
Value
UseLoginID Indicates which of the following will 1 – use the specified 1
be used as a login ID: LoginID
UID0 and UID1 0 – do not use the

specified LoginID.
Or
If UseLoginID is set to
LoginID0 and Login ID1 0, UID0 and UID1 are
used for
authentication.
LoginID0 This parameter is not relevant to N/A
Gilat configuration
LoginID1 This parameter is not relevant to N/A
Gilat configuration
SIPPort Indicates the port through which Integer from 1 to 5060
the Cisco ATA listens for incoming 65535
SIP requests and sends outgoing
SIP requests.
SIPRegInterval Indicates a number of seconds Seconds from 1 to
between the Cisco ATA 86400
registration renewal with the SIP
Proxy server.
To avoid registration expiration,
the Cisco ATA renews the
registration some time earlier than
the specified timeout.
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 Integer from 1 to 16384
Cisco ATA uses to receive RTP 65535
media streams.
AudioMode Enables/disables silence To enable silence 0x00150015 – by
suppression. suppression, set to default the silence
0x00150015. suppression is
enabled.
To disable silence
suppression, set to
0x00100010.

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Recommended
Value
LBRCodec Indicates the low bit rate codec Values:
(Low-bit-rate codec) that can be used during codec
negotiation. 0 – select G.723 as
low-bit-rate codec.
low bit rate codec.
6 – select G.726-
32kbps as the low bit
rate code.
RxCodec Indicates the Receiving audio Values:
codec preference.
be selected only if
LBRCodec is 0.
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 Values:
codec preference.
be selected only if
LBRCodec is 0.
1 – G.711A
2 – G.711u
low bit rate codec can
be selected only if
LBRCodec is 3.
6 - G.726 – 32 kbps
NumTxFrames Indicates the number of frames per To obtain 180 ms of
RTP packet that the Cisco ATA G.723 audio, set to 6.
transmits.
To obtain 120 ms of
The frame size for each G.729 G.723 audio, set to 4.
data packet is 10 ms.
To obtain 180 ms of
The frame size for each G.723 G.729 audio, set to 18.
data packet is 30 ms.
To obtain 120 ms of
The frame size for G.711 is fixed at G.729 audio, set to 12.
20 ms per packet and is not
configurable.

Configuration 81
September, 2007

Recommended
Value
ConnectMode Enables/disables fast start To enable the fast By default, this
procedure. (Connection mode of start procedure, set parameter should
the protocol used). 0x00060401. be (enabled) -
0x00060401
To disable the fast
start procedure, set
0x00060400.
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
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 ConfigurationPortsDataIP
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 CommandsReboot to reboot a non-

redundant DPS or select CommandsReboot 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:

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6. In the left pane of the VSAT Data Template configuration window, click
PortsEthernetIP 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.
NOTE

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September, 2007
9. Fax Support Configuration on the SkyEdge NMS
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:

Configuration 87
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7. Set the ADA Operation Scheme parameter to Enhanced DA.
Result: The Enhanced (Dynamic) DA parameters are displayed.
Figure 62: Data Template Dynamic Parameters

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

DataADADynamic DA.
Result: The Dynamic DA parameters are displayed.

Configuration 88
September, 2007
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.

Configuration 89
September, 2007
NOTE
9.3 Configuring Fax Support in a System with Legacy DA
To configure a SkyEdge system to support fax in a Legacy DA environment:
7. Under General, set the ADA Operation Scheme parameter to Legacy ADA.
Result: The Legacy ADA parameters are displayed.

Configuration 90
September, 2007
Figure 65: VSAT Data Template – Legacy ADA

9. Click DataADART 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
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
14. Click Save.
15. Click Commit.

Configuration 92
September, 2007
NOTE

Configuration 93
September, 2007
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.
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.

Configuration 95
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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 2abcABC
3 3defDEF
4 4ghiGHI
5 5jklJKL
6 6mnoMNO
7 7pqrsPQRS
8 8tuvTUV
9 9wxyzWXYZ
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.

Configuration 97
September, 2007
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.

Configuration 98
September, 2007
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.

Configuration 99
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To refresh and display the most recent Cisco ATA configuration:
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:
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.

Configuration 100
September, 2007
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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September, 2007
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.

Configuration 102
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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 $giLat$.

Configuration 103
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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
sh9voice9port Display E1 configuration
sh9voice9call Display E1 call status
sh9voice9dsp Display E1 DSP status
sh9call9active9voice Display all active calls
sh9call9history9voice Display calls stored in the voice history table
sh9call9resource9voice9stats9dsp Display the voice DSP resources utilization statistics
sh9dial-peer9voice Display configuration of the Voice Encapsulated
Peers (POTS or VoIP)
sh9gateway Display the VoIP gateway status
sh9h3239gateway9cause-codes Display the disconnect cause codes and statistics
sh9h3239gateway9h225 Display the H.225 Gateway statistics

Configuration 104
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11.2.2.1 SHOW9VOICE9DSP
Purpose
To display the Digital Signal Processes (DSP) status and configuration.
Syntax
sh9voice9dsp
Example
Figure 69: sh9voice9dsp command
Explanation
The sh9voice9dsp 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 sh9voice9call command
(see Section 11.2.2.2, page 106).

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11.2.2.2 SHOW9VOICE9CALL
Purpose
To display the system call setup.
Syntax
sh9voice9call
Example
Figure 70: sh9voice9call command
Explanation
The sh9voice9call 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.

Configuration 106
September, 2007
11.2.2.3 SHOW9H3239GATEWAY9CAUSE-CODES
Purpose
To display the call disconnect cause codes and statistics.
Syntax
sh9h3239gateway9cause-codes
Example
Figure 71: sh9h3239gateway9cause-codes command
Explanation
The sh9h3239gateway9cause-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.

Configuration 107
September, 2007
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
sh9gatekeeper9calls Display current Gatekeeper call status
sh9gatekeeper9endpoints Display all endpoints registered with this
Gatekeeper
sh9gatekeeper9gw-type-prefix Display Gateway Technology Prefix table
sh9gatekeeper9performance9stats Display Gatekeeper performance data
sh9gatekeeper9status Display the Gatekeeper status
sh9gatekeeper9zone9status Display all zone statuses known to this gatekeeper

Configuration 108
September, 2007
11.2.3.1 SH9GATEKEEPER9CALLS
Purpose
To display the current status of the Gatekeeper calls.
Syntax
sh9gatekeeper9calls
Example
Figure 72: sh9gatekeeper9calls command
Explanation
The sh9gatekeeper9calls 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).

Configuration 109
September, 2007
11.2.3.2 SH9GATEKEEPER9ENDPOINTS
Purpose
To display information about the endpoints registered with this Gatekeeper.
Syntax
sh9gatekeeper9endpoints
Example
Figure 73: sh9gatekeeper9endpoints command
Explanation
The sh9gatekeeper9endpoints 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.

Configuration 110
September, 2007
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.

Configuration 111
September, 2007
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

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.

Configuration 112
September, 2007
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 StartRun.
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.

Configuration 113
September, 2007
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.

Configuration 114
September, 2007
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.

Configuration 115
September, 2007
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 – Electronically converts audio or video into digital codes and

Coder/Decoder 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

Configuration 116
September, 2007
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.

Configuration 117
September, 2007
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

Configuration 118
September, 2007
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

Configuration 119
September, 2007
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.

Configuration 120
September, 2007
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.

Configuration 121
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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.

Configuration 122
September, 2007

Basic VoIP

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Basic VoIP

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Configuring Basic Voice over IP and

Fax over IP Solutions

Document No. DC-4272-10

Gilat Satellite Networks Ltd.

2. SkyEdge Basic VoIP Main Features....................................................................16

3. Cisco ATA Basic Configuration ..........................................................................24

Basic VoIP and Fax over IP

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

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

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

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

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

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

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

Basic VoIP and Fax over IP

10.2 Remote Gateway Commands ................................................................................99

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

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

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

Basic VoIP and Fax over IP

Figure 1: SkyEdge VoIP Network Architecture ................................................................ 2

Basic VoIP and Fax over IP

Figure 46: VoIP Filter Instances ....................................................................................69

Basic VoIP and Fax over IP

Table 1: Voice Configuration Menu Commands ............................................................ 26

Basic VoIP and Fax over IP

About This Manual

How to Use This Manual

This manual is to be used as a reference guide by trained personnel. It is not

Chapter Chapter Title Description

Chapter 1 Introduction Provides an introduction to the

Basic VoIP and Fax over IP

Chapter Chapter Title Description

Chapter 6 SIP Proxy Configuration on Explains how to configure the

Basic VoIP and Fax over IP

Basic VoIP and Fax over IP

This document describes SkyEdge Basic VoIP implementation in SkyEdge versions

This section describes the following:

 SkyEdge VoIP Network

 SkyEdge VoIP Network Components

 SkyEdge VoIP Solution Description

 SkyEdge VoIP Calculations

 Basic VoIP Network Call Flow

 Approved VoIP Components

1.1 General Overview

1.2 SkyEdge VoIP Network

The VoIP packets destination can be one of the following:

 Residential Gateway (another VSAT)

 Hub VoIP phone

Basic VoIP and Fax over IP

 The Hub can communicate with an ISP connection to provide VoIP

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

Figure 1: SkyEdge VoIP Network Architecture

Basic VoIP and Fax over IP

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

1.3 SkyEdge VoIP Network Components

The standard SkyEdge VoIP Network contains the following components:

− Provisioning server (TFTP)

 At the VSAT site:

− Remote VoIP Gateway (Cisco ATA-186)

SkyEdge VoIP Network

SkyEdge VoIP Network Components

SkyEdge VoIP Solution Description

SkyEdge VoIP Calculations

Basic VoIP Network Call Flow

Approved VoIP Components

Residential Gateway (another VSAT)

Hub VoIP phone

The Hub can communicate with an ISP connection to provide VoIP

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

At the VSAT site:

Cisco 3725 router to support 60 simultaneous calls

Cisco 3745 router to support 120 simultaneous calls

Cisco 5400 router to support 240 simultaneous calls

E1 - An E1 line can carry 30 digitized voice channels

FXO---Foreign Exchange Office interface. The FXO interface is an RJ-11

FXS---The Foreign Exchange Station interface. This interface is an RJ-11

Translates phone numbers into IP addresses and vice versa.

Assigns resources to calls

Monitors the operation of Remote Gateways

0.5 G RAM memory

Standard VGA card

Minimum of 20 G Byte hard disk space

One Ethernet Port of 10/100

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

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

(C) 12V power connector

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

Ability to close an H.323/SIP connection for which there is not enough

SIP is fully supported.

H323 Direct mode is supported.

Use this algorithm to calculate the codec for VoIP requirements.

Time slot duration

S = bytes per voice frame

T = voice frame per time

MI – bytes per time slot.

For G.723 (6.3) codec:

For G.729 codec:

For G.711 codec:

For G.728 codec:

Time slot duration – 7.9967 ms

Multi slot – 15 slots

MI – bytes per time slot – 250 bytes

Example for G.723 (6.3) codec: