EZ-Phone

(HSS SIP UA)

User’s Manual
HSS-BLR-SDC-ICP-0048

Revision: 1.0

Hughes Software Systems Ltd.

Prestige Opal, 146 Infantry Road
Bangalore 560 001, Karnataka, INDIA
91-80-228-6390
EZ-Phone v1.0 User’s Manual

COPYRIGHT INFORMATION

 Hughes Software Systems Limited 2001

All rights reserved. No part of the software may be copied, reproduced, stored in any
retrieval system, or transmitted in any form or by any means, either electronically,
mechanically, or otherwise without prior written permission.

TRADEMARKS
All the brand names and other products or services mentioned in this document are
identified by the trademarks or service marks of their respective owners.

DISCLAIMER
The information in this document is subject to change without notice and should not be
construed as commitment by Hughes Software Systems. Hughes Software Systems
assumes no responsibility or makes no warranties for any errors that may appear in this
document and disclaims any implied warranty of merchantability or fitness for a
particular purpose.

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Table of Contents

1. INTRODUCTION .................................................................................................................................... 4

2. FEATURES............................................................................................................................................... 4

3. TAB CONTROLS .................................................................................................................................... 5

A) KEYPAD .............................................................................................................................................. 5
B) ONLINE BUDDY TAB ........................................................................................................................... 6
4. BUTTON CONTROLS............................................................................................................................ 7
MAKE A CALL .............................................................................................................................................. 7
HANG UP ................................................................................................................................................. 8
AUDIO CONTROL ......................................................................................................................................... 8
DO NOT DISTURB........................................................................................................................................ 8
CALL HOLD ................................................................................................................................................. 8
5. MENU CONTROLS ................................................................................................................................ 9
FILE............................................................................................................................................................. 9
C) EDIT .................................................................................................................................................. 10
D) VIEW ................................................................................................................................................. 14
E) HELP ................................................................................................................................................. 14

6. STATUS MESSAGES............................................................................................................................ 16

7. ERROR MESSAGES ............................................................................................................................. 17

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User Manual
For
HSS EZ-Phone Ver 1.0

1. Introduction
The Session Initiation Protocol (SIP) is an application-layer control protocol that can
establish, modify and terminate multimedia sessions or calls. These multimedia
sessions include multimedia conferences, distance learning, Internet telephony and
similar applications. SIP can invite both persons and "robots", such as a media storage
service.

The HSS EZ-Phone is a SIP client for Windows NT built upon the SIP stack of Hughes
Software Systems.

2. Features
EZ-Phone (Evaluation Version) supports the following features:

ΠUDP support
ΠCall Hold
ΠCall Forward all calls
ΠCall Forward when busy
ΠDo Not disturb
ΠChoice of GSM, ADPMCM and u-Law codecs
ΠBuddy List

Note: This evaluation version of EZ-Phone uses SpeakFreely for transmitting/receiving

media.

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3. Tab controls
The following are the tab windows present in the client.

Keypad Online Buddy List

a) Keypad

The keypad provides a mouse-click interface for entering the SIP URL in the textbox.
Presently it supports only digits from 0-9, “.” and “:”. The client automatically adds the
“sip:” prefix if it is not provided.

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b) Online Buddy Tab

This shows the list of buddies. A green bullet next to the name indicates those buddies
who are online (Online means that the remote user’s SIP client is up). Similarly, a red
bullet is used to indicate buddies who are offline.

One could keep a list of '

buddies'to call, instead of remembering the IP every time. New
buddies can be added using the “Add New Friend” option.
For example, if one want to create a buddy called ' joe', we need to:
• Go to Edit | Add New Friend.
• In the dialog box, enter IP/host , port, and the name of the user (in this case Joe)
• Check the Add to Buddy List check box if one wants to show this in the buddy list (If
not checked, then it will be a part of the address book, but not buddy list)

Now, one can click on the buddy list cue tab, and double click on the name to call.

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4. Button Controls
When a user places the mouse pointer on one of the buttons, tool tip for that button is
shown. The following lists the buttons’ tool tips and functionality it provides to the user.

Make a Call

The User can call any other remote client which supports the SIP protocol by
entering the remote client’s SIP URL in the text box provided (through the keypad
on one of the tab controls or through the keyboard) and clicking on this button.
(Note that a call can also be initiated by entering the URL and hitting “enter” in
the Call box.

For example, one could type any of:

sip:[email protected]:port

OR

company.com:port

Note that if no port is specified a default of 5060 is taken

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Hang up

The User can terminate an already established call by clicking on this button. In
case the user is currently in two calls (using the call hold feature), clicking this
button brings up a dialog giving the user the option to hang up one of the people
he is connected to.
Also, the user can reset his client at any time (when he is not in a call-established
state) by clicking this button.

Audio control
The User can adjust the preferred and playback settings by clicking on this
button.

Do not Disturb

This is a toggle button. On clicking this button, the icon changes to an ear with a
red cross. In this mode, the client will automatically reject incoming calls. Press
again to disable.

Call Hold

This button is provided to the user so that he can switch between two calls. It
works as follows:
The user has an established call with two remote clients in place. The calls may
have been initiated by other remote clients or by the user himself. The user can
alternately switch between these two established calls by clicking the Call Hold
button. The client currently supports only two simultaneous calls.

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5. Menu Controls
The Menu bar appears in a separate window along with the EZ-Phone. This Menu
Dialog can be closed by clicking the close button in the title bar. To access the Menu
Dialog again just right click anywhere on the EZ-Phone. The following describes the
functionality of the various menu options provided:

Menu Dialog

File
i) Register
This feature is not available in the evaluation release.

ii) Skin Browser

Use this option to select different skins for the client. The user could add any
other skins by simply creating a new directory and inserting the new bitmaps with
the following naming convention:

main.bmp – skin for the main client

volctrl.bmp – skin for the volume control
common.bmp – skin for the rest of the dialogs.

iii) Reset
Use this option to reset the client.

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iv) Quit
When this menu item is selected, it exits the SIP client.

c) Edit
i) Address Book

When the user clicks on this menu item, a dialog pops up with the first entry in
the address book. The user can navigate the addressbook using the “Previous”
and “Next” buttons. He can delete any entries by navigating to a particular entry
and clicking on the “Delete” button. He can add a particular entry to the buddy list
by checking the “Add to Buddy List” option. When he clicks on the “Done” button,
it closes the pop up dialog.

ii) Add New Friend

Explained earlier (Online Buddy Tab)

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iii) Configurations

When the user clicks on this menu item, a dialog pops up using which the user
can configure the following
• The port on which he wants to receive the SIP signals
• The protocol to be used for exchanging SIP signals (This feature is not
available in the evaluation release)
• If a proxy is used, the proxy’s address and port number (This feature is not
available in the evaluation release)
• If a gateway is used, the gateway’s address (This feature is not available in
the evaluation release)
• The frequency of spooling of buddies (for displaying the buddies who are
online in the tab control)

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iii) Media Preferences

When this menu item is selected, a dialog pops up allowing the user to specify
the codecs which he supports.

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v) Forwarding

The user is provided a facility to forward incoming calls automatically to another

SIP address. He can customize it further by checking one of the boxes for
• forwarding calls when busy(when the user has already established a call)
• forwarding calls at all times.

vi) Jitter Compensation

This option can be used to set the jitter compensation during conversation
(audio).

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d) View
i) Debug Window

This allows the user to view the complete flow of the SIP signaling messages
exchanged between the two or more parties through a dialog window that pops
up. It has a “Clear” button that can be used to periodically clear the contents of
the window. The dialog can be closed by clicking on the “Close” button.

e) Help
i) About

This pops up a dialog with the details of the client.

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ii) Quick Help

This gives a brief description and quick tips for usage of the EZ-Phone.

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6. Status messages

The following are the status messages that appear in the status bar during the operation
of the EZ-Phone.

• Trying to call yourself!: When the user tries to call himself (this is not allowed)

• Ready: This indicates that the EZ-Phone is ready for use. It also indicates that the
client is currently idle (not in any call) or has been reset.

• Ringing: This indicates that the SIP client at the remote end is ringing (The remote
user is being alerted)

• Calling Remote User: This indicates that the remote SIP client is being invited to a
call.

• Call Established: This appears once a connection has been established.

• Attempting to answer call: This appears when the user accepts an incoming call.

• Hung up: This appears after a call has been terminated.

• Remote user is Busy: This appears when the remote user is either busy or rejects
a call.

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7. Error messages

The following are the error MessageBoxes that pop up during the operation of the SIP
client.

Error Message Text Reason

“Please enter a SIP URL” This appears when the user attempts to call up without
entering a SIP URL.
"The destination address should be of This appears when the user attempts to call up using an
the form sip:[username]@<host>[:port]" invalid (syntax) SIP URL.

"Unable to get remote client's host This appears when the client is unable to resolve the
address." remote hostname to a valid IP address.
Retransmissions Timed Out when trying This occurs when the client is unable to send a SIP
to send <msg>. Terminating the call. message to a remote client after 7 retransmissions.
Spool Time should be atleast 60 This appears when a user tries to set the buddy list spool
seconds. time to less than 60 seconds.
Please fill the SIP URL to forward calls This appears when the user tries to set forwarding options
to without specifying a URL.
Please select atleast one media type This appears when the user does not select any media
type from the Media Preferences dialog. Atleast one
codec should be supported.
Cannot modify Jitter Compensation This appears when the user tries to modify the Jitter
during a call. compensation value during a call. Jitter compensation
value can be modified only when the client is idle.
Please fill all the fields This appears when the user leaves any of the fields in the
addressbook blank while navigating the addressbook.

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