MizuPhone VOIP SoftPhone
User
Guide
Contents
6.8. Sending and sharing files
7.1. I can’t receive incoming calls
7.2. The voice are cutting. How can I improve the
voice quality?
7.3. How to call other UA directly (bypassing the
sipserver)
7.4. How to monitorize SIP messages
7.5. How to clear my saved login password
Version
MizuPhone v0.9.2 User Guide
Revisited July 17, 2008
Copyright
This document is copyrighted by
MizuTech SRL.
Copyright ©2008 MizuTech SRL
Disclaimer: MizuTech SRL. reserves
the right to change any information found in this document without any written
notice to the user. No warranty is made in regard to specifications or features.
License Agreement
You must accept the license
agreement (LicenseAgreement.txt) before you use any MizuTech software!
Trademark Acknowledgement
LINUX is a
registered trademark of Linus Torvalds in the United States and other
countries.
Windows is a trademark of Microsoft
Corporation, registered in the United States and other countries.
OpenH323 are licensed under MPL.
VideoLAN is a software project, which
produces free software for video, released under the GNU General Public License (GPL
version 2).
UltraVNC is licensed
under the the GNU General Public License.
Some video or audio codecs usable
by MizuPhone requires patent
license (per channel) unless it is used in pass-through mode.
For modules licensed under the the
GNU General Public License, the source
code is included on the install CD.
Other logos, products, brand
names and service names contained in this document are the property of their
respective owners (trademarks or registered trademarks of their respective
companies)
MizuPhone is a VOIP softphone based on the open
standard SIP protocol. With MizuPhone you can connect to any SIP (proxy and/or
registrar) server on the public internet or on your local area network.
MizuPhone’s puprose is to combine the SIP
compatibility with P2P inteligence with an intuitive user interface. It is
fully interoperable with most of the VOIP service providers, VOIP software and
hardware (workstations, notebooks, PDAs, IP-Phones, etc). Mizu-Phone is higly
optimized. Doesn’t eat your system resources and doesn’t disturb you when you
work. When you use it for HD quality video calls, it can take advantage of
dual-core systems.
ü VOIP
calls
ü Multiple
accounts, multiple SIP server registrations.
ü Ultra
WideBand codec
ü HD
quality video calls (depending on your camera and bandwidth)
ü Instant
messaging and presence using the SIMPLE protocol
ü Presence,
session timers
ü Mute,
Hold, Redial, Transfer, Forward, Conference
ü Local and
remote voicemail
ü File transfer
and file sharing (compaitibile with any SIP server)
ü History
(with audio and video records)
ü Acoustic echo
cancellation, automatic gain control, VAD, Denoise fillter, Auto QoS, Dynamic
Jitter
ü Codecs:
G711 (PCMU,PCMA), , G.723.1, G729, iLBC, Speex, MPEG1, MPEG4, Theora, DIV3,
MJPG, H263, H264
ü Fax (beta
version)
ü Network
handling: UPNP, STUN, ICE, firewall and NAT detection
ü Transport
protocolls: UDP, TCP, TLS
ü XCAP,
WebDAV, FTP and HTTP profile storage
ü LDAP,
Outlook, WAB, Vcard, CSV contact import
ü DTMF (Inband
DTMF or SIP INFO messages)
ü Balance/credit
display, Microsoft Outlook synchronization, LDAP, WAB contactlists
ü Full
encypted communications (protocoll and media too)
ü Intelligent
P2P based network path detection (will work even if the server is down)
ü Customizable
interface and language
ü RFC 2543
Compatibility
ü RFC 3261
Compatibility
ü RFC 3262
Reliability of Provisional Responses in Session Initiation
ü RFC 2976
The SIP INFO Method
ü RFC 2617
HTTP Authentication
ü RFC 3891
"Replaces" Header
ü RFC 3325
Private Extensions to the Session Initiation
ü RFC 2778
A Model for Presence and Instant Messaging
ü RFC 3428
Session Initiation Protocol (SIP) Extension for Instant Messaging
ü RFC 3263
Locating SIP Servers
ü RFC 3265
Specific Event Notification
ü RFC 3420
Internet Media Type message/sipfrag
ü RFC 3515
Refer Method
ü RFC 3311
UPDATE Method
ü RFC 3581
Symmetric Response Routing
ü RFC 3842
Message Summary and Message Waiting Indication Event Package
ü RFC 1889
RTP: A Transport for Real-Time Applications
ü RFC 2190
RTP Payload Format for H.263 Video Streams
ü RFC 2327
SDP: Session Description Protocol
ü RFC 2833
RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
ü RFC 3264
An Offer/Answer Model with Session Description Protocol
ü RFC 3550
RTP: A Transport Protocol for Real-Time Applications -replaces RFC 1889
ü RFC 3555
MIME Type Registration of RTP Payload Formats
ü draft-ietf-mmusic-ice-02
A Methodology for NAT Traversal for Multimedia Session Establishment Protocols
ü draft-ietf-avt-rtp-ilbc-04
ü draft-ietf-sipping-cc-transfer
Call Control - Transfer
ü draft-ietf-sip-referredby-05
ü Custom
protocol extensions
ü and many
others
Minimum
system requirements: Intel Pentium II 500 MHz, 128 MB RAM, Windows
2000, XP or Vista, Sound Card, IP network connection, Internet Explorer and a
headset.
For HD
video calls you need a HD quality camera device, directx8
compatible VGA card and a modern processor (2 GHz).
Download the install package from www.mizutech.hu and follow the
instructions. The installation will take only 2 minutes.
On the login
screen, type your username and password. If you are a first-time user, click on
the "New user" label and enter any username/password. This
username/password will protect your data (settings, history, etc) on the local
computer and this will be your MizuTech login also (the MizuTech server will
give you its SIP service and profile storage for free).
After login, you can set-up your sip account(s) if you would like to
call traditional landline or mobile phones (the free MizuTech SIP server will
route calls only between MizuPhone users). This can be done from Menu ->
File -> Settings -> Accounts ->
Now you can:
Ø set-up
your details (Menu -> File -> My Contact Details). At least setup your
display-name properly.
Ø add your
contacts (Menu-> Contacts -> Add new contact)
Ø call directly a name or phone number
("Dial" tab)
Ø send
instant message, files or initiate video calls (right click on a contact)
The MizuPhone configuration can be accessed from
Menu -> File -> Settings. There are settings which belong to the selected
account or which are accont independent. The most important settings are the Accounts
setttings where you can add your sip service providers. Advanced/Basic view can
be changed by the “Show advanced settings” checkbox.
On the Application section you can set the application behavior. In the Network section the local endpoint
settings can be set. The local IP and ports are discovered automatically if you
don’t set it explicitly. The MizuPhone has many built-in features that help to
properly route your IP packets, NAT discovery, automatic firewall
configuration, UPNP discovery, STUN and ICE technologies help for proper call
routing. If you have problems with NAT/firewall traversal, set the local signalling
and rtp ports explicitly and make a rule in your NAT/firewall device to bind
your ports properly (known as virtual server settings). You usually need to
check only “UDP” as the listening protocol because very few SIP server connects
to you by TCP or TLS.
The Accounts section is the most important one. Here youn can define
your sipservers and login settings. In the “Account Name” you can type anything
or leave it empty. The “Server Address” must be filled properly as your VOIP
provider requires it. It can be a domain name or an IP address. The standard
SIP port is 5060, so you usually don’t have to change it unless your provider says othervise. Leave the “Proxy Address” empty if you don’t
know its meaning! The username and password will be provided by your VOIP
provider. The username can be a phone number too. You can set other details by
clicking on the “Edit my profile” link (For example it is very useful to set
your display name properly because it doesn’t have to be the same with your
username wich is usually a phone number). When you configure the built-in
MizuTech account, you cannot simply rewrite your password. The password change
must be handled by the server, so you have to hit the “Change password” and
follow the instructions. You can load the server settings from a config file
(if you have such file) by clicking on the “Load settings from template”.
Phone numbers can be normalized
(insert/remove prefix/suffix) on the Dial
Plan section. For example if you enter the number 1111 but you would like
the dialed number to be 991111 you have to set the “Original number or prefix”
as 1111, and the “Add to begin” to 99,
than hit the “Add” button.
On the SIP Settings tab some advanced SIP protocol related settings can be
done. Set the “keep-alive interval” to a low value, because this will keep your
NAT open by periodically sending the message configured by “Keep-alive message”
to the server. Uncheck the “Register interval” checkbox if you don’t want
registrations at all, but take care because most of the SIP servers will
require periodic reregistrations! The “presence interval” means how often your
contacts status will be required. But they will send it automatically on every
change, so there is no reason to set it shorter than 600 sec (10 minute). The
“session timer” helps to detect disconnected calls. “Cache DNS records” will
speed up DNS requests. Uncheck it only if you have problems with it (your
server or contacts change its domain name very often). If the “Encrypt
communication” is checked, the softphone will encrypt the media and/or the
signaling whenever the other endpoint supports encryption. If the transport is
encrypted you will see a small lock icon in the bottom-right corner of the
application. Most SIPservers use UDP as the transport protocol, so change it
only if you are sure.
On the Codecs tab the audio and video codecs properties can be changed
such as enable/disable, priority, frame/packet and video bitrate. Be sure to
use pattented codecs only if you have license for it. Additional codecs can be
added as plugins.
You can change your audio and video devices
(soundcard and webcam) and RTP settings on the Devices tab. If you have only one audio and video codec, than you
can leave the microphone, speaker and webcam settings as “default”. Othervise
set your preferred devices. The AGC (automatic gain control) feature is very
useful when your peers have too loud or noisy audio volume. The AEC (acoustic echo
cancellation) helps to reduce the echo from your speaker. You have to leave the
Auto QoS option usually checked for proper prioritization of the UDP packets.
The QoS will have more significant result if your routers do support QoS. The “Silence
supression” can save some bandwith but this can
result in disconfort for your peers (they will hear totally silent periods
without background noise). You can finetune you dynamic jitter buffer setting
by changing the jitter values. Make a bigger jitter if you hear cropps in the
voice, and make it lower if you experience delays in the voice. The “audio
settings” button is a shortcut to the windows audio control panel applet.
You can change the default
ringtone and other sounds on the Sounds
section. Wave and mp3 files are accepted. On the Archive section you can change how much time the history will be kept
for you. Turn off the voice recording if your harddisk doesn’t have enough
space. Finetune your blacklist and other security settings on the Privacy section. On the Messaging section you can set the DTMF
type to In-Band, INFO or In-Band and INFO.
Check the “Send typing notifications” checkbox if you would like your
peers to know when you have started writing a chat message. You can specify a short text to be appended
after every SMS messages sent by you (usually your name, because the target
wouldn’t know who is the sender of the message. Finally you can set the mode
your chat messages will be sent. By pressing the Enter button, or only in
combination with Ctrl. On the Voicemail
tab you can configure a caller number from wich all calls will be accepted
automatically. Set it as long as possibile. It is useful for example when you
are on vacation, you call your softphoe and see what’s happening in your house.
On the Storage tab you can define a
“Shared directory” to be accessible by your contacts without your permission.
It is very useful for you documents, pictures and music that you want to share
with your colleagues, friends or family members. You can also set a remote
location where your profile and contactlist will be stored (FTP, HTTP, WebDav
and XCAP servers are supported)
Quick tasks such as contact management, view modes
can be accessed from the program menu. From the File menu you can LogOff and LogOn as a different user, change your
status, access your configuration and
see your contact details or exit the application. From the View menu you can change the sort order of the contacts, show/hide
some modules or set the phone always on top of other applications. From the Contacts menu you can manage your
contacts: add new contact, search for peopels, delete, edit, backup and
categorize your contacts. From the Help menu you can access the documentation,
see the “about box” with your program verzion and access important links (from
where to buy, project webpage, support contact email address). When you receive
the product key you can add it from the “Enter activation code” menu.
In the StatusBar you can change your online
status, see your sipservers status and check important system messages and logs.
The most important tabs are the followings: Contacts: here you can see you contacts and its presence (online/offline status). Right click on a contact to select the desired action (call, send message, send file, etc)
Dial: when
you want to call ordinary numbers (a people that are not on your contact list),
you can do a quick dial here.
History: here
you can see your missed calls and other events.
Settings:
accesibile from Menu -> File -> Settings. You can configure your
softphone on this tab. For more details see the Configuration chapter.
Messenger
window: can be placed on a tab or in a separate window (depending from your
configuration). Here you can send chat and SMS messages for peopels or groups.
Contact
details: in this tab you can see or edit your contacts profile.
Add to
conference: on this tab you can select people to add to
conference (IM or call). If the people are not in your contact list, you can
enter their number directly
Call tabs: every
session are displayed on its separate page. Caller ID, call status and time
will be displayed here. From the actions rigth-click menu you can add more
participants, add video, hold the call or do other actions.
MizuPhone can be started in multiple ways:
Ø Started
automatically with windows. Click on the MizuPhone icon on the bottom-rigtht
corner of the desktop
Ø Use
windows start menu or double-click on the desktop icon
Ø Pressing Ctrl+Alt+M
Ø Click on a
browswer hyperlink. Example: <a href="sip:4444@domain.com">call
displayname</a>
Ø Double
click on a file with .sip extension. The file must contain an URI or must be a
configuration file
Depending on your version, the login
username/password can mean the followings:
Ø Username/password
to protect your local profile
Ø MizuTech
username/password (for free MizuTech SIPserver and profile storage)
Ø Username/password
for your VOIP provider sipserver
If you don’t already have a MiziTech account, click
on the “New user” link and enter your username/password. Your new account will
be created in 5 seconds. If the “Create SyTech account” checbox is not checked,
only a local profile will be created to protect your configurations and history
from other users on the same computer.
If you use the softphone on your
private computer, you can check the “Remember my password” and the “Login
automatically” checkbox, but if you use it from a public place, make sure, that
these checkboxes are cleared and always click on Menu -> File -> LogOff
when you are finished.
Change the language for your
needs (currently only English and Hungarian are available) and hit the “Login”
button to begin.
You can initiate voice calls in multiple ways:
Ø Right
click on a contact and choose “Call”
Ø Click on
the green call button
Ø Click on
a phone numbers link to begin a call to the selected number (this is useful
when a contact has more than one number)
Ø You can
configure to call the contact when you double-click on it (the default double-klick
action is chat message)
Ø From the
Menu -> Action choose Call
Ø Switch to
the Dial tab, and enter a phone number or username.
You can also
enter the full URI. Eg: username@address:port, where the username can be
a phone number also. The address can be a domainname or an IP address. If the
port is missing, then the default port is used which is 5060 for the SIP
protocol. On the dial tab you can quickly select your last called numbers from
the phone number drop-down list.
When a call is initiated (or on
incoming calls), the call tab will be showed where you can see important
information about the call such as contact panel, caller ID, call status,
ring and call timer, disconnect reason and transfer status if any. Right
click on the contact or click on the Action button to do more actions when you
are in call, such as: send DTMF, hold the call, transfer the call, add more
participants (conference), add video, etc.
You can initiate instant message (chat) sessions in
multiple ways:
Ø Right click on a contact and choose “Send
Chat”
Ø Click on
the chat icon on the contact panels
Ø From the
Menu -> Action choose Send Instant Message
From the Chat tab you can select the desired number
(if the selected contact has more than one) and the preferred sipserver (if you
are connected to more than one sipserver). Than you can send plain or formatted
(html) message to your peer (depending on your configuration. Default is html).
You can insert emoticons and set text color and style.
When already in call, right click to the contact
and select “Add video”. There are lots of video codecs to choose from. Make
sure to set finetune your preferred video bitrate to achieve the best possible
quality. You can make even HD quality calls if you have enough hardware and bandwidth.
When you are in a voice or chat session you can
always add more participipants to the session.
The conference is handled by the softphone local mixer, so you can speak
with multiple persons even if your VOIP provider doesn’t support conference.
The standard SIP conference protocoll is also supported, so you can be invited
to such conferences too.
Contacts can be easily added, dropped, renamed,
imported or exported from Menu -> Contact. To see and edit a contact
profile, click on the “Details” link or double-click on the contact photo. When
a new contact is detected (by presence or automatic neighbour discovery) you
will get a confirmation form when you can accept or deny the presented people.
If somebody spams you, you can easily put it on the blacklist and you don’t
receive any more message from that person. To force a neighbourhood discovery, click on Menu ->
Contacts -> Find People near Me. If you have many contacts you can put them is
separate groups (Menu -> Contacts -> Create new Group)
Right click on a contact and
select “Send file”. You can send files only to other MizuPhone users, but the
protocol is compatible with every SIPserver (the sipserver will see an ordinary
call only with a special codec)
You can configure a shared
directory on your computer in Settings -> Misc. Your contact than can access
this file easily. To access other contacts shared files, just right-click on
the contact and click on Shared Files. If this option is not available, than
the contact doesn’t have shared files or doesn’t use MizuPhone.
For an easier collaboration you
can use Remote descktop. Right click on a contact, select “Remote Dektop” and
you will be logged in on your contact PC. You can work on their desktop remotely.
This feature is enabled only between MizuPhone users but it is compatible with
every sipserver (the sipserver will see an ordinary call only with a special
codec).
The fax functionality is still in
beta version.
You can receive fax images by
setting an account as a fax channel. Than the calls will be threated as faxes
(no local ringtone generated, the incoming fax messages are stored in the
predefined fax directory converted to pdf). To send fax messages, select Menu
-> Actions -> Send Fax, or alternatively,
you can mark some users as beeing fax devices (Is Fax option in the user
details form); in this case, when you right click on the users, the “Send Fax”
option will apear. Click on it, select the fax image (jpeg or tiff) and push
the “Send” button.
MizuPhone supports the voicemail
related SIP RFC’S, and additionally implements local voicemail functionality.
Voicemail can be configured at File Menu -> Settings -> Voicemail. Check
the “Show advanced settings” checkbox to see more options here. You will be
notified when your servers has new voicemail messages if the “Remote Voicemail”
option is set to “Auto” or “Yes”. For the local voicemail to work, you have to
set your greeting sound and the circusmstances when voicemail will be activated
(always, on busy, etc).
The call-forward functionality
can be set on the “Voicemail” section. Just specify phone number(s) when you
want your calls to be forwarded (always, on busy, etc). You can be notified by
missed calls/IM messages if you activate the “send missed calls/messages to my
email address” checkbox. In this case, you need to specify your email address
correctly for your profile, because notification messages will be sent to this
address (File Menu -> Edit your profile). Additionally you can accept
incoming calls automatically by filling the “Auto accept call from” editbox.
For a complete set of communication channels, there
are also a builtin quick link to your email client. Just right click on a
contact a press “Send Email”. Alternatively the embedded email client can be
used to send emails. You doesn’t need to have a vorking email client in order
for this to work. You can also send SMS messages to mobile phones. SMS messages
are sent like chat messages, but the number of characters is limited to 160 and
the format is always plain text.
You can explicitely set the exact
name/number and the used sipserver (if you have more than one) at every call.
However if you doesn’t specify these settings, the softphone will automatically
guess the best sipserver for your call based on your call history and the
called prefix. You can also type a complete URI (number@address:port) instead
of a name. In this case the call will be routed trough the server specified by
address:port. The address part can be a domain name or an IP address. You can
also control the dialed number by the Dial Plan. See the Configuration chapter for more details.
MizuPhone can send the initial
INVITE to multiple address where a contact can be. It always remembers the
contact’s last location, so even if your SIPserver is down, you can still call
your contacts. For proper RTP routing the sofpthone will automatically detect
your network, nat and firewall and will try to pass them correctly using
various technologies (firewall control, UPNP, STUN, ICE and other proprietary methods). When the remote
party RTP port is not obvious, it can start to send the RTP stream to multiple
locations, and will choose the one from where it receives answer.
Function |
Keyboard Shorcut |
Launch
application |
Ctrl+Alt+M |
Answer |
Enter |
Hang
up |
Esc |
Hold/Mute |
Spacebar |
Increase/Decrease
Volume |
Up/Down
Arrows (or +,-) |
If you use a NAT or firewall that cannot be handled
automatically by MizuPhone, you can explicitely set the listening SIP and RTP
ports on the sofphone, then configure your device (router, NAT or firewall) to
route the requested ports to your softphone. (setup a virtual server
configuration). You usually have to route the default SIP port (5060) and some
RTP ports (for example between 50000 and 50005). And, of course, you have to be
registered on your SIP server(s).
The best quality builtin codec is SpeexUWB (speex
ultra wideband), but in order to be used, your peer must also have this codec.
PSTN calls don’t have wideband codecs, because their characterisitcs. For this
codec to work properly, you will need a quality 45 kbits upload and download
(29 kbits for raw codec payload + rtp and udp payload + some signaling). To
achieve the best quality, you must use a headset. If you hear noise, enable
noise supression. If you hear cutting voice, check this FAQ.
Enable the silence detection
Increase the jitter buffer
Use a low bandwidth codec (for example iLBC)
Make sure that some other program don’t eat all of
your CPU time
Make sure you have at least 10 kbits/sec upload and
download.
Switch to the Dial page.
Enter the “phone number” in the following format:
username@domainname:port
for example:
(if you omit the port, the default port will be
5060)
Set Setting->Application->Trace Level to
“Details”
Eventually you can install a network sniffer (for
example Wireshark) and set to the port to local signaling port.
When you check the “Remember my password” checkbox,
your password will be saved.
To delete the saved password, just click on File
menu -> LogOff (for example when you leave a public internet location)
Go to File Menu -> Settings -> Codecs
(activate the “Show advanced settings” if you don’t see codecs).
Set the priority for your codec to the lovest
value. (0=highest priority, 100=lowest priority)
Eventually you can disable all other codecs
(uncheck the “Use this codec” checkbox; but in this case, calls may fail if the
remote endpoint don’t have the selected codec)
Copyright © 2008 MizuTech SRL