Psip-1.4 Puppy Phone
Table of Contents
The Main Interface
The Right Panel
Logged out/Logged in Button
Testing & Setting up
Making calls to Landlines and Mobiles
Psip is a Voice Over Internet Protocol (VOIP) application that
uses the open SIP standard.
This means it is compatible with all applications that use the
same SIP protocol, and there are many.
Psip evolved from a command line version of Pjsua created by Benny
Prijono. In 2008 a small group of us
created Psip-0.12 which has been part of Puppy Linux for about
three years. It did have a few minor
issues and some users had difficulty getting it to work. Because
of this the the Psip project was reinvigorated.
Some work had been done on the original structure to fix broken
links and increase functionality. Psip-0.26 was
an improvement over 0.12 and consequently Psip-0.26 was released,
however some issues remained.
What was needed, was a skilled programmer to iron out the last
remaining problems. This is when James Budiono
AKA jamesbond joined the team. James has done a fantastic
job. He completely rewrote the Graphical User Interface,
combined all files into one and now Psip is a single
executable binary file. It does create a configuration file when
it's first closed.
The main interface is simple but effective. On the top panel there
are seven buttons.
Setup will open the preference dialog which has four tabs,
Account, Audio and Misc. Settings and Network.
If you don't have a sip account it is recommended that you create
one at http://serweb.iptel.org/user/reg/
Enter your details as above substituting your details for mine.
Make sure you include the prefix sip:
From version 1.4 you can setup multiple accounts however, you can
only be logged into one at a time.
As you can see I have three accounts. Select the one you wish to
log in to and click the Active check box.
When you set up an account at iptel.org you will then have the
ability to manage your account. Once you
log into http://www.iptel.org/toboggan/login
you will see the following page:
To setup a voicemail account click on the forward tab.
Note the right arrow, select both.
To set up a personal greeting call sip:firstname.lastname@example.org
You will also notice a radio button, Secure voice channel,
that when ticked will
provide secure communications.
This is voice only, not messaging and only from the person who has
it enabled therefore both parties need to
activate it to have two way voice security.
The Sip Proxy
allows you to enter the sip address for the proxy server if
required. It's not normally needed.
The audio tab will display available sound card devices. The left
and input side refers to the microphone
and the right to the speakers/headphones. It will normally select
the best choice but if not, you have
the ability to change it. Select by clicking on the radio button
then click on close. Your new settings
will be saved to the .psip.conf file in /root This will probably
look different to yours.
It is also recommended that you use dmix, if not automatically
selected. Dmix provides better sound card
sharing. What this really means is you could be having a
conversation and listening to a song at the same
time. I'm not really sure why you would want to do that but it
will improve sound handling. It also allows
you to play music or a prerecorded presentation to others in a
party conversation or conference room.
Sharing a presentation is done via the Call window using the Play
By ticking the Activity log button a file will be created in the
root directory called psip-activity.log.
This file will capture a lot of information that may be useful at
a later date. Even your text chat messages are captured.
The Console log level
determines the verbosity of logging. Zero being the least and
seven the most. Logging
is very useful for fault finding. You don't want to know how many
hours we have spend reading log files.
If you start Psip from a terminal window like ./psip32
you will see the data
scroll in the terminal window.
This information can be redirected to a file with this command ./psip32 > psip.log
Incoming call time out
determines the number of seconds to wait before the incoming call
is dropped. The default
is 15 seconds but this can be changed by entering a value in
seconds. Entering 30 would increase it to 30 seconds.
allows you to
select a personal ring tone. It needs to be a .WAV file and it can
be tested with the Test Button.
Command to execute on ring
allows you to use an external program to generate an incoming
Example: aplay -Dplug:dmix
It's probably best to use one or the other as both will play on an
It is also recommended to use dmix in audio settings for output as
Command on execute on IM
This allows you to set up an audio file to warn you when someone
has sent you
a text chat message. Once again it could be defined as above or
use an external command.
Command to execute on help
provides flexibility on how help is accessed.
To see this help when clicking on the help button on the Main
Interface type the following in the field:
Beep on incoming IM
ticked, will provide a short beep to your local speaker.
These days not all computers have this function.
Minimise to tray
the main window when the main window is iconified.
Auto-login on startup
will automatically register Psip with the SIP server on startup.
The Network tab provides a whole lot more functionality and
The Max calls
allows you to increase the amount of people in a party line call.
The default is 4
and this can be increased to 32. A party line call is when more
than two people can be involved
in a conversation. Example: Party A calls party B, then party C
calls either party A or B. All three
people can then hear and talk to each other. Very useful for
is the port used
for SIP communications. The default is 5060 and if the field is
port 5060 will be used. If for some obscure reason you need to use
a different port, you can define it here.
allows you to
disable TCP functionality. Psip uses UDP but could use TCP if it's
Disable optional SRTP
(Secure Real-Time Transport Protocol).
If disabled you will not be able to receive secure calls.
You may not be able to conduct a music or echo test at Iptel.org
The Public IP address
field is for true peer to peer communications. This is where you
unique public IP address. For this to work properly you will need
to setup your router with port
forwarding from your local IP to public IP address. Consult your
manual on how to do this.
The STUN server
where you can define a STUN server. When using iptel this is not
To learn more about STUN search for Trans NAT etc on the web. This
is a good place to start:
Activity Detection). The idea behind this is to reduce bandwidth
if you are not speaking.
ICE and TURN
are other SIP
server resources. Under normal circumstances you won't need to
This is as expected. When clicking on the quit button the program
will exit and close all associated windows. This
is an improvement over previous versions of Psip where some
process were not killed if closed with the little X in the
top right of the window. Using the quit button will save all data
to the .psip.conf
You have just added a couple of buddy's and you want to save them
at the end of the session, click on Quit. Clicking
on the X is the same as clicking on the Quit button which will
also save your changes and Quit the program.
Also note the ".
" in front
means the file is hidden. You will need to click on the eye
in Rox Filer to see this file. All files preceded by a dot are
The refresh button allows you to refresh the buddy/friends list.
Sometimes buddy's are shown offline when
in fact they may be online. To force the SIP server to update the
buddy list, simply click this button.
This button will open up the call window dialog which looks like
There is some very cool stuff in here and some of it is not
obvious. For example: right at the top of the
window there is a field with a Make Call button to the right of
it. This allows you to make an adhoc call.
In other words you do not have to add the address to the buddy
list. Lets assume you wanted to call Jack but he's
not in your buddy list. Simply type sip:email@example.com
in the field and click on Make
Call or press enter.
don't have to be registered to make a call to another person as
as the other person is logged in. The callee will only see your
IP address so they won't
know who is calling them. Expect to be declined as it's
You will however need to be registered to check your voice-mail.
The main white area of the dialog will display active calls, the
state/status and if the call is on hold or not.
It is possible to make more than one call at the same time and
place one on hold while talking to the other.
For example: You are speaking to a friend when someone else calls
you. You can answer the call and place the first call on
hold. You can even swap back and forth between them. You can also
have a three way conversation, this is known as a party call.
I told you this was cool. You can also have a real live three way
chat in a conference
room. You can have many people chatting in a
conference room. More about that later.
On the right you will see either three or four buttons depending
on the version you are using. Some countries do not
allow recording of phone calls so there is a version without the
recording function. As you can see, this version has
The top button Hangup
used to terminate a call. You first need to select the active
call in the left window then press the Hangup
button. If you make an illegal call, say to
yourself, then the call
will be terminated automatically.
allow you to record the conversation. It will ask you to select a
filename and path, press enter and the recording
will commence. Others in the conversation, in a conference call,
will be sent a series of beeps to advise the conversation is being
If you are in a party call or a two way call you will also receive
a message showing who is recording.
It would be polite and possibly a legal requirement to let others
know you are recording the conversation prior.
If you don't want to be recorded, you have the option to either
complain or hangup.
Once you have finished recording, you can listen to the recording
by pressing the Play
button. The recording is saved in /tmp
which will be deleted when you shutdown your computer, everything
in /temp is, so if you really want to keep the recording
you have better move it to some other location.
show you some interesting Statistics about your call such
as call quality, who the call is with, amount of data transferred,
dropped packets etc.
Under these buttons you will notice 12 small buttons. These are
required when checking your voice-mail. A recorded
message will say things like "press 1 to listen to the message, 2
to save it and 3 to delete it"
This window will display your incoming and outgoing calls. It's
very handy when you are away from the computer
for a while as you will be able to see who has called.
The help button is used to display this help.
Information on the software, Credits and Licensing
The Right Panel
The right hand side panel also has six buttons.
out/Logged in Button
This button uses the information from your account details in
.psip.conf and registers Psip with
the SIP server. Logging in provides functionality for you to see
other people online, retrieve
voice-mail. Maybe other stuff too.
The call button is used to make calls. You need to select someone
from the buddy/friends list then click
the call button. The call window will open and provide other
functionality already explained above.
If you don't select a buddy first, a box will popup so you can
type in a sip address.
This will open up a dialog window and allow you to text chat to
someone. You select someone
from your buddy's list just like making a call then click on
message. You type in the bottom window
and the responses will be delivered to the top window. Outgoing
responses are blue while return
responses are red, nice. The date and time is also displayed. You
can simply press enter to send
your message, you don't have to click on apply. If you want to
send more than one line of text at
a time press either CTRL+Enter or ALT+Enter to move down a line
without sending it.
Another improvement. The text no longer scrolls down the window so
you can't see it. It stays
in focus so you can see the latest incoming text without having to
There is indication too when when the other party is typing.
Another nice feature is the Save As button. This allows you to
save the entire conversation
of the session. This can be handy when you later need to refer to
some details. You can choose
where to save the file. It saves the file as Rich Text Format
(RTF). The reason for this is to
preserve the formatting and colours. Use AbiWord to read it.
This is how you add buddy's. When you click on the button a dialog
will appear like this:
You can type any name you like in the Nick name box and that's
what will be displayed in the buddy list
The SIP address must be typed in this format: sip:firstname.lastname@example.org
Make sure you include the sip:
at the beginning of your SIP address
. If you don't you
will notice the
absence of a ?
mark in the buddy list.
field is for
grouping your contacts. You may have one for family, another for
The categories can be expanded and collapsed in the main window.
The edit button allows you to edit your buddy's one at a time.
Simply click on the buddy you wish to
edit and click the edit button. So if you forget to add the sip:
to the beginning of the address
it's actually quite easy to fix it.
you to delete a buddy. Click on the buddy you wish to remove then
on the remove button. You will be given a warning like this before
the buddy is removed:
Click on Yes to remove the buddy.
Now make sure you press quit button to exit Psip so all of your
changes are saved.
I have created a couple of utilities that will be useful when
testing and using Psip.
They are called Simple Voice Recorder and Toggle Fire Wall.
SVR will help you to setup your sound. All it does is records you
voice and lets you play it back.
TFW is a simple tool to turn your firewall on and off. Hopefully
we will get around this last issue.
Simple Voice Recorder: http://www.smokey01.com/software/multimedia/svr-1.1.pet
Toggle Fire Wall http://www.smokey01.com/software/network/tfw-1.0.pet
Before you run Psip it is important to make sure that your sound
input and output is working.
If neither are working then Psip will not work. A good way to
achieve this is to download Simple Voice
Recorder above and install it. Clicking on the svr-1.1.pet file
will install it to Menu > Multimedia > Media Tools.
Now open up your sound mixer. In lucid this is Retrovol and mine
looks like this, your will probably be different:
Make sure your controls; Master, PCM and Front
are all turned up.
I have Mic and speaker connections at the front and back so it's
important to turn up the relevant ones.
This takes care of output. Now to get input to work you need to
make sure, as in my case, turn up Front Mic
If recording sound is a bit low then you can increase the Mic Boost
Make sure you have selected the correct Input So.
I have it
set to Front
In my case I can choose between Mic, Front Mic, Line & CD.
If you're not sure what you are doing just keep SRV and Retrovol
open and fiddle until it works.
If you really can't work it out, then post in the forum here:
To use Psip you don't actually have to subscribe to a SIP
service provider. There are two alternative
methods of communicating. One is via a service called no-ip and
their website is here: http://www.noip.com/
I'm not going to explain how no-ip works as all the information
is on their website. Basically it manages your
dynamic IP address. This means you can select a domain name and
it will be associated with your ever changing
external IP address. What this means in real life you will be
able to be located quite easliy by your friends if they
know your no-ip address which might look something like yourchosenname.no-ip.info.
The other method is what is know as peer to peer. This is not
unlike no-ip which really is peer to peer as well except
this time we use your external IP address. In some routers there
is an option called SIP ALG
this may need to be disabled.
In your Add Friends or Buddy List this is what you need to
enter for peer to peer.
For a no-ip entry, your buddy list should look something like
Making calls to
Landlines and Mobiles
It is possible to make calls to landlines and mobile phones but
you need to subscribe to a SIP provider
to achieve this. There are many to choose from but the one I'm
going to suggest here is Voipbuster.
Voipbuster has very good rates and provides a decent service.
Voipbuster will allow you to make calls
from Psip to landlines throughout Australia for free. The same
offer is available to many other countries
so please read their website here https://www.voipbuster.com.
Making calls to landlines and mobiles
normally incurs a cost so you have to buy credit from the SIP
provider. You will notice that their rates are
normally much lower than many Telco's.
Not all SIP providers are equal and their setups can differ
quite a bit. Below is the setup requirement
To add a friend to your buddies list, using Voipbuster, to dial a
landline, it should look like this:
and a Mobile entry should look like this:
Please take careful notice of the phone number format and no
spaces in the numbers.
The spaces in the number below is for display purposes only.
I'm sure I have missed a lot of important and useful information
but this is a good start.
I really do want to give credit to jamesbond
for his valuable
contribution, because without him, we
would be still fumbling around in the dark trying to find our
There are other Puppy community members that I would like to
mention for their testing,
interest, ideas, frustrations, time, commitment and dedication,
they are, in no particular order:
caneri, lobster, dogle, russoodle, CatDude, Stripe, Sylvander,