PSIP - a simple GTK GUI for pjsip
PSIP is a software phone using SIP
one of many
It enables one to make phone calls to other phones using the same SIP protocol - either hardware and hardware.
- Audio calls
- Text messaging
- Registrar-based & registrar-less (aka serverless) operations
- Built-in conference bridge with independent volume and hold control
- Playing and recording (as allowed by law - if it's illegal in your country to record messages, you can build PSIP
with recording function disabled).
- Secure voice channel (SRTP)
- Maximum 256 friends.
- Maximum 32 simultaneous calls
Quick Start Guide
How to use
- Get a SIP account.
- Open preferences window by clicking the "Setup" button.
- Enter your account details, prefix your SIP account and Registrar's URL with sip:.
Enter * as your Auth Realm, and your username and password as directed by your SIP Registrar.
- Click "Register" button.
- Start adding friends and making calls. Remember, every sip address you enter must be prefixed with sip:
For more information, please refer to Smokey01's PSIP Help
The application is extensively decorated with tooltips.
Place the cursor on the field / label, and the tooltip will show you what it is for.
FAQ (Frequently Asked Questions)
What is a SIP account?
A SIP account is just like a phone number - it gives one an unique identity across the
vast seam of networks.
A SIP account can be obtained from a SIP Registrar - a central server that maintains
mapping between a SIP account with the actual, physical device of the account owner.
Do I need a SIP account?
Only if you plan to make/receive Internet calls, and only if your PC/device is behind a NAT router.
If your device is not behind a NAT router (i.e. you are directly connected to the Internet, if
your IP address on your device is the public Internet IP addres) - then no, you don't need a SIP account.
Your Internet IP address *is* your SIP account.
Note: Most of computers at home connected via ADSL and/or Cable are behind NAT routers.
Where can I get a SIP account?
You can get a SIP account from anyone who acts as a SIP Registrar. There are many of them -
all you need is just to google for them
Two that I personally use and can thus recommend are:
How to test my SIP connection is working?
Most SIP Registrars have a special account for testing.
Help, I can't connect!
- Registration failed. Reason: bad user id, bad password, server down.
- Can't make outgoing call. Reason: Make sure your called party is "registered".
- Can't receive incoming call. Reason: Make sure you are "registered".
- Outgoing calls work when I'm not registered. It stops working once I'm registered. Reason: you have a network problem. See below.
Help, there is no sound!
First, make sure your firewall is turned off (open terminal, then type /etc/rc.d/rc.firewall stop
The test you can do:
- Get a compatible WAV file (e.g. from here).
Open the "Call Window" and click the "Play" button to play the above file. If you can't hear anything, it's sound problem,
so do these:
- Play with your the "audio settings" section of the "Preferences" menu.
- Close all other applications that may be using sound (e.g. youtube) - they may be using the sound device exclusively.
- If sound works as above, it's not sound problem. Most likely it's network problem. And most likely it's your firewall.
Try turning it off and see if PSIP works after that.
(Don't worry, PSIP can run with firewall "on", but you must learn how to allow traffic on UDP port 5060 to go through).
- For other problems, please read more here.
OK sound works if I turn off the firewall but I want to use PSIP with my firewall on!
Well you can do that as long as you open the SIP ports and the media (RTP) ports.
The SIP port is usually at 5060 and 5061 (if you don't change the defaults), the media ports usually
start at 4000 and counting upwards twice the number of maximum simultaneous calls.
If you use default settings, maximum number of simultaneous calls are 4 so there will be 8 (eight) media
ports, from 4000 to 4007.
Which public STUN server I can use?
Don't use STUN server unless you are having connection problems.
That being said, there are some public STUN
servers you can use:
There may be others, those are the ones I know.
My connection works if I don't use STUN, but it doesn't work if I use STUN!
See the above comment. That being said, STUN resolution can also fail if you have specified "maximum number
of simultaneous calls" which is too high - the failure happens because your firewall / NAT router cannot
open that many ports for you. The default number of max calls is 4, which is usually fine.
Which public TURN server I can use?
Don't use TURN server unless you're having connection problems.
That being said, there is one public
TURN server that I know - numb.viagenie.ca. But this server doesn't seem to work with pjsip.
Note: You need to enable ICE to use TURN server.
I'm good, now tell me about this peer-to-peer SIP!
You mean registrar-less operation. SIP is peer-to-peer for the most part.
If you use free (as in free beer) registrars, it's most likely that your voice communication is already
transported peer-to-peer - the Registrars just act as an exchange to connect the parties
together, and once they are connected, the voices are transported directly peer-to-peer and not through them
(they wouldn't be able to afford the cost of the bandwidth otherwise).
If you want to run a registrar-less operation - make use of the Public IP address, and ensure that you have
done port-forwarding your NAT router to your PSIP PC. I will write about it if I have more time, meanwhile, Google is your friend.
How do I get PSIP to work with multiple accounts from multiple Registars?
You can't, PSIP does not support mutilple accounts at this stage.
A possible workaround is this: the account information is stored in $HOME/.psip.conf - so you can have
multiple config files, and use shell script to switch between them.
PSIP wouldn't exist without these fine software:
- Grant "smokey01" (website)
- Eric "Caneri" (website)
- Ed "lobster" (website)
- Suz "russoodle"
- James "Dogle"
- Don "Stripe"
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