VOIP

Under development: PCMCIA, wireless, etc.
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Lobster
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VOIP

#1 Post by Lobster »

Image

Latest version
http://tmxxine.com/sip/psip-0.9.11.pet

--------- links - old info

Puppy Sip 0.9.3
http://tmxxine.com/sip/ps9.3.tar.gz

create the directory puppysip6 in /root
and run ps7

New for this version
New improved and smaller pjsua base from Benny
icons from Smokey



Now updated to include Smokey01 aka Grants
config file with extra SIP numbers . .
and CEL's enhancements.
program recompiled by HairyWill
quality enhanced and details retained
install and run from /root

http://www.opensourcesociety.org/2008/0 ... for-linux/
Get it from:
http://www.pjsip.org/

and the developer sfs to compile . . . (I used an earlier one but this is currently the latest)
ftp://ibiblio.org/pub/linux/distributio ... -4.1alpha2

"Pjsua" that seems to do it all...
http://www.pjsip.org/pjsip/docs/html/pa ... amples.htm

Pretty easy to use too...
http://www.pjsip.org/pjsua.htm#basic
Last edited by Lobster on Sat 12 Jul 2008, 11:27, edited 11 times in total.
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Re: VOIP

#2 Post by mcewanw »

Lobster wrote: To much for me to attempt
but if you are tempted all the info for compiling is there
The above refers to:

http://www.murga-linux.com/puppy/viewto ... 406#206406
mcewanw wrote:Here is something that looks very interesting to me. If this could be made to work, maybe someone could even write a simple gtkdialog or similar frontend to it?
...
Seems to be pretty easy to use too...
http://www.pjsip.org/pjsua.htm#basic
...
I think this is a job for muggins!
I still think muggins would like to try this one, or Newcrest maybe, they've both compiled and tested several VOIP apps before, and I think this commandline programmable one is going to be a winner. Will it connect successfully with Gizmo5? I imagine it will.

Alas, I have no time to do any more computing for one month, (or I'd have done tried it already) - since flying away in 2 weeks time and in process of getting organised.

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CEL
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#3 Post by CEL »

Here is the pjsua executable I've compiled for everyone's use. It's a nice 1.1mb when uncompressed. I've been able to make calls with it, but the sound is choppy. I think it might be just my machine, so let me know how it works for you.

I really like the idea of a gtkdialog frontend for this. We could have a Gizmo alternative in just a few megabytes! Very Puppy-like.

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#4 Post by Lobster »

Many thanks for compiling . . . :)
The raw material is now in place :)
I've been able to make calls with it
eh how? :?

The prog is running but from the documentation I could not work out much . . .
Do we need a server?
How does one obtain or use a Sip address?
Did you phone to a mobile phone?

How can we set up a one to one call?
Who is up for it?
What do I do? :roll:
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#5 Post by Lobster »

OK found out my sip phone number

sip:crusty_lobster@proxy01.sipphone.com

Jack Tronkel has phoned me - no sound yet (sounding like a robot apparently) - so connection . . . of sorts . . . 8)
Last edited by Lobster on Mon 23 Jun 2008, 10:23, edited 1 time in total.
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CEL
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#6 Post by CEL »

Lobster wrote:sip: 17473301020@prox
Yeah all Gizmo accounts have a SIP address like sip:1747#######@proxy01.sipphone.com
So to start pjsua with your gizmo account you should be able to run it like this:
./pjsua --id sip:17473301020@proxy01.sipphone.com --registrar sip:proxy01.sipphone.com --realm proxy01.sipphone.com --username 17473301020 --password yourpassword

When I use this with my gizmo account though, I'm having the same problem as you with the audio being sounding like a robot… For some reason it sounds okay with accounts on other SIP providers I've tried. (freecall.com and callwithus.com are good cheap ones) But there is still a constant jitter in all my calls, and my microphone isn't being recognized at all.

To make a call in pjsua, type m and press enter. Then type in the SIP address to call and press enter. The SIP address for a PSTN phone # looks like this: sip:1##########@proxy01.sipphone.com (Or if not using gizmo, it's at a different domain of course)

Also, you don't necessarily need a registrar server to use SIP. If you have a client like pjsua running, you can use your ip address as a SIP address (sip:###.###.###.###). But then you can't call out to the PSTN and you might have firewall issues or something.

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#7 Post by Lobster »

Image

Thanks CEL

Hacking the FreeBasic buttondialogue provided by MU
http://www.murga-linux.com/puppy/viewto ... 351#208351
I have created this rudimentary front end
which includes pjsua (the phone part)

http://tmxxine.com/sip/ps6.tar.gz

Now updated to include Smokey01 aka Grants
config file with extra Ip numbers . . .

If so this very simple prog can have those commands added
modification welcome . . .
Last edited by Lobster on Thu 26 Jun 2008, 04:09, edited 3 times in total.
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#8 Post by CEL »

Nice! But how can the program interface with pjsua to send commands once its running? Or would we just type them?

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#9 Post by Lobster »

:) You need variables and a programmer for that . . .

:)
The most I have been able to do is send voice one way
Really what I have created is pre alpha - more buttons can be added

Sip phone numbers can be found in Gizmo (under personal profile)
right clicking - allows you to find others profile and their Sip numbers.

Can we register sip numbers somewhere?
Are there interfaces for pjsua written in C?
Can we phone Sip phones?
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#10 Post by CEL »

Ah, I see. Well I'm looking at gtkdialog now so we'll see if I'm able to come up with anything.

pjsua itself is written in C, as are the other pj… libraries, and they're meant to be able to be embedded. The site describes pjsua as a "High level SIP UA library, combining SIP and media stack into high-level easy to use API."

When a sip client "registers," it means that it is sending its server its location (IP address) so the server knows where to find it when another client wants to call it. Or do you mean a different kind of registering?

To call a sip phone from the PSTN (Public switched telephone network, aka traditional phones), you need to have a "DID" number (Direct Inward Dialing). This is a phone number that you buy and then calls to it are gatewayed to your SIP address. There is one site I know that provides DIDs for free: ipkall.com. (They have an odd business model but it works!)

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pjsua

#11 Post by smokey01 »

Guys, I have some good news with the im part anyway.

The simple instructions:

Put the single file in /urs/bin/pjsua

from a terminal window type pjsua and you get the menu

What I did to advance past this stage was create a config file. You can name the config file anything you like. I actually created this file from scratch with geany.

It is possible to create the file from the menu with the dc (dump config) and f (save config) commands.

Here is an example what is in the config file:

#
# Logging options:
#
--log-level 5
--app-log-level 4

#
# Account 0:
#
--id sip:wombat01@proxy01.sipphone.com
--registrar sip:proxy01.sipphone.com
--reg-timeout 300
--realm *
--username wombat01
--password ***********

#
# Network settings:
#
--local-port 5060

#
# Media settings:
#
#using default --clock-rate 16000
#using default --quality 5
#using default --ec-tail 200
#using default --ilbc-mode 20
--rtp-port 4000

#
# User agent:
#
--max-calls 4

#
# Buddies:
--add-buddy=sip:crusty_lobster@proxy01.sipphone.com
--add-buddy=sip:puppyluvr@proxy01.sipphone.com
#

The im becomes easy to drive when using the config file.

To use the config file:

pjsua --config-file whatever.cfg

You will notice the config file above register me onto the gizmo server.

when I press the i command, it displays my buddy list:

See attachment.

Now you can select [1 - 2] press enter, then you can type your message and press enter to send it.

I tested it between two computer with different logins and it seemed to work fine. Both people have to be online for it to work. It seems to produce error messages if the receiving person is not online.

I'm still working on the voice quality. Currently it is machine gun like.

I did have limited success. On one call to lobster, the gizmo message said lobster was not available, played back perfectly. I have no idea what I did to make it work but I guess it does produce hope.

So far we have IM and Buddy Lists, next the world.

Smokey
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Aitch
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#12 Post by Aitch »

add me to the universal puppybuddy list, please

17473313699@proxy01.sipphone.com

Aitch

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#13 Post by Lobster »

Ah, I see. Well I'm looking at gtkdialog now so we'll see if I'm able to come up with anything.
I wanted to test out FreeBasic . . . :)

gtkdialog would be more efficient for small progs like this
- use the included graphic if useful

Everyone attempting a prog in gtkdialog3 . . .
Barry mentions on his blog for 23 Jun 2008
this directory in Dingo - slight differences in the 4.1 Alpha

Code: Select all

/usr/share/doc/gtkdialog3/examples
gives examples

but change
this bottom line on all code

Code: Select all

gtkdialog --program=MAIN_DIALOG
to

Code: Select all

gtkdialog3 --program=MAIN_DIALOG
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#14 Post by Lobster »

This part of the code that Grant (smokey01) mentions

Code: Select all

#
# Buddies:
--add-buddy=sip:crusty_lobster@proxy01.sipphone.com
--add-buddy=sip:puppyluvr@proxy01.sipphone.com
--add-buddy=sip:aitch@proxy01.sipphone.com

# 
can have as many sip numbers as possible during testing (available in Gizmo)

I have added Aitch in the above snippet of the config file :)
the numbers and names are equivalent - numbers is what you see when viewing profile . . .
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#15 Post by Lobster »

Aitch wrote:add me to the universal puppybuddy list, please

17473313699@proxy01.sipphone.com

Aitch
added here under Puppy buddies . . . more welcome
http://tmxxine.com/wik/wikka.php?wakka=PuppySip
Last edited by Lobster on Tue 01 Jul 2008, 06:23, edited 1 time in total.
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#16 Post by Lobster »

Update - this rudimentary program now available for testing
see first post in this thread for latest release

Now updated 25 June to merge with CEL's code and drop the FreeBasic Part
CEL's code implemented - not tested
Smokey01 aka Grants config file with extra SIP numbers . . .

you should change the cofig1.cfg
to match your details, like so:

Code: Select all

--id sip:crusty_lobster@proxy01.sipphone.com
--registrar sip:proxy01.sipphone.com
--reg-timeout 300
--realm *
--username crusty_lobster
--password ***********
Last edited by Lobster on Tue 01 Jul 2008, 06:24, edited 4 times in total.
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#17 Post by Caneri »

Ok...I got past the registering on PuppySip

my login is caneri@proxy01.sipphone.com

All I have heard so far is machine guns but maybe my local router is in the way as NAT is not working/configured I think...I guess...doh..hmmm

By the way @ Lobster...I love the front end gui picture :-)

Eric
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Chinese Proverb[/color]

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#18 Post by Lobster »

Glad you liked the Sip pic Eric,

:) had a go connecting with Tom and we also got the machine gun effect
not sure what parameters to change though - so advisement welcome . . .
It is getting the right commands here maybe . . .
http://www.pjsip.org/pjsua.htm
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#19 Post by smokey01 »

I spent another few hours trying to get the voice to work. I still have machine gun affect.

I have been scouring through the FAQ and help on the PJSIP site but no solution yet.

I wonder if the Gizmo server might be the problem. Maybe we could try another SIP server, anyone got any suggestions.

Has anyone tried an IP to IP connect yet?

I'm also going to post a message on the PJSIP site and try and find someone with experience. This may save us some time, although it is fun to play it's starting to get frustrating.

Smokey

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#20 Post by smokey01 »

Lobster, have you checked this out?

http://www.murga-linux.com/puppy/viewtopic.php?t=24342

Twinkle. Is quite small and open source.

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