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 Forum index » Advanced Topics » Cutting edge
Puppy Phone - VOIP using SIP
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gcmartin

Joined: 14 Oct 2005
Posts: 4078
Location: Earth

PostPosted: Mon 22 Aug 2011, 19:17    Post_subject:  

I want to share that the items I share in the thread on PSIP is not to be viewed as criticism.

It is NOT!. I am in favor of this effort and I am in favor of a Peer-to-Peer telephony (SIP) subsystem in Puppy.

I am sharing, thru questions, (tough ones in some cases) about the issues we are up against as we progress.

I have and do offer again to write a document with instructions for how to use standard SIP compliant devices and clients in a Puppy Peer-to-Peer environment.

We just need to come up with an implementation which matches some simple structure.

I am not a coder, therefore my skills that I offer, here, is in testing and documenting what's observed and what's useful.

So, if anything I have shared seems negative, or discouraging, please don't take it that way. I only mean to share items I think we may have to address as we progress in the peer-to-peer discussion.

All other SIP implementations, either via a soft/hard VOIP PBX or an external VOIP provider for SIP registration and IP call connections are already done for us. Even SIP gateway servicing for ringing real telephones in the world is done for us. Much is free, but also, many requires some type of implementations to purchase SIP devices, and adapters to interface with real telephony calling devices that users are accustomed to around the house or office..

In summary, I have been trying to make clear that there appears to be 2 issues.
One - which is already solved for us via SIP VOIP Registrars and Gateways
Two - Peer-to-peer which they don't make it easy for us to implement, directly, one user to another.

Please understand that by sharing this, I am trying to help.

Hope this helps.

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Flash
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PostPosted: Mon 22 Aug 2011, 21:44    Post_subject:  

The second issue is the only one I'm interested in solving. Smile
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Lobster
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Joined: 04 May 2005
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PostPosted: Tue 23 Aug 2011, 04:13    Post_subject:  

have registered at
http://www.iptel.org/service
as smokey recommends - this will be my default
sip:lobster@iptel.org

this is my backup sip address
sip:crustylobster@ekiga.net

Thanks to smokey for new help files
simplified menu etc

Will be ensuring mic and sip account works
You will probably be able to phone me in a day or two

Ok...I'll bite.

Quote:
sip:caneri@ekiga.net

I'll prolly regret posting this in public but what the 'ell.

I'm on FatDog64....I may need to load another iso/version....any ideas?


Hi Eric phoned you but you were probably busy making pizza (not online)
Will try again . . .
You will be safe - have placed my SIP and yours here (we can always remove - don't expect much traffic)
http://puppylinux.org/wikka/Psippy

I don't see any reason why PSIP should not work on any Puppy that it is compiled and installed on - that is the aim for the updates if it is not working Smile

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Lobster
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PostPosted: Tue 23 Aug 2011, 05:34    Post_subject:  

Quote:
I have attached a single file called psip_gui which need to be copied to /usr/local/psip. I have zipped it up to make it upload compliant.


Nice job Grant Cool
the help menu is now useful
the credits will need updating
it uses javascript (as I am running noscript I got no credits)
- anyway that is for later

Ran the tests
- tests working
- got music
- next trying to send voice mail

Might be worth getting hold of
Evil20071@proxy01.sipphone.com
as he helped either in testing or coding the original version to some degree.
He might even have renounced the way of the Sith . . . Wink

The peer2peer connect is a great possibility
- something to aim for Cool

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Caneri

Joined: 04 Sep 2007
Posts: 1580
Location: Canada

PostPosted: Tue 23 Aug 2011, 09:10    Post_subject:  

Hooray!!

I had a mishap with the psip.config but now it works with iptel.org on an old 409 install.

I need a newer iso as Fatdog beta5 doesn't work with psip so far.

caneri@iptel.org

Smokey..what's your number? (EDIT: found it and added to buddy list)

EDIT: I can't send voicemail to an inbox...howto?
EDIT!: hooray!!! sent voice message to smokey01...now to try and hit on lobster...bwahahahaah.
EDIT:2 this is a handy page http://www.iptel.org/service

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Lobster
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PostPosted: Wed 24 Aug 2011, 05:30    Post_subject:  

I added the following SIP addresses
but they went (disappeared, AWOL, gone, vanished) and no refreshing of the refresh buddies list made any difference . . . Confused

Grant
sip:smokey01@ekiga.net
Lobster
sip:crustylobster@ekiga.net
sip:lobster@iptel.org
Eric (caneri forum name) running Fatdog64
sip:caneri@ekiga.net

This file or whatever is the config file is not activating an editor or
some way of manually altering settings (I am using Slackobeta1)
/usr/local/psip/pjsua.cfg.default

I stared at the code for a while, like a slice of lemon watching fish
but nothing seemed obvious . . .

We need more coding help . . . Cool

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smokey01


Joined: 30 Dec 2006
Posts: 1795
Location: South Australia

PostPosted: Wed 24 Aug 2011, 06:18    Post_subject:  

lobster, you will have an error in the config file.

all commands must start with --

If you make an error or a typo your buddy list will disappear and psip will not work properly.

Have a look at my help file from within Psip.

The config file is located in /root/.psip/pjsua.cfg

If you send me your config file I will take a look, correct it and send it back.

BTW you need to activate your voice mail IPTel.org then I will be able to leave you messages.

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Caneri

Joined: 04 Sep 2007
Posts: 1580
Location: Canada

PostPosted: Wed 24 Aug 2011, 09:23    Post_subject:  

Here is a working psip.config

Code:
#
# Network settings:
#
--local-port 5060
#I think ice is important for maintaining routing information for those with dynamic internet ip addresses
--use-ice
#I think this might only work if you have a gizmo account
#--stun-srv=stun01.sipphone.com

#
# Media settings:
#
# using default --clock-rate 12000
--quality 10
# using default --ec-tail 200
# using default --ilbc-mode 20
--rtp-port 4000

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

#
# Buddies:
--add-buddy sip:crustylobster@ekiga.net
--add-buddy sip:520@ekiga.net
--add-buddy sip:lobster@iptel.org
--add-buddy sip:smokey01@iptel.org
--add-buddy sip:1001@iptel.org

#












# Account 0:
--id sip:usrid@iptel.org
--registrar sip:iptel.org
--realm *
--username (your userid from whatever sip provider you sign up with)
--password xxxxxxxxxxxx
--reg-timeout 55


Always add one blank line at the end of the psip.cfg.
The cfg will not work without the extra blank line at the end.

I use call quality 10...this seems to improve the playback.

@smokey...got the voice messages and now have my usb headst working...should be better sound quality.

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gcmartin

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PostPosted: Wed 24 Aug 2011, 11:07    Post_subject: OTB Configuration Generator  

Would it be helpful if we had an OTB Configuration Generator for PSIP? A tool like this would also be a syntax checker as well.

If so, who could design such?
Should it be integrated into PSIP OR should it be a companion to PSIP?

I think this would make it easier for us (and newbies) to use without having to appeal for assistance.

Hope this helps

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smokey01


Joined: 30 Dec 2006
Posts: 1795
Location: South Australia

PostPosted: Wed 24 Aug 2011, 16:29    Post_subject: Re: OTB Configuration Generator  

gcmartin wrote:
Would it be helpful if we had an OTB Configuration Generator for PSIP? A tool like this would also be a syntax checker as well.

If so, who could design such?
Should it be integrated into PSIP OR should it be a companion to PSIP?

I think this would make it easier for us (and newbies) to use without having to appeal for assistance.

Hope this helps


Psip already has one, of sorts. The first time Psip is run it does not have a pjsua.cfg file in /root/.psip. It copies a default one from /usr/local/psip which is called pjsua.cfg.default and renames it to pjsua.cfg.

It's a little hard to automate it more it than this as you need to add your details, servers and buddy's.

This can all be done manually with a text editor but it can also be done under menu item configure.

The error checking is another issue but I'm not sure who has the skills to address this.

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Lobster
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PostPosted: Thu 25 Aug 2011, 03:50    Post_subject:  

Quote:
BTW you need to activate your voice mail IPTel.org then I will be able to leave you messages.


I believe I have now done that - voice messages welcome Smile

I want to go through some of the history of the PSIP project
that I hope is relevant. We did look at a great variety of software from Skype (Now a Balmer detergent software product) to Gizmo (bought and abandoned) and many others such as 'teamtalk' . . .
Most were too big, hogged bandwidth etc

I wrote the first prototype front ends using the current command line pjsau, to get the whole system working. HairyWills efforts later on very much surpassed my programming aptitude and I was no longer able to comprehend the code . . . Smokey and Evil20071 were also coding and working on the look and feel (from what I remember). Others such as Eric were kindly testing . . .

Part of the problem for Will was making a variable (config file) available to the gtkdialog3 front end and Will had to do all sort of programming gymnastics to get this working . . .

A lot of initial work was done with the configuration file
(a text file) where we edited in a text editor (as Smokey has kindly offered to do and I may have to take him up on).

Later on this configuration file became dynamic and automated.

So my efforts are now to try and manually get the config text files
working as suggested by Eric and Grant (Smokey)

Thanks guys for your patience.
I will hopefully leave voice messages soon. Smile

Puppy Linux
Speak Boy!

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gcmartin

Joined: 14 Oct 2005
Posts: 4078
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PostPosted: Thu 25 Aug 2011, 13:01    Post_subject:  

Lobster wrote:
... I have got a SIP client on my Android phone ...
Did anyone ever get PSIP to Android going over your WiFi (P2P)?

And on this site would you post the location for the current PSIPs (both 32bit and 64bit)

Thanks in advance

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Lobster
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PostPosted: Thu 25 Aug 2011, 23:02    Post_subject:  

Quote:
It's a little hard to automate it more it than this as you need to add your details, servers and buddy's.

This can all be done manually with a text editor but it can also be done under menu item configure.


Thanks guys - getting closer . . . to a working system . . .

Thanks Grant for the modified config file Smile
The buddies are not appearing on the right where they should
. . . the 'item configure' is not working in slacko
(as far as I can see)

My mic is now working in Slacko I did a pawedcast
(in the announcements section)
- sorry did mean to mention PSIP but forgot Rolling Eyes

Has anyone got VOIP working in Lucid as expected?
Do I need to turn off my firewall?
(tried turning off firewall, as we did this for a while during development)

Here is how to edit the wiki for those wishing to help Cool
http://puppylinux.org/wikka/UsingThisWiki

May try SIP from Android but that is secondary
to getting Puppy version working. Smile

I must admit that these troubles are what I had
3 years ago
It was working fine during development
and then the 'improvements' meant it was no longer working (for me)
during the last two weeks (aprox) of development
That is the situation I still find myself in . . .

I have no idea what I am doing wrong . . .
Smokey has kindly edited my config file
- and given some useful pointers

when putting in details:
sip URL: lobster@iptel.org (your iptel registered name replaces 'lobster')
registrar URL: sip.iptel.org

It has to be easier than this . . . Rolling Eyes

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smokey01


Joined: 30 Dec 2006
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PostPosted: Fri 26 Aug 2011, 05:47    Post_subject:  

Lobster wrote:
Quote:
It's a little hard to automate it more it than this as you need to add your details, servers and buddy's.

This can all be done manually with a text editor but it can also be done under menu item configure.


Thanks guys - getting closer . . . to a working system . . .

Thanks Grant for the modified config file Smile
The buddies are not appearing on the right where they should
. . . the 'item configure' is not working in slacko
(as far as I can see)

My mic is now working in Slacko I did a pawedcast
(in the announcements section)
- sorry did mean to mention PSIP but forgot Rolling Eyes

Has anyone got VOIP working in Lucid as expected?
Do I need to turn off my firewall?
(tried turning off firewall, as we did this for a while during development)

Here is how to edit the wiki for those wishing to help Cool
http://puppylinux.org/wikka/UsingThisWiki

May try SIP from Android but that is secondary
to getting Puppy version working. Smile

I must admit that these troubles are what I had
3 years ago
It was working fine during development
and then the 'improvements' meant it was no longer working (for me)
during the last two weeks (aprox) of development
That is the situation I still find myself in . . .

I have no idea what I am doing wrong . . .
Smokey has kindly edited my config file
- and given some useful pointers

when putting in details:
sip URL: lobster@iptel.org (your iptel registered name replaces 'lobster')
registrar URL: sip.iptel.org

It has to be easier than this . . . Rolling Eyes


Lobster you still haven't got it quite right, notice the "=" signs.

--id sip:lobster@iptel.org
--registrar=sip:iptel.org
--realm *
--username=lobster
--password=xxxxxxxxxx
--reg-timeout 55

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smokey01


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Location: South Australia

PostPosted: Fri 26 Aug 2011, 06:16    Post_subject:  

I think I mentioned iVisit a few years back.

It truly is a an amazing bit of software for it size, unfortunately there is not a Linux version, just Windows and Mac.

It does however run quite well in Wine except for two very important features, Sound and Video.

Why is it so amazing, take a look. It will fit on a floppy disk in it's compressed EXE file and it does audio and video.

It's all propriety though just like Skype but heaps smaller.

http://www.ivisit.com/products

http://www.ivisit.com/classic The small classic version.

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