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 Forum index » Advanced Topics » Cutting edge
Puppy Phone - VOIP using SIP
Moderators: Flash, Ian, JohnMurga
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Lobster
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PostPosted: Thu 18 Aug 2011, 07:16    Post subject:  Puppy Phone - VOIP using SIP
Subject description: was Psip32 & Psip64
 



http://puppylinux.org/wikka/PuppyPhone
New interface and compile available
compiled 32bit and 64 bit NEW versions

Hi Guys,
I have not used PSIP for 3 years Embarassed
We need to get the existing PSIP working in Slacko, Lucid etc
and then move to new serverless SIP feature of PJSUA
and then release PSIP2/PSIPPY end of Sept with Puppy 5.3 Final 'Slacko'

Can we do it? We can certainly try Smile
Update: Reported working on Slacko,
Midnight Sun
http://www.murga-linux.com/puppy/viewtopic.php?p=538517#538517
and Puppy 4.13

Last edited by Lobster on Mon 03 Oct 2011, 04:50; edited 6 times in total
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Flash
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PostPosted: Thu 18 Aug 2011, 08:15    Post subject:  

Serverless SIP feature? Would that mean direct computer-to-computer connection over the internet?
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Lobster
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PostPosted: Thu 18 Aug 2011, 10:06    Post subject:  

It may be possible . . .
http://www.murga-linux.com/puppy/viewtopic.php?p=548336#548336

I have registered a new sip address with Ekiga
https://www.ekiga.net/

sip:crustylobster@ekiga.net

Now to get it running on something
Skype did not download from Iblio on Slacko . . .
Gonna try Android next,
still working on getting my mic working in Puppy . . .

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Caneri

Joined: 04 Sep 2007
Posts: 1580
Location: Canada

PostPosted: Thu 18 Aug 2011, 11:50    Post subject:  

Hi Ed,

Ok...I'll bite.

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?

Eric

PS
@kirk,
Can a version of Ekiga's sip software be compiled for FD64?

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ttuuxxx


Joined: 05 May 2007
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PostPosted: Thu 18 Aug 2011, 13:11    Post subject:  

let me know if you get it going without a server, I probably won't use it but I'll added it Smile
ttuuxxx

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Lobster
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PostPosted: Fri 19 Aug 2011, 03:00    Post subject:  

Thanks for the interest guys
This is what Smokey (Grant) wrote to the creator of pjsua
which is the 'engine' behind PSIP

Quote:
> What I am trying to do now is make it a peer to peer voip application which
> apparently it can do, according to your documentation. I have managed to get
> it to recognise various IP addresses on a local network but not the IP
> addresses on the internet.
>
> For example my home network has three of four computers connected at any one
> time with IP addresses like 192.168.0 3 and 192.168.0.4 etc. It works fine
> between each of these addresses.
>
> My real or external address on the internet is something like 118.210.*.*
>
> Is there some way to make pjsua see these external ip addresses if they are
> online like the local network?
>
> Thanks
>
> Grant


and this is the reply
Quote:
Hi Grant,

We do support various NAT traversal techniques, for example STUN, ICE,
and TURN. Have a look at these protocols (start with STUN) and then
how to use them with pjsua.

Best regards,
Benny


It sounds to me like 'geekity-geek-geek-geek'
If anyone with a bit of brain can help with translation or what we should be doing, that would be great (I only speak the lesser geek dialect)

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Lobster
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PostPosted: Fri 19 Aug 2011, 03:18    Post subject:  

My progress so far:

I have got a SIP client on my Android phone
http://code.google.com/p/csipsimple/
and installed Slacko on an Eeepc
- this will be my test rig - initially I will try phoning myself

The beauty of the Woof system is if we upgrade PSIP to PSIPPY
( m m m maybe we should keep the original name - depends how much change we incorporate)
- anyway any upgrade will be available on future woof builds such as Spup (slacko) Dpup, Wary, Drake and so on.

Our first priority is seeing what we have
and getting it working . . .

Ideally we would be able to communicate across platforms
- so for example to SIP phones, Ipad, other distros, Windows
and so on . . .

Grant seems very amenable to Puppys phoning him - as soon as I am confident enough, that my mic and config is right,
then we might update the list of SIP powered Puppys that we had.
Is Google Talk a SIP system?

We should also have manuals and considerable documentation
that I will be looking for next.
Any working links?
Maybe a Youtube or Wink tutorial?

Just getting up to speed here . . . Cool

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smokey01


Joined: 30 Dec 2006
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PostPosted: Fri 19 Aug 2011, 05:52    Post subject:  

I think the name Psip is still relevant.

Psip still works fine on Lucid. It just needs some tidying up.

It would be nice to get P2P working though.

It works fine within my network but I would like to be able to get it to work with NAT. Not having to use a sip server would be very cool especially if it could display who is online.

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Flash
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PostPosted: Fri 19 Aug 2011, 08:24    Post subject:  

Lobster wrote:

Quote:
Hi Grant,

We do support various NAT traversal techniques, for example STUN, ICE,
and TURN. Have a look at these protocols (start with STUN) and then
how to use them with pjsua.

Best regards,
Benny


It sounds to me like 'geekity-geek-geek-geek'
If anyone with a bit of brain can help with translation or what we should be doing, that would be great (I only speak the lesser geek dialect)

Here's what I came up with by Googling 'NAT traversal STUN'
http://en.wikipedia.org/wiki/STUN
http://www.voiptraversal.com/
http://searchunifiedcommunications.techtarget.com/answer/How-do-NAT-Traversal-and-STUN-work
http://calthrup.blogspot.com/2009/02/stun-nat-traversal.html
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smokey01


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PostPosted: Fri 19 Aug 2011, 08:32    Post subject:  

Excellent flash, but do you understand it, because I don't. I've done lots of reading but the penny won't drop.

I get the basic idea but how does one implement this in pjsua.

Have a look at the pjsua manual under the help menu in Psip.

Thanks

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Flash
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PostPosted: Fri 19 Aug 2011, 10:05    Post subject:  

Understand it? I haven't even read much of it yet. Laughing
This page seems very instructive. It may also explain why some people have had problems with the forum logging them out after just a few minutes.
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smokey01


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PostPosted: Fri 19 Aug 2011, 17:57    Post subject:  

Looks like a good article. I will have a good read of it later.

Thanks

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Lobster
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PostPosted: Sat 20 Aug 2011, 02:32    Post subject:  

Quote:
hi lobster nice to hear from you. Feel free to do whatever you want.


HairyWill aka Will has just confirmed that we can
change the code anyway we see fit.
He is busy being at leisure . . .

I am having difficulty getting my mic (always had problems)
working with precord-7.0.1.pet (as a test)
http://www.murga-linux.com/puppy/viewtopic.php?p=370446#370446

The code for PSIP is in
/usr/local/psip/psip

I won't being making any modifications until
mic working
So for now, if any easy changes to broken links in the GUI etc
just give us the code file and we will add to our PSIP directory Smile

We can stick with PSIP name as Smokey suggests
but use Psippy to refer to this initiative and any modifications/improvements/updates Cool

. . . now where is that USB headset . . . Rolling Eyes

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gcmartin


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PostPosted: Sat 20 Aug 2011, 13:57    Post subject:  

I have 2 questions which seem to be key for me
  • Why are we trying to change the name of PSIP. Why aren't we just calling it SIP? What up?

  • If you do NOT have a server, where is the registration going to come from for the caller to contact the callee?
SIP is a protocol that is kinda similar to Internet. It relies on some authority who know where things are so that connections can traverse for an end to end tunnel for 2-way audio-video traffic.

For example: has this been used on a local LAN to get one telephony to connect to another? For a client to client operation, this will help anyone understand what is happening at which point you will begin to see why there is an Asterisk (and others that exist).

I do not have the skills for programing a STUN or any other NAT traversal techniques for getting past your router into the iNet to support audio communication connection services. But, before I step that far, I would most certainly insure that I can pick up a telephone on my LAN and talk to some other LAN user to insure that much of the system you envision is operating with the clarity you want.

Trying to help with this posting by providing a starting point for a roadmap.to successful implementation.

If need, and there is some document which points me to doing this on the local LAN using the product your reference, please steer me so that I can get the 1st step for direct SIP to SIP communication going.

BTW aren't there some SIP to SIP or SIP to POTS implementations already available off-the-shelf?

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gcmartin


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PostPosted: Sat 20 Aug 2011, 14:52    Post subject: SIP registration and use on Puppy desktops.  

A starting point for this community is to have some one of us (@Lobster or @Smokey01 or etc) to add Puppy instructions here for the Ekiga client's use.
If there's some reason why we should NOT offer an update to that webpage, please share in this thread.

Also, please share why we are planning to go around using Ekiga after they have agreed to provide a free opportunity to users (meaning us)?

Thanks in advance

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