Puppy Phone - VOIP using SIP

Under development: PCMCIA, wireless, etc.
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smokey01
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#16 Post by smokey01 »

gcmartin wrote: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?

    It seems the name will remain as Psip.
  • If you do NOT have a server, where is the registration going to come from for the caller to contact the callee?
I suggested we remain with Psip, it appears lobster has agreed.
gcmartin wrote: 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).
Apparently pjsua, the engine of Psip, is able to conduct peer to peer communications without a sip server. There are a number of people who would like this functionality including myself. I'm just having a few problems working out how to configure it. I understand if I was to put your IP address into my buddy's list I could see when you were online and I could call you. Of course this is more useful if you have a static IP address rather than dynamic one.
gcmartin wrote: 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.
Neither do I yet but I'm working on it.

With the current Psip I am able to communicate between my current network on 192.168.0.* without a sip server.

For example: on computer A, with an IP of 192.168.0.3, I can communicate with computer B, with an IP address of 192.168.0.4. This works very well. What I am trying to do is communicate with someone outside of my local network in the same manner.
gcmartin wrote:
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.
The pjsua manual can be read from the existing Psip under the help menu item or here http://www.pjsip.org/pjsua.htm
gcmartin wrote: BTW aren't there some SIP to SIP or SIP to POTS implementations already available off-the-shelf?
Yes but they are all quite large. Psip is way under a Meg which fits better with the Puppy vision.

Psip sound quality is also brilliant. It's probably better than all of the rest I have tried.

It can also be used on dialup, try that with Skype.

Register an ekiga account here. https://www.ekiga.net/?page=register

Here is the list of changes that I am currently working on:
  • Tidy up the menu system
    Remove and update redundant links
    Include a username along with the sip address in the buddy list
    Include the text-chat in main dialog instead of floating windows
    Peer to Peer communications
I don't know what lobster has in mind as we have not spoken at length about it yet.

If I was really skillful I would like build the entire GUI into the C code of pjsua.

It would be nice just to have two files, the main and the config file.

I hope that answers your questions.

Regards smokey

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#17 Post by smokey01 »


gcmartin

#18 Post by gcmartin »

NAT traversal is done in your router. Contact your router vendor if you are unsure about how.

Hope this helps

gcmartin

#19 Post by gcmartin »

Could someone share a location of the steps to connect one user with another. Is this just audio or is it video as well.?
Can a single SIP user connect to a SIP telephone on the LAN?
Can a SIP user connect to a SIP device which can ring telephones in the home?
Can the SIP user connect to another SIP user via VPN?

The power of SIP is not how small it is...Its its functionality for audio communications to any/all SIP compliant devices. This is what attracts people to SIP clients....functionality! functionality!! functionality!!! and of course clarity.

One of is problems is that the specs has been TOO loose such that vendors would NOT create proprietary islands and platforms. That has been one of the major issues to date. The patents has stifled a lot of what was planned.

Thanks in advance
Last edited by gcmartin on Sun 21 Aug 2011, 17:50, edited 1 time in total.

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Flash
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#20 Post by Flash »

I don't understand why a router or NAT server doesn't seem to have the problems when I connect PSIP or another VOIP client to a server that it apparently has if I try to connect PSIP directly to another computer (with the NAT server or router between them.) If having a server between the two computers will make the NAT server or router happy, why not configure one of the servers that comes with Puppy to do the job, and then connect PSIP or whatever through it? I'm asking this in order to demonstrate that I am completely ignorant of anything to do with servers or VOIP. How hard can it be? :lol:

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#21 Post by smokey01 »

gcmartin wrote:Could someone share a location of the steps to connect one user with another. Is this just audio or is it video as well.? Just audio
Can a single SIP user connect to a SIP telephone on the LAN? Yes
Can a SIP user connect to a SIP device which can ring telephones in the home? Yes but you need to connect through a sip server that has a gateway to the PSTN telephone network. This will cost a small fee.
Can the SIP user connect to another SIP user via VPN? Not sure

Thanks in advance

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#22 Post by smokey01 »

Flash wrote:I don't understand why a router or NAT server doesn't seem to have the problems when I connect PSIP or another VOIP client to a server that it apparently has if I try to connect PSIP directly to another computer (with the NAT server or router between them.) If having a server between the two computers will make the NAT server or router happy, why not configure one of the servers that comes with Puppy to do the job, and then connect PSIP or whatever through it? I'm asking this in order to demonstrate that I am completely ignorant of anything to do with servers or VOIP. How hard can it be? :lol:
Flash if you go to this link https://www.ekiga.net/?page=register and register a sip account, say flash@ekiga.net and set up the account in Psip I will be able to call you. The problem with Psip at the moment it is not indicating when users are logged in. This worked perfectly fine with sipphone, another sip server. I'm not sure why this doesn't work at the ekiga sip server.

It is easy enough to connect through the sip server but it would be nice to connect without having to use a server, less dependencies.

Psip works perfectly fine as it is at the moment. We don't need to be connect to the same sip server either.

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#23 Post by smokey01 »

As I said earlier a number of the links are broken.

The help manual I created also seems to have disappeared.

I will attach a PDF version of help that should assist.

This is the link to the original thread when we first developed Psip.
http://www.murga-linux.com/puppy/viewto ... 572#212572
Attachments
Psip-Help.tar.gz
Psip Help Manual
(137.95 KiB) Downloaded 977 times

gcmartin

#24 Post by gcmartin »

Here's some helpful pics which might help one visualize SIP

Little while ago, IBM made this to easily understand. Modeling SIP via architecture.

The Architecture includes a Gateway. Your non-server product must build all of that functionality into your product versus using a server to connect.

I think what you are trying to do is to have the freedom of NOT using someone else's server for SIP registration and SIP G/W services. Is that right?
But, this is going to require some very creative thought, designing a model, and building the model+client to match your objective.

Other options are, maybe, just build your own personal local PUPPY SIP Server_Gateway, or use OpenSER or Asterisk to do all the dirty work, easily. Locally,.Asterisk will run on a PUP. Others SIP Gateway_servers can be deployed in Puppy, too.

Or, here's a thought. Suppose the Puppy Community has set up its own SIP Server_Gateway somewhere on the internet; say "http://puppy-linux.org/SIPserver" to provide call connection services needs. And suppose, that each PUP which run a Puppy SIP client registers there. And, suppose, that each PUP SIP which registers uses some "new" Puppy client so that Server_Gateway processing was shared, this would created the world's largest telephony switch. It would become enormous for Puppy and would draw every person wanting to have voice-video connections into the Puppy Community environment ...directly or indirectly. Imagine how beneficial this could be.

Hope this helps
Last edited by gcmartin on Mon 22 Aug 2011, 06:13, edited 1 time in total.

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#25 Post by Flash »

gcmartin wrote:... just build your own personal local PUPPY SIP Server_Gateway, or use OpenSER or Asterisk to do all the dirty work, easily. Locally,.Asterisk will run on a PUP. Others SIP Gateway_servers can be deployed in Puppy, too.
In my ignorant way, I was attempting to suggest something like that, only the server would run on the same machine as the VOIP client. Perhaps the Hiawatha server which comes with most Puppys could be configured to satisfy whatever the router or NAT server requires, then a SIP client would run as a separate process (if that's the right terminology) that uses the Hiawatha server. Is it even possible to run a server and a client on the same machine? Seems like it ought to be.
Or, here's a thought. Suppose the Puppy Community has set up its own SIP Server_Gateway. And suppose, that each PUP which run a Puppy SIP client registers there. And, suppose, that each PUP SIP which registers uses some "new" Puppy client so that Server_Gateway processing was shared, this would created the world's largest telephony switch. It would become enormous for Puppy and would draw every person wanting to have voice-video connections into the Puppy Community environment ...directly or indirectly. Imagine how beneficial this could be.

Hope this helps
I can see millions of people demanding help from the forum to get their VOIP working. Someone would have to come up with a really good GUI or wizard for configuring VOIP in Puppy. :lol:

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#26 Post by smokey01 »

I have spend the best part of the day updating Psip.

I have also done quite a bit of research to find a good SIP server that gives away free SIP accounts. My final decision was http://www.iptel.org

Psip seems to work very well with this server.

I have also updated the Psip help from the help menu. Be sure to have a read it has some useful information in it.

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.

WARNING do back up your original psip_gui first. Just because it works here doesn't mean it's guaranteed to work for you.
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#27 Post by Caneri »

Hi Grant,
I used your new psip-gui and all seems ok.

Psip doesn't seem to show anyone online but it may be that there is nobody online...dunno.

The old sipphone.com doesn't seem to work any longer.

Best,
Eric

EDIT: what is the latest version of psip?
I have PSIP 0.20
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#28 Post by gcmartin »

Ah yes, IPTEL.
Edited: Note: All IP adapters or telephones must support STUN. (2009)

Did you notice their mention of the SIP Router Project? (latest project where the forks have agreed to combine their efforts)

If would be nice if they made it very, very easy to build a local SIP server for "calling" around on your LAN. To do it ourselves, we need
  1. dedicated PC with the following
    • 256-1GB RAM
    • 2 LAN NICs
    • adapter in the PC to route calls to Plain old telephones in your homes
  2. OS (distro)
    • Small PUP with DNSMASQ instead of DHCPD
    • SIP Router Project for local call management and connection services for SIP registration for calls outside of the home and to manage SIP calls inside the home
  3. Pictorial Instruction Manual
The above is a simplistic view of a system for running in the home or small location allow the use of internal home (line1+line2 = rj11 wiring OR line1-4 = rj45 wiring) and also allows for use of LAN connected IP phones or SIP PC clients (both wired/WiFi)

There are several "free" SIP-telephony servers available for doing this, OTB.

Biggest problem with this is that MOST newbies would NOT dare attempt this because the management issues are just beyond their reach and much too complex.

If this thread/community should undertake this effort, we must make it as easy and attractive as jumping into a "pool of naked beautiful people".

Hope this understanding helps

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#29 Post by Flash »

gcmartin wrote:...It would be nice if they made it very, very easy to build a local SIP server for "calling" around on your LAN...
Perhaps I misunderstand, but Smokey01 seems to be saying he was able to connect computers directly, within his LAN, here, without needing a server. It seems that you're screwed if you try to connect two computers directly, with a router between them. Even so-called P2P applications seem to need the initial connection to be established by a server. Is that correct?

gcmartin

#30 Post by gcmartin »

Flash wrote: ... Perhaps I misunderstand, but Smokey01 seems to be saying he was able to connect computers directly, within his LAN, here, without needing a server. ...
Yes @Flash, you are correct, and I saw that too.

But, lets say you have several PC users on your LAN whose PC names are Jack, Dick, Jane and Jill. If you want to call either one of them from your PC called Flash, how do you do that?
  • Can I use their PC name
  • Can I use their SIP information
  • How would I use an IP address
  • What will happen on my PC should they happen to call me, Flash?
I think I may have already discussed the need for this with someone somewhere....EH.

This is one of the reason this thread's discussion about SIP peer-peer is wrestling with. In the past, this function for larger networks is done in the LAN's PBX (local SIP Registrar). But, in SIP peer-peer, all tables for how to talk must be maintained and kept consistent, constantly, somewhere, locally. That's what I am trying to piece together based on the little I know about the architecture. (and to be able to call yourself a "SIP" client, that has to be adherence to some base architecture.)

@Smokey01 you are helping us you provide guidance in PET use. Thanks as we progress.
Hope that shares what I was getting at.

gcmartin

#31 Post by gcmartin »

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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#32 Post by Flash »

The second issue is the only one I'm interested in solving. :)

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#33 Post by Lobster »

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.
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 :)
Last edited by Lobster on Wed 24 Aug 2011, 09:05, edited 1 time in total.
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#34 Post by Lobster »

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 8)
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 8)
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#35 Post by Caneri »

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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