Puppy Phone - VOIP using SIP
- Lobster
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Puppy Phone - VOIP using SIP
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
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
Update: Reported working on Slacko,
Midnight Sun
http://www.murga-linux.com/puppy/viewto ... 517#538517
and Puppy 4.13
Last edited by Lobster on Mon 03 Oct 2011, 08:50, edited 6 times in total.
- Lobster
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It may be possible . . .
http://www.murga-linux.com/puppy/viewto ... 336#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 . . .
http://www.murga-linux.com/puppy/viewto ... 336#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 . . .
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?
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
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let me know if you get it going without a server, I probably won't use it but I'll added it
ttuuxxx
ttuuxxx
http://audio.online-convert.com/ <-- excellent site
http://samples.mplayerhq.hu/A-codecs/ <-- Codec Test Files
http://html5games.com/ <-- excellent HTML5 games :)
http://samples.mplayerhq.hu/A-codecs/ <-- Codec Test Files
http://html5games.com/ <-- excellent HTML5 games :)
- Lobster
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Thanks for the interest guys
This is what Smokey (Grant) wrote to the creator of pjsua
which is the 'engine' behind PSIP
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)
This is what Smokey (Grant) wrote to the creator of pjsua
which is the 'engine' behind PSIP
and this is the reply> 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
It sounds to me like 'geekity-geek-geek-geek'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
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)
Last edited by Lobster on Mon 29 Aug 2011, 23:20, edited 1 time in total.
- Lobster
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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 . . .
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 . . .
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.
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.
Here's what I came up with by Googling 'NAT traversal STUN'Lobster wrote:It sounds to me like 'geekity-geek-geek-geek'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
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)
http://en.wikipedia.org/wiki/STUN
http://www.voiptraversal.com/
http://searchunifiedcommunications.tech ... -STUN-work
http://calthrup.blogspot.com/2009/02/st ... ersal.html
- Lobster
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HairyWill aka Will has just confirmed that we canhi lobster nice to hear from you. Feel free to do whatever you want.
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/viewto ... 446#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
We can stick with PSIP name as Smokey suggests
but use Psippy to refer to this initiative and any modifications/improvements/updates
. . . now where is that USB headset . . .
I have 2 questions which seem to be key for me
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?
- 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?
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?
Last edited by gcmartin on Sat 20 Aug 2011, 18:56, edited 1 time in total.
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
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
I suggested we remain with Psip, it appears lobster has agreed.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?
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: 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).
Neither do I yet but I'm working on it.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.
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.
The pjsua manual can be read from the existing Psip under the help menu item or here http://www.pjsip.org/pjsua.htmgcmartin 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.
Yes but they are all quite large. Psip is way under a Meg which fits better with the Puppy vision.gcmartin wrote: BTW aren't there some SIP to SIP or SIP to POTS implementations already available off-the-shelf?
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
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
Some interesting reading here:
http://wiki.ekiga.org/index.php/Underst ... assing_NAT
http://wiki.ekiga.org/index.php/Underst ... assing_NAT
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
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.
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?