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The time now is Mon 26 Jun 2017, 17:01 All times are UTC - 4 |
 Forum index » Advanced Topics » Additional Software (PETs, n' stuff) » Browsers and Internet |
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PSip - Puppy SIP
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| Page 15 of 18 [258 Posts] | Goto page: Previous 1, 2, 3, ..., 13, 14, 15, 16, 17, 18 Next |
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smokey01
 Joined: 30 Dec 2006 Posts: 2554 Location: South Australia |
it said Call has been disconnected. Temporarily unavailable. That's because I had gone offline. I need to be logged into my sip server for you to see me. The good thing about psip is you don't have to be logged into a server to use it. For example: if you have Psip running and I know your IP address I could add it as a buddy and call you direct. The buddy address entry would be something like sip:192.168.0.6 This is really good if you have a static IP address, not dynamic as it continually changes each time you log on. _________________ Software <-> Distros <-> Tips <-> Newsletters |
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Flash
Official Dog Handler  Joined: 04 May 2005 Posts: 12543 Location: Arizona USA |
I've wanted to try that, emailing my DHCP-assigned IP address to my brother after I've booted my computer for the day. It is my understanding that a "client" program running in Puppy, such as Psip, won't accept a connection it didn't initiate. If I understand correctly, that's what makes it necessary to log into the intermediate server. Will Psip, running in my computer, somehow get around that if I try to make a direct connection with my brother? Do we have to attempt to connect with each other at pretty much the exact same time (so that each one thinks it initiated the connection?) |
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smokey01
Joined: 30 Dec 2006 Posts: 2554 Location: South Australia |
Flash, it suppose to be able to do it. I've been thinking about making a very simple P2P version but I am having a similar problem. Here is the manual http://www.pjsip.org/pjsua.htm In particular here http://www.pjsip.org/pjsua.htm#basic I can make it work within my own network eg: 192.168.0.3 etc but how to make it work on the internet is the issue. I know what my dynamic IP address is but how do I make it translate to my internal network address? This is the problem I'm having. I guess NAT should be able to sort it but I'm not sure how to define it in the config file. Maybe we can work together to sort it out. _________________ Software <-> Distros <-> Tips <-> Newsletters |
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technosaurus
 Joined: 18 May 2008 Posts: 4718 |
This is kinda an old thread, is psip still being maintained? _________________ Web Programming - Pet Packaging 100 & 101 |
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smokey01
Joined: 30 Dec 2006 Posts: 2554 Location: South Australia |
It hasn't been for some time but I am considering doing some work on it. I would like to get it working as a proper peer 2 peer VOIP application without using a sip server. The engine, pjsua, says it can be done as described in the docs in my last post. I might even rework the GUI but I might need some help. _________________ Software <-> Distros <-> Tips <-> Newsletters |
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Flash
Official Dog Handler Joined: 04 May 2005 Posts: 12543 Location: Arizona USA |
All I can contribute are some general thoughts based on my understanding of how the internet transmission protocol works. 
When you enter an IP address in a browser window and click "go", an IP packet is sent from your computer through your router to the server at the internet address you entered. Your router has to somehow keep track of where that packet originated in your network so it can forward the reply back to the originating program (the browser.) Ports come into it somewhere, but I think at the very end, after your computer has received the packet from the router. The same general description must apply for VOIP applications such as Psip. I think VOIP uses a different protocol from TCP, called UDP. UDP is the same protocol used for streaming internet video. UDP is used because it does not require the destination to acknowledge receipt of each packet, saving time and wear and tear on the network.  Some packets might never make it, but for many applications such as VOIP, that's okay; you just hear a bit of noise.
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Aitch
 Joined: 04 Apr 2007 Posts: 6815 Location: Chatham, Kent, UK |
Smokey Try dynamicDNS - it allows a fixed IP Address of your dynamically assigned one http://www.dyndns.com/services/dns/dyndns/ I found PSIP to be unreliable, from memory, I only ever made brief connections to Eric, HairyWill and Lobster, I think I now use skype, but it uses huge resources for what it gets used for.... 
Aitch
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smokey01
Joined: 30 Dec 2006 Posts: 2554 Location: South Australia |
Aitch I've just heard from Benny, the author, who has given me a couple of avenues to explore. I'm surprised that Psip was unreliable for you as Will and myself spent many hours communicating with it while we were in development. Anyway I will also investigate your suggestion. Thanks _________________ Software <-> Distros <-> Tips <-> Newsletters |
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smokey01
Joined: 30 Dec 2006 Posts: 2554 Location: South Australia |
Aitch maybe you should try a free sip account from https://www.ekiga.net/ Your account in Psip would look something like: --id sip:Aitch@ekiga.net --registrar sip:ekiga.net --realm * --username Aitch --password xxxxxx --reg-timeout 55 Let me know a suitable time and we could test. I'm GMT +9.5 hours. _________________ Software <-> Distros <-> Tips <-> Newsletters |
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tide
Joined: 13 Aug 2010 Posts: 56 |
Could some kind soul please please tell me how to configure PSIP? I signed up with sipgate.co.uk but after trying for hours, google, forumsearch etc. just can't get it to work! Sipgate tells me that:
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smokey01
Joined: 30 Dec 2006 Posts: 2554 Location: South Australia |
tide, PM me a copy of your ./psip/pjsua.cfg file and I will have a look for you. _________________ Software <-> Distros <-> Tips <-> Newsletters |
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WB7ODYFred
 Joined: 13 Dec 2008 Posts: 144 Location: Oregon & Washington |
Hello Tide, and others. I posted a working copy of my pjsua.cfg in a previous forum entry, May 9 2008 WB7ODYFred in this forum thread, Page 14. I also posted testing ideas, that I used to figure out why PSIP died/Hung with no errors reported. Start PSIP from command line in RXVT terminal window. http://murga-linux.com/puppy/viewtopic.php?t=31061&start=205 Here is Tides quoted entry from above.
Tide edit your /root/.psip/pjsua.cfg with Gedit or Geany. This is what I think your file should look like. Then start pjsua from a RXVT terminal window to see what it reports back during startup. Copy paste the line below (after the '#" prompt) into a terminal window. maybe the version of pjsua is different on your version of PuppyLinux. Use what works for you. Mine was from 4.1.11 version of Puppy Linux.
Let us all know (Smokey01 and WB7ODYFred and other readers) if the above works for you. If you need to use a stun server with sipgate you might uncomment the last line which I modified to reflect your SIP provider, ie sipgate.co.uk. #I think this example might only work if you have a gizmo account # --stun-srv=stun01.sipphone.com # --stun-srv=sipgate.co.uk second look at creating a free Ekiga.net account at www.ekiga.net you can the use PSIP with your free ekiga.net account settings and call SIP 500@ekiga.net and play with echo test to verify your microphone settings. Gizmo (ie sipphone.com ) was bought out and no longer provides free SIP accounts. Ekiga.net does provide free sip accounts and you don't have to use their Ekiga software to make ekiga.net calls. So this seems like a replacement for sipphone.com where Puppy PSIP users can talk with each other. I also wanted to try using a ekiga chat room channel to have a round table discussion with a few other PuppyLinux users on PSIP or Ekiga sip phones. At a set UTC time we could call room number 5012345@ekiga.net and have several people chat on an issue. Like testing PSIP or hearing what would you like different in the simple user interface. from this web page https://www.ekiga.net/index.php?page=services Dial the sip:501xxxx@ekiga.net (where x = any digits from 0 to 9). These rooms are public or private, anyone can join a conference at any time if he chooses the right number or you can protect the access with a PIN number. The first person to enter a 'room' may specify a PIN by entering a number (followed by #) to limit access to the conference room, just entering # will make this conference room public. When the last person leaves a conference room, the PIN will be cancelled and others may use the 'room' Fred Finster WB7ODYFred  Waiting for your forum comments and PSIP test results via VoIP messages
SIP:wb7odyfred@ekiga.net SIP:fredfinster@sip7.vitelity.net ps. Smokey01, thanks for keeping PSIP going and answering questions. Last edited by WB7ODYFred on Fri 26 Aug 2011, 17:35; edited 2 times in total |
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Caneri
Joined: 04 Sep 2007 Posts: 1569 Location: Canada |
_________________ Be not afraid to grow slowly, only be afraid of standing still. Chinese Proverb |
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smokey01
Joined: 30 Dec 2006 Posts: 2554 Location: South Australia |
Hi Fred, long time no see. Is Psip-0.12 working for you with the latest Puppies? Have you tried replacing the updated psip-gui? http://www.murga-linux.com/puppy/viewtopic.php?p=555750#555750 _________________ Software <-> Distros <-> Tips <-> Newsletters |
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Caneri
Joined: 04 Sep 2007 Posts: 1569 Location: Canada |
I'm online now..I just saw smokey01 but he disappeared. sip:caneri@iptel.org _________________ Be not afraid to grow slowly, only be afraid of standing still. Chinese Proverb |
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tide, on page 14 I posted a copy of /root/psip/pjsua.cfg| Page 15 of 18 [258 Posts] | Goto page: Previous 1, 2, 3, ..., 13, 14, 15, 16, 17, 18 Next |
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