PSip - Puppy SIP

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Evil20071
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#21 Post by Evil20071 »

Been doing some things on the windows side of things so I haven't been able to download the new version. I'll do that in a minute.

EDIT: Correction. I've done that. Seems to work fine for me. I don't' see any massive change in CPU usage with the new version over the old. Yes, I've made a call on it to check the voice mail that our crustacean friend left. Sound quality is still just as good as it was before. If you want any detailed information about my computer, you can click the CPUz Validation bar in my signature.
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puppyluvr
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#22 Post by puppyluvr »

:D Hello

@Lobster,
Got your voicemails, so here are my system stats..
Compaq Evo N610c... Pentium 4...1.8ghz 768 mb ram, ATI mobility 7500, 60 gig HD,

@Eric has vm`d me about having a time/cpu tested conversation. What is a good CPU monitor for
Dingo? Conkey?

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Lobster
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#23 Post by Lobster »

puppyluvr wrote:What is a good CPU monitor for Dingo? Conkey?
Hi Jay, thanks guys
Open a console, type 'top'.
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puppyluvr
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#24 Post by puppyluvr »

:D Cool, Thanks...

Cpu usage for psip...3.2%
Firefox...11%
Good job ... :D

@ Lobster, I checked, and I am me, in my configuration...

@ Just had a conversation with David "down under" and he was clear as a bell. However, he said I was breaking up, but it may be my mike.....

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#25 Post by gazb »

puppyluvr wrote::D


@ Just had a conversation with David "down under" and he was clear as a bell. However, he said I was breaking up, but it may be my mike.....
It was me gazb that you had a chat from Down Under
Gizmo Project name
gazb689
I think it might have been your upload packets getting lost, voip is good with short packets as in codec 729a

John Doe
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#26 Post by John Doe »

'John Doe' reporting in...

1-Success with 0.9.9; I signed up, logged on and checked my voice main, heard some news options, and even started a conference call at one point. A mic is around somewhere but I'll have to dig it up and get it working before I try speaking.

2-One small note.

After having entered a completely incorrect buddy in the adduser dialog, everything bombs. (even on X restart, etc...)

See following (last line is probably most important):

Code: Select all

# psip
# killing pjsua process 5936
./psip-pjsua-controller: line 15: kill: (5936) - No such process
killing PSIP process 5642
/usr/local/psip/psip-gui: line 243:  5642 Terminated              gtkdialog3 --program DIALOG --name PSIP
waiting for pjsua to shutdown
killing pjsua process 5946
./psip-pjsua-controller: line 32: kill: (5946) - No such process
I am 5934, I failed to kill pjsua process: 5946
killing pjsua process 5955
./psip-pjsua-controller: line 32: kill: (5955) - No such process
I am 5934, I failed to kill pjsua process: 5955
 00:40:38.205 os_core_unix.c pjlib 0.9.0-release for POSIX initialized
 00:40:38.227 sip_endpoint.c Creating endpoint instance...
 00:40:38.228          pjlib select() I/O Queue created (0xb7ba0098)
 00:40:38.228 sip_endpoint.c Module "mod-msg-print" registered
 00:40:38.228 sip_transport. Transport manager created.
 00:40:38.228    pjsua_app.c Error: invalid URL 'nouser' in --add-buddy option
perhaps some type of regex validation can be added to that dialog?

I tested the three following entry types for buddy in the config file and only the first did the aforementioned behaviour:

--add-buddy nouser

--add-buddy sip:puppy_jd@proxy01.sipphone.com

--add-buddy sip:nouser

Perhaps I tried them all with = for spaces also but can't remember for sure.

cheers! :-)

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Lobster
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Low down and dirty users required

#27 Post by Lobster »

Thanks guys :)
Low down and dirty users required:

In order to test if the new pjsua part of PSIP is working we need command line users:

go to user/local/psip
open a terminal from that directory and
type at the console

Code: Select all

./pjsua-i686-pc-linux-gnu --config-file=pjsua.cfg.default
You may also find your cfg file in /root/.psip
in which case

Code: Select all

./pjsua-i686-pc-linux-gnu --config-file=/root/.psip/pjsua.cfg
should work with your settings

report back how it works

the console commands appear every time you press enter

how they work here:
http://tmxxine.com/wik/wikka.php?wakka=PSIP
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#28 Post by peppyy »

Still a no go on 3.01, Thinkpad T22 full hd install. I have been comparing files between 0.7 which runs very well on this installation and 0.99 to see if I can find where the problem is. I understand that this is a (New) app but I would like to see a version that runs on older Pups.
I believe it must have been when it was recompiled?

Hopefully I will have a machine with Dingo soon and can be of more help.
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HairyWill
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#29 Post by HairyWill »

@peppy
Are you able to use the included version of pjsua from the commandline?

I had a 40 minute conversation with tom today, no problems. It also helped to clean out my head. We both have static public IP addresses and I have a strong suspicion that dynamic public ip addresses may be the common factor in some peoples dropped calls. To solve this it may help if you include the following line in your config file
--use-ice

http://www.voip-info.org/wiki-ICE
http://www.jdrosen.net/papers/draft-ros ... ce-00.html

also as CEL said
specify gizmo as your stun server
--stun-srv=stun01.sipphone.com

fingers crossed
Will
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Evil20071
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#30 Post by Evil20071 »

Working on adding to my userbar animations and was going to add a PSIP User frame to it, but I was hoping to use a graphic for it kind of like the ones for XP and Puppy Linux. Is there an actual logo image for PSIP yet? If not, I'll just use this one from the Wiki page:
Image
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Evil20071
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#31 Post by Evil20071 »

Updated to the 0.9.10 Debug release. Opening with the pjsua_custom_01-0.9.0 will load PSIP, but I get no audio, even after changing the settings as per the VOIP thread. I can't load PSIP with the pjsua_standard-0.9.0 version. Ideas? Suggestions?

EDIT: The Standard one started working?
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#32 Post by Lobster »

restart x
use the standard version

makes sure you are using Puppy 4 Dingo
start Dingo with 'puppy pfix=ram' if you have a lot of modifications

I had the Macpup add on and this caused some inability to read the buddy list - so I removed it.

Use the latest version
http://tmxxine.com/sip/psip-0.9.11.pet


NEW
  • More help messages on the pull down menu
    More pop dialogues
    Text messages should work for everyone.
    The gui now runs independently of pjsua, ie if you close the gui pjsua will be left running. if you want to stop pjsua then there is an entry under phone on the pull down menu.
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Evil20071
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#33 Post by Evil20071 »

Lobster, that works great. I got it installed and it works like a charm . I had some small problem getting it to install at first, but after a quick X restart, it went like a charm. I would like to suggest that anyone else installing it save the .pet to your computer and close all programs prior to installing the .pet. Reason being that when I attempted to install it with Pidgin and FF3 running, X locked up on me and I had to do a hard restart to get it back up and running. (I hate doing that too... ) Other than that, the sound quality is top notch on my end. No problems getting it to run once it installs either. If I have any issues, I'll let you know.
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HairyWill
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psip-0.9.12

#34 Post by HairyWill »

After much error trapping here is psip-0.9.12

I have replaced the standard portaudio in the pjsua build with those from pjproject-0.8.0 which has stopped the audio cut-outs that lobster and smokey were experiencing.

If you want it to, the application should run nicely in the background, just popping up windows when you get calls, IMs or status change messages (these should probably be optional).

The debug option is now able to display the call quality which includes call time and also total MB transmitted.

I have cranked down the quality to 4 in the included config file which means it will probably work on dialup. If both ends of your calls are always broadband and your cpu is not on its last legs this can be set as high as 10.

The crash reporter now includes your config file minus the password. It also uses tar.gz instead of tgz so that the logs will work with forum PMs and the log should be named with your username, to make it easier for me to keep track of them.

Have fun.
Last edited by HairyWill on Tue 15 Jul 2008, 20:45, edited 1 time in total.
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HairyWill
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#35 Post by HairyWill »

for those that know anything about ALSA and OSS in puppy, here is the email from Benny about portaudio.

> On Fri, 11 Jul 2008 16:33:59 +0100
> "Benny Prijono" <bennylp@pjsip.org> wrote:
>
> > So this is not a call drop in the usual sense then since signaling seems
> to
> > be still in place (calls are still CONFIRMED on both ends), but rather
> lost
> > of audio after few minutes of conversation. Is that right?
> >
> I think we have made the problem go away.
> The last version I compiled that worked properly for this user was from
> subversion around 26th June. But svn on puppylinux doesn't work with ssl
> which is needed to download portaudio. So I just used the portaudio source
> distributed with pjproject-0.8.0.
>
> We tried today using pjproject-0.9.0 but with portaudio replaced with that
> from pjproject-0.8.0.
> The call gets successfully moved to the CONFIRMED status and we had a call
> that stayed up for over 20 minutes before we hung up.
>
> Thank you for your help in solving this problem. If you want to investigate
> further and need any further information give me a shout.
>
>
That's strange indeed. But it's good to know that the problem doesn't seem
to come from pjsip.

Re: PA, one of the thing that changed between 0.8 to 0.9 is (IIRC) PA
changed the default sound backend in Linux from OSS to ALSA (so if both are
present, ALSA will be used unless pjsua is configured to use other device
ID). That might be the reason.

Cheers
Benny
Will
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Caneri
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#36 Post by Caneri »

Hi All,

Ok...tested 0.9.12 with Will....very good sound quality...other than playing phone tag when we were both calling each other at the same time...and I was voice messaging Lobster...lol

Grant and I had a great chat this morning (here it was morning)..his voice was like it was next door and the call was from Australia to Canada with no lag noticeable...impressive!

I use alsa mixer for my headset and need to run this at every boot but I think that's a dingo thing...dunno.

My overall impression is this ones a keeper..well done Will and crew.
Best,
Eric
[color=darkred][i]Be not afraid to grow slowly, only be afraid of standing still.[/i]
Chinese Proverb[/color]

Caneri
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#37 Post by Caneri »

Ok..tested 0.9.12 on Fire Hydrant 3.01 and wNOP v0.2 and so far nothing worked for me....crash report has been sent to Will. Psip installed ok by clicking on the .pet file...alas

This was a very short run so I didn't get to deep into it...I hope the report helps Will.

Best,
Eric
[color=darkred][i]Be not afraid to grow slowly, only be afraid of standing still.[/i]
Chinese Proverb[/color]

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HairyWill
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#38 Post by HairyWill »

thanks for the log eric

It is only 5 seconds long and shows pjsua starting up and then shutting straight down again. There seem to be no events triggered by pjsua and the input file for commands to pjsua is also empty.
Did you get the gui?
We might need a sync after the controller creates the temp files, if this is the case then trying to run it again may produce better results.
Now we've solved smokey and lobsters audio problems I will give it a spin on 3.01, later.
Will
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Caneri
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#39 Post by Caneri »

Hi Will,

Yes the gui popped up but there was no functions. I tried to call voicemail..no joy.

The terminal with top said it was running...the whole test was only 5 minutes on each puplet.

Eric

EDIT: when run from the terminal it said "failed to rm /tmp buddylist no such file" or something close to that..sorry I'm not a cli guy...
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HairyWill
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#40 Post by HairyWill »

The gui doesn't currently show whether pjsua is running or not. I'm not sure why pjsua shut itself down. I can use a gtkdialog progressbar to show a one line status message, or maybe there should be handler that complains when you try to execute any pjsua commands and pjsua is not running.

features I'm considering.
1) multiple call handling, at the very least allowing you to reject an incoming call if you are already talking.
2) integrating the call, IM and buddy status buddy lists using a gtkdialog tree
3) confirmation as to whether IMs are delivered
4) adding tones when dialing and pushing dtmf keys

I know how to do 2,3,4 but probably 1 is the most important

add your feature requests here
Will
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