Puppy Phone - VOIP using SIP

[Lobster]

This Sunday conference meet, Grant took us through all the updates.
Very good help file now available online
http://www.smokey01.com/help/psip/psip.html
Which chromium (on my system) is still incapable of displaying. Que?

Thanks to everyone who turned up
We need more users. Puppy phone certainly works.
We managed to boost or lower peoples mic settings.
and discuss Puppy Pi
http://elinux.org/RaspberryPiBoard
http://www.raspberrypi.org/?page_id=8

Next conference call
Sunday 18th September 2011 at 1300h GMT daylight saving time

[Sylvander]

Testing my 2 microphones:
1. USB standalone desktop mic by Logitech.
2. Swivel mic included on my headset = "Plantronics Gamecom 367".

A. Tested my USB mic in Win2000Pro using the sound recorder program, and it worked OK.

B. Rebooted into "Midnightsun-003-Tmix", and used the sound recorder included there.
It was hearing my headset mic, but rather faint, so in Retrovol I ticked the "mic boost" & "mic boost capture", which boosted the sound very noticeably.

C. Rebooted into my number-1 Puppy = Lupu -526.
Ran the Puppy_phone program...
Used the headset mic switch to switch on the mic, and could immediately hear myself in the headphones.
Clicked "IPTEL->Echo_test_call" and heard everything I said being echoed back through the headphones.
What a GREAT way to do the job!
So both headset mic and headphones are functioning well.
All ready to go methinks.

[dogle]

We managed to boost or lower peoples mic settings.

- this was enlightening. Smokey01 and I had had a chat shortly before the conference, and tweaked our mic settings then. At the start I could barely hear Lobster, and Caneri was pretty faint.

It seems that the path through the SIP process attenuates the sound level quite a bit, and setting the mic using the Test Sound menu does not reflect this. I wonder, would it be possible for Test Sound to 'simulate' this attenuation, thus inducing the user to set the mic a tad higher?

I find that I need a mic setting close to max +20dB boost. This prevents me from using a handset, which sets up acoustic feedback howl across itself.

Thinking that there must be some spurious electrical crosstalk in the system to cause this, I spent some fruitless hours chasing my tail trying to nail it ..... then the elephant in the room broke wind, kaboom! - ALSA is simply set up to feed mic input to speaker by default! Quite reasonable huh, except when you don't want this to happen. Not good for VoIP, you want to hear the other guy's voice, not your own (loud).

Unfortunately none of the various mixer GUIs seem to offer any way to overcome this. Trying to figure out a way from the ALSA documentation I just got savaged ..... my head is still spinning.

Any ALSA gurus out there?

[Lobster]

Think you mentioned you had this Grant.
Just in case still relevant . . . the enclosed quickstart guide goes in the resources directory and is for off-line usage.

Can we make use of it?

[mcewanw]

ALSA is simply set up to feed mic input to speaker by default! Quite reasonable huh, except when you don't want this to happen. Not good for VoIP, you want to hear the other guy's voice, not your own (loud).

Not sure if this is what you are after but if you start the program alsamixer you don't want any mike gain on the Playback settings (only on the Record or Capture settings). That is, tab to the alsamixer Playback control settings, move the cursor to the Mic setting control, once there make sure the gain for Mic playback is set at an absolute minimum and also press key M to toggle the Mic control such that is not selected at all in Playback mode.

[Sylvander]

1. "All ready to go methinks."
...Except I'd like to be able to choose WHICH mic to use.
Using software...
Rather than being forced to only connect 1 mic...
Or else only having 1 mic switched on using its inbuilt switch.
.
.

[smokey01]

mcewanw, excellent advice. It works a treat here.

dogle this will probably fix your problem.

[smokey01]

1. "All ready to go methinks."
...Except I'd like to be able to choose WHICH mic to use.
Using software...
Rather than being forced to only connect 1 mic...
Or else only having 1 mic switched on using its inbuilt switch.
.
.

Why would you want to connect two mic's?

[dogle]

mcewanw, many thanks
smokey01, PM

[Sylvander]

1. "Why would you want to connect two mic's?"
(a) If I'm on my own, and I want to make a call, I'd use the headset with it's mic.

(b) If my wife and self, want to make a call to someone [our son, his wife and their children?]...
Such that we can both speak, and both hear...
Then I'd use the desktop mic, and the speakers.
We've done that in the past using Skype.

(c) But I don't want to be forced to begin re-arranging the hardware connections, or do any such thing that takes lots of effort.
Would be nice to leave all the hardware connected [I can do that with the headphones and amplified speakers, and turn down the speaker volume if not needed, or not wear the headset], and use software to choose which mic to use.
There is a mute switch on each mic, but that doesn't help if the OS only sees ONE mic, especially if it happens to be the WRONG mic.
e.g. Lupu-526 sees only the headset mic, so it's only good for me, not both of us.
Windows sees the desktop mic, but not the headset mic.
And I don't have the Puppy_phone in Windows, only Skype.

[oliver69]

Hi All,

how can i set the name and the port for the stun-server?

[gcmartin]

Hi All,

how can i set the name and the port for the stun-server?

So you mean in your router?

Hope this helps

[sc0ttman]

I added the latest pet to Akita Linux... Good stuff, I got an account and everything, heard the test music, added a contact, it showed up... Dont have a mic at the mo, and the Eee PC built-in mic wont work in Akita yet... So not made any calls as yet..

PSIPPY will be in the next Akita ISO, out very soon... Also, I managed to hack it up a little bit to use ash, not bash - this means it should use a little less RAM and will be a little bit quicker - every little helps for some users!!

French translated version: http://murga-linux.com/puppy/viewtopic. ... 148#563148

The problem is that the translations are hard coded, all english removed.. So we still do NOT have an internationalised version, but now we have 1 english-only version, and 1 french-only version... I hope to tackle this soon, but if anyone wants to beat me to it, then please DO!

Cheers!

[oliver69]

Hi All,

how can i set the name and the port for the stun-server?

So you mean in your router?

Hope this helps

no the phone application has to use the stun-server and need to know the servername and port. if you abroad (like a hotel or a customer site) you have no access to the local router and need a way to connect from the sip provider to your phone (signaling income calls). It worked with linphone (which i could not get to work in puppy).

[Caneri]

this is a handy page http://www.iptel.org/service

[Caneri]

I seem to have lost my post...grrr

[oliver69]

this is a handy page http://www.iptel.org/service

fine site....but my problem is that a already have an account at sipgate.de (the largest provider in germany) and a lot of my customers know the number - so it's difficult to change and i need a service which can dial to the pstn and even can be called from pstn. Without stun-settings i can call but can't be called (the call is not signaled). it looked around for a linphone pupplet but can find only a very old (cli only) which also did not work - seems a lot of missing dependencies.

[smokey01]

this is a handy page http://www.iptel.org/service

fine site....but my problem is that a already have an account at sipgate.de (the largest provider in germany) and a lot of my customers know the number - so it's difficult to change and i need a service which can dial to the pstn and even can be called from pstn. Without stun-settings i can call but can't be called (the call is not signaled). it looked around for a linphone pupplet but can find only a very old (cli only) which also did not work - seems a lot of missing dependencies.

oliver69 this page appears to describe what you are after.
http://www.sipgate.com/faq/article/397/ ... oIP_device

Modify your ./pjsip/pjsua.cfg to:

--stun-srv=stun.sipgate.net
--id sip:yournumber@sipgate.com
--registrar=sip:sipgate.com
--realm *
--username=yournumber
--password=yourpassword

This should work.

[Caneri]

pstn at iptel.org under "other"

[smokey01]

I added the latest pet to Akita Linux... Good stuff, I got an account and everything, heard the test music, added a contact, it showed up... Dont have a mic at the mo, and the Eee PC built-in mic wont work in Akita yet... So not made any calls as yet..

PSIPPY will be in the next Akita ISO, out very soon... Also, I managed to hack it up a little bit to use ash, not bash - this means it should use a little less RAM and will be a little bit quicker - every little helps for some users!!

French translated version: http://murga-linux.com/puppy/viewtopic. ... 148#563148

The problem is that the translations are hard coded, all english removed.. So we still do NOT have an internationalised version, but now we have 1 english-only version, and 1 french-only version... I hope to tackle this soon, but if anyone wants to beat me to it, then please DO!

Cheers!

sc0ttman it works quite well. I would be interested in testing Psip with you some time soon when you get a mic.

There is still quite a bit of development occurring with Psip. Shortly you may see a single file version complied entirely in C, no gtkdialog.

How good will that be?