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Linphone alpha testing
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Old 02/09/2011, 02:00 AM   #1 (permalink)
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This is the thread for alpha testers of the Linphone application to discuss their results and assist each other.

At the moment, to get Linphone you have to already have a good understanding of VoIP (the open standard SIP kind, not the closed proprietary Skype kind) and use the following links:

http://bit.ly/linphone-0-1-8-pre (Pre, Pre+, Pre 2, Veer, TouchPad)

http://bit.ly/linphone-0-1-8-pixi (Pixi, Pixi+)

Thibaud is the lead developer of this app, and WebOS Internals is collaborating with some coding, packaging and release assistance.

If you are not actively alpha testing Linphone, please do not post in this thread. Random questions from folks who are not alpha testing will probably be ignored - that phase of release and support will come later.

If you are alpha testing, please report your configuration and results on the wiki page: http://www.webos-internals.org/wiki/...ation:Linphone

Thanks for your interest, and if you do not yet meet the criteria above, please be patient.

-- Rod

Last edited by rwhitby; 01/25/2012 at 06:12 PM.
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Old 02/09/2011, 02:09 AM   #2 (permalink)
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Already saw this on P|C site. I'm very excited to hear this. I'm using sipgate for years.
Thanks so much webOS-Internals.
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Old 02/09/2011, 02:53 AM   #3 (permalink)
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Can you summarize requirements for visitors to this thread? For example, are they required to be operating a SIP server such as Asterisk, etc. If you can just list a few requirements, people will be able to just check through the list to know if they should request access to this for testing.

Thanks!
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Old 02/09/2011, 03:15 AM   #4 (permalink)
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You need to have an existing SIP account and know how to use it.

-- Rod
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Old 02/09/2011, 03:18 AM   #5 (permalink)
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Just tried this on my Pre for the first time.

Using a GSM Pre over Wifi at home and called to my landline. My SIP provider is 2Talk, a Local New Zealand VOIP Provider and I just entered in the simple Server, Username, Password.

The call connected very quickly and the audio that came through was quite clear. It wasn't up to the standard of normal calls (it was a little sharp and digitized) but it was more than good enough to use. The application was quite stable and responsive (which is more than what I could say about the old Android SIP clients I used).

Though I wasn't able to receive a call.

Will do more testing later.

Though, I know
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Old 02/09/2011, 03:33 AM   #6 (permalink)
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Hi,

THANKS for your wokr! I just tested it and it works. I have a AVM Fritzbox router, which includes a SIP server. Thus I can use my Pre as a cordless phone at home. Placing and receiving calls work

I have 2 of theses. 1 @home and 1 @work. Both connected via VPN and I could even register on the router @home (from the office) and call my landline office phone via my landline home phone

I used the IP adresses of my routers as domain in the prefs and it worked. Each router can be accessed at "http://fritz.box" from any client that is attached locally (LAN or WIFI). If I set "fritz.box" or "http://fritz.box" linphone can`t connect. I set up both sip servers with identical "phone data". Thus using my Nokia E71 I had before my pre, it would connect to both boxes automatically and I didn`t have to choose home/work.

One question:
Can I use "fritz.box" as domain? Then the settings in the preferences would be the same @home and @work.

Thanks for your efforts!!!! VOIP and Skype are the missing pieces in the webOS puzzle for me

Marcel
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Old 02/09/2011, 05:42 AM   #7 (permalink)
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This sounds really great ! Want to donate. Is there a seperate link for donations towards development of Linphone or the standard donation link can be used ?
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Old 02/09/2011, 06:18 AM   #8 (permalink)
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Quote:
Originally Posted by kbywal View Post
This sounds really great ! Want to donate. Is there a seperate link for donations towards development of Linphone or the standard donation link can be used ?
Thank you, this is much appreciated.

Until I set up a link (will share between WOI & dev), please donate to WebOS Internals for now!
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Last edited by Thibaud; 02/09/2011 at 06:47 AM.
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Old 02/09/2011, 06:21 AM   #9 (permalink)
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Mark the donations in the comment field with Linphone and we'll keep track of them.

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Old 02/09/2011, 06:31 AM   #10 (permalink)
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Quote:
Originally Posted by D2 View Post
Just tried this on my Pre for the first time.

The call connected very quickly and the audio that came through was quite clear. It wasn't up to the standard of normal calls (it was a little sharp and digitized) but it was more than good enough to use. The application was quite stable and responsive (which is more than what I could say about the old Android SIP clients I used).

Though I wasn't able to receive a call.

Will do more testing later.

Though, I know
Sounds quality: most probably depends on the CODEC used (I let the Linphone core decide for now, although there are internal options I'll make available later in the app).

Call (not) received: can you elaborate on your configuration and how you manage to initiate the call on the other side? Please PM if you want to keep your personal data ... private!
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Old 02/09/2011, 06:46 AM   #11 (permalink)
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Quote:
Originally Posted by mgmft View Post
Hi,
I used the IP adresses of my routers as domain in the prefs and it worked. Each router can be accessed at "http://fritz.box" from any client that is attached locally (LAN or WIFI). If I set "fritz.box" or "http://fritz.box" linphone can`t connect. I set up both sip servers with identical "phone data". Thus using my Nokia E71 I had before my pre, it would connect to both boxes automatically and I didn`t have to choose home/work.

One question:
Can I use "fritz.box" as domain? Then the settings in the preferences would be the same @home and @work.

Thanks for your efforts!!!! VOIP and Skype are the missing pieces in the webOS puzzle for me

Marcel
The app will register the proxy information from the domain if you don't specify the proxy explicitly.

Having Linphone operate fine with the router's IP address as domain and fail with the plain router's name suggests a DNS issue. Can you try the following commands on the Pre (in the terminal app or through ssh):
Quote:
/bin/ping -c 1 fritz.box
and check the reported IP address matches the one you manually entered?
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Old 02/09/2011, 07:11 AM   #12 (permalink)
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Hi Thibaud,

strange, but now it works! Who knows, why it didn`t work before. Thanks!

Marcel
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Old 02/09/2011, 08:01 AM   #13 (permalink)
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Originally Posted by mgmft View Post
strange, but now it works! Who knows, why it didn`t work before. Thanks!
Good to hear anyway!

Remember this is alpha-stuff... I find it quite robust though, but I'm biased here!
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Old 02/09/2011, 08:55 AM   #14 (permalink)
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Alpha - because there are very few featuresfor now!? But for "normal" phone calls it is stable

My recent issues:
- Most of the times it uses the loudspeaker. Thus the other side can hear it own echo! Is there a way to toggle the loudspeaker?

- My sip server has an answering machine build in. It has the number "**600" (other internal numbers are "**6xx"). When I dial "**600" the button changes to "Call idle" and the only thing I can do is close the app.

- Sometimes I also couln`t receive calls.

- BTW: Which codes does linPhone support? Only G.711 or also HD (G.722)?

Thanks

Marcel

Last edited by mgmft; 02/09/2011 at 09:29 AM.
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Old 02/09/2011, 04:39 PM   #15 (permalink)
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everytime i make a calls it says "error needs to confirm registration" domain is iptel.org and i entered my username where it says name.
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Old 02/09/2011, 05:00 PM   #16 (permalink)
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Quote:
Originally Posted by mgmft View Post
Alpha - because there are very few features for now!? But for "normal" phone calls it is stable
Thanks, much appreciated.

Quote:
- Most of the times it uses the loudspeaker. Thus the other side can hear it own echo! Is there a way to toggle the loudspeaker?
You mean all time, right? The loudspeaker is currently a "feature"... Kidding: must be in the "com.palm" to have media control over front & back speakers (and proximity sensor too). I got it to work fine as soon as the app would pretend to be in Palm's domain. But there are other ways to achieve this, working on it. Headset with mic is the way to prevent echo at the other end at the moment.

Quote:
- My sip server has an answering machine build in. It has the number "**600" (other internal numbers are "**6xx"). When I dial "**600" the button changes to "Call idle" and the only thing I can do is close the app.
Unexpected for sure, thanks for reporting it.

Quote:
- BTW: Which codes does linPhone support? Only G.711 or also HD (G.722)?
You'll want to read the Linphone features page, guess it speaks for itself. Selection is possible in the core, but not priority #1 in the app. Will do for sure.
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Last edited by Thibaud; 02/13/2011 at 06:40 AM.
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Old 02/09/2011, 05:05 PM   #17 (permalink)
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everytime i make a calls it says "error needs to confirm registration" domain is iptel.org and i entered my username where it says name.
You mean you got a popup with this message? This is triggered by the core reporting a CALL_ERROR, which is of course an exception.

Guess we'll need someone else to test iptel.org so we can discriminate the app over the context.
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Old 02/09/2011, 05:20 PM   #18 (permalink)
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I tried at home: Tried to call from another internal sip phone with # "**622". The phone I use for calling said "Pre ringing". The button on the pre kind of flickers. It changes text very fast. Something with "start" and back to idle. Too fast to read. No call signalized. Happens every time I try to call from **6xx. Haven't tried calling from outside though.

Marcel

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Old 02/09/2011, 05:35 PM   #19 (permalink)
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I'm able to make outgoing calls but incoming calls never work. it just never prompts me to pick up. very similar if not identical to mgmft.
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Old 02/09/2011, 05:42 PM   #20 (permalink)
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Keep in mind that if you have a Gizmo account and a Google Voice account, you can use this for completely free calling...
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