Application:Linphone

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Icon WebOSInternals Linphone.png

Linphone - 0.1.8


This is the very first true VoIP application running on any webOS device.

Linphone for webOS is a port of the original Linphone command-line application.

Summary

Linphone is a general purpose SIP softphone. It allows you to place and receive VoIP calls on any SIP account you may have.

Linphone isn't bound to any operator. Because it is compatible with SIP, it can work with any VoIP operator using SIP (most of them use SIP, the most notable exception being Skype).

Alpha Test

Linphone / webOS is currently in alpha test.

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 find the download links in the PreCentral thread.

Alpha testers, please report at the bottom of this page success & failure with your SIP provider(s). Add or update your own rows in the table (please use your P|C or WOI identity so we can PM you if needed).

Any other information/request you would like to share/ask, please report in the dedicated thread at PreCentral.

SIP Account

You will have received some registration information from your SIP provider by the end of the sign-up process:

  • username or phonenumber
  • password
  • domain
  • proxy (optional)


Your SIP identity is formed by grouping the Username or Phonenumber and the Domain together:

  • sip:username@domain

or

  • sip:phonenumber@domain

and will be used by other people willing to call you on your webOS device when running Linphone.

Dialpad

Linphone Dialpad.png

This is (obviously!) where you dial a number you want to call...












Preferences

The Linphone preferences screen can be accessed by tapping the menu button on the top left corner of your device.

Linphone Preferences.png


The "SIP" group:

  • NAME - enter your username or phonenumber
  • PASSWORD - enter your password
  • DOMAIN - enter your SIP provider's domain
  • USE PROXY: select
- NO if your SIP provider does not require a specific proxy (PROXY field hidden)
- YES if a proxy is required with an address different from domain (PROXY field shown)
  • PROXY - enter your SIP provider proxy (or proxy:port), if any

The "NETWORK" group:

  • FIREWALL - select
- NONE if you have a direct connection to the internet or you are behind a router with an application-level gateway for SIP (ADDRESS and SERVER fields hidden)
- NAT to provide the public IP address of your router (ADDRESS field shown)
- STUN to rely on a STUN server to discover the public IP address of your router (SERVER field shown)
  • ADDRESS - enter your router's public IP address
  • SERVER - enter the address of the STUN server

Limitations

  • The sound comes from the back-speaker for now. Work is underway to bring the sound from the front speaker
  • Until echo-cancellation is activated, the person on the other end will suffer some echo because of the back-speaker...
  • The proximity sensor is not activated (yet!). To prevent unintentional screen touch from the ear when close to the head, it is better to turn-off the screen for now...

Test Report

SIP Provider VoIP Test Test Conditions
Name Domain Proxy Firewall Register Call-Out Call-In Linphone Network (WiFi/EVDO/3G) Device When Who Comments
Freephonie freephonie.net (empty) NONE OK OK (pstn) (untested) 0.1.8 WiFi (DSL router, 18/1Mbps) Pre- 1.4.5 Mar 2011 Thibaud Call-in not really a feature of Freephonie
iptel.org iptel.org sip.iptel.org (optional) (n/a) OK (untested) OK (sip) 0.1.6 WiFi (DSL router, 18/1Mbps) Pre- 1.4.5 Feb 2011 Thibaud Call initiated from linphonec-3.1.2@Ubuntu-9.10
SIP2SIP sip2sip.info proxy.sipthor.net (n/a) OK (untested) OK (sip) 0.1.6 WiFi (DSL router, 18/1Mbps) Pre- 1.4.5 Feb 2011 Thibaud Call initiated from linphonec-3.1.2@Ubuntu-9.10
Gizmo proxy01.sipphone.com (empty) (n/a) OK (untested) OK (sip) 0.1.6 WiFi (DSL router, 18/1Mbps) Pre- 1.4.5 Feb 2011 Thibaud Call initiated from linphonec-3.1.2@Ubuntu-9.10
Draytel draytel.org (empty) (n/a) OK (untested) OK (sip) 0.1.6 WiFi (DSL router, 18/1Mbps) Pre- 1.4.5 Feb 2011 Thibaud Call initiated from linphonec-3.1.2@Ubuntu-9.10
sipgate sipgate.com sipgate.com NONE OK OK OK (most of the time- inconsistent), (sip) 0.1.8 WiFi (University of Kentucky WiFi) Pre- 2.1 Mar 2011 (updated) Ricyteach Call initiated from Voogle (Google Voice)
sipgate sipgate.com sipgate.com STUN, stun.sipgate.net:10000 OK untested hangs at INCOMING_INITIATED (sip) 0.1.8 WiFi (University of Kentucky WiFi) Pre- 2.1 Mar 2011 Ricyteach Call initiated from Voogle (Google Voice)
pbxes pbxes.com pbxes.com REG PENDING 0.1.8 Pre2 2.1 Mar 2011 wprater
vonage sphone.vopr.vonage.net (empty) Tried With and Without OK* OK* (to vonage & sprint mobile) OK* (from vonage & sprint mobile) 0.1.8 Both Wifi+NAT and Sprint Pre 1.4.5 Mar 2011 jjeansonne *with some struggle to get sucessful registration and calls placed. Had to restart the app many times and go in/out of account setting page
pbxes pbxes.org pbxes.org NONE OK hangs at CALL_OUT_RINGING untested 0.1.8 WiFi (University of Kentucky WiFi) Pre- 2.1 Mar 2011 Ricyteach Entry for pbxes above has wrong domain entered
1und1 1und1.de sip.1und1.de STUN, stun.1und1.de REG OK 0.1.8 WIFI (DSL router, 16/1Mbps) Pre 1.4.5 Mar 2011 einalex
ekiga ekiga.net (empty) STUN, stun.1und1.de REG OK 0.1.8 WIFI (DSL router, 16/1Mbps) Pre 1.4.5 Mar 2011 einalex
SIP2SIP sip2sip.info (empty) (n/a) OK (untested) OK (sip) 0.1.8 WiFi Pixi+ 1.4.5 Feb 2011 puuj Call initiated from Google Voice via IPKall number
pbxes pbxes.com (empty) (n/a) OK (untested) OK (sip) 0.1.8 WiFi Pixi+ 1.4.5 Feb 2011 puuj Call initiated from Google Voice via IPKall number
linphone sip.linphone.org (empty) (n/a) OK (untested) (untested) 0.1.8 WIFI (DSL router, 8/1Mbps) Pre 2.1.0 Mar 2011 kostka
callcentric callcentric.com (empty) (n/a) REG_PENDING 0.1.8 WIFI (DSL router, 8/1Mbps) Pre 2.1.0 Mar 2011 kostka
pbxes pbxes.org (empty) (n/a) OK (untested) hangs at CALL_IN_INVITE 0.1.8 WIFI (DSL router, 8/1Mbps) Pre 2.1.0 Mar 2011 kostka Call initiated from Google Voice via IPKall number
SIP2SIP sip2sip.info (empty) (n/a) OK (untested) hangs at CALL_IN_INVITE 0.1.8 WIFI (DSL router, 8/1Mbps) Pre 2.1.0 Mar 2011 kostka Call initiated from Google Voice via IPKall number
sipgate sipgate.com sipgate.com NONE OK only for the first start up. After restart the APP, it shows Reg_Pending untested OK if registered 0.1.8 WiFi Pixi+ 1.4.5 April 2011 y_one2000 Call initiated from GV
messagenet sip.messagenet.it (empty) NONE OK untested OK 0.1.8 WiFi Pre- 2.1.0 June 2011 Darkmagister Call initiated from standard home phone line
onsip login.onsip.com NO NONE Reg_Failed untested untested 0.1.8 WiFi Veer July 2011 dougmany usually need auth_username, tried both and failed.
pbxes pbxes.org (empty) NONE REG_OK hangs at CALL_OUT_RINGING hangs 0.1.8 WIFI Pre (minus) 2.1.0 Sep 2011 SystemR89 Internal calls from/to internals: no audio, hang up a device doesn't hang up the other one. Can't start the app without reboot: "Waiting for the service to come up..."
8x8 (packet8) NO NO NO Requires encryption
New entry... Please copy this row so the template still exists... :-)