Application:Linphone
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
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.
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... :-) |