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[APP][UPDATE Aug 23,2010] PortGo(PortSIP CE) VoIP/SIP free softphone

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MaxDan
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(Last edited by MaxDan; 22nd August 2010 at 06:31 PM.) Reason: Aug 23,2010 update
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Lightbulb [APP][UPDATE Aug 23,2010] PortGo(PortSIP CE) VoIP/SIP free softphone

PortGo for Windows Mobile 6/6.5 V1.3
PortGo is the newest free softphone application from PortSIP, it's built base on PortSIP VoIP SDK, allowing users to enjoy multimedia communications in a dynamic way.

PortGo works seamlessly with your internet connection you can chat away with free calls and never worry about cost, time or distance. It including great features to help you stay in touch with friends, family and co-workers, share your thoughts and views and find the information you need. You can use it on your PDA

PortGo is a VOIP softphone that running on Windows Mobile 6.0/6.5, allows you to speak with any other softphone , any stand-alone IP-phone, or using Internet Telephony Service Provider (ITSP) with any traditional wired or mobile phone. It supports SIP and is fully inter-operable with most major VOIP vendors and ITSPs.

Standard Features
Support platforms: Windows Mobile 6/6.5
Support Device Screen: QVGA(320*240),VGA(640*480),WVGA(800*480)
Support servers: Cisco CallManager, Open SER, SER, Asterisk, Portaone, Radvision, Nortel, Rainbow, Avaya and other SIP Platforms.

Switch between earpiece and load-speaker(Does no support some devices, likes Samsung I708)
Audio call: G.711 aLaw/uLaw, GSM
Call transfer: Blind transfer
Call hold, mute speaker, mute microphone.
Audio record: record audio as wave file.
DTMF support: Send DTMF tone(RFC2833 and SIP INFO method), detect DTMF tone(RFC2833 and SIP INFO method).
Silent Ringtone
Audio conference
Acoustic Echo Cancellation
Automatic gain control
Comfort Noise Generation
Voice Activity Detector
STUN support
Bluetooth headset support
Outbound proxy server support
Jitter buffer
Customization SKIN,Default support:CIF(240*320),HVGA(240*400),WVGA(480*800)
Multiple Language support.User can add new language support.
Freeware

Todo:
1.Audio conference.
2.IM Support: SIMPLE(Presence, Subscribe, Pager message).
3.Support video call.
4.Audio record: record audio as MP3 file.
5.Support TLS/SRTP.
6.Customization Ringtone
7.Multiple sip accounts


Attention!
Having PortSIP installed is not enough to talk except over your local network. You have to obtain an account with any ITSP like Vonage or SipIptel.org, or install your own VoIP server.


If you have any advice, please let us know, we are happy to listen to.

Change log:
AUG 22, 2010 Version 1.3.8.18
1: Added set ptime options.
2: Added Acoustic Echo Cancellation(AEC),Noise Suppressor(ANS) options.
3: Support 2 line, audio conference.
4: Added MessCall notify
5: Added auto connect to Internet.
6: Add HVGA(240*400) skin.
7: Optimized call to PSTN sound quality.
8: Fixed some BUG.


Jun 17,2010
1: Reduce the voice delay.
2: Customization SKIN,Default support:CIF(240*320),WVGA(480*800).
3: Multiple Language support.User can add new language support by self.
4: Added Display name Multiple Language support.
5: change soft name to PortGo.
6: Support receive incoming call on Power suspend state.
7: Remove audio codec G.729,G723.1(Because the G.729 and G.723.1 codec is not free)
8: Fixed dial prefix BUG.
9: Agent name change to :"PortGo for Mobile"


Jul 8, 2009
1:eek:ptimized sound quality.
2:Update SIP protocol stack.
3:Add button to remove account.
4:The number on the contact list auto Remove ()-, and space.
5:Fixed call out has the early media,the ringing sound comes through speaker,Now change the ringing sound comes through earpiece.

Jun 24,2009
1.When callnum is null, click call button will open "Call History".
2.The number on the contact list auto Remove ()- and space.
3: Set default port to 5060.
4.Disable "Auto screen off " .
5.Fixed "The application exit,but ringing tone continues" BUG.
6.Order account by last of user access.
7.Multiple tone files for the DTMF key presses.
8.Added show call failure message.

Jun 15,2009
1.Add windows contact support.
2.Disable Suspend when call is active(quickly drain the batteries).
3.Auto screen off when call is active.
4.Dial prefix support.
5.Command line support.
6.Long press(more then 2sec) of "0" changing to "+", when call num is empty.
7.Check for update.
8.Agent name change to :"PortSIP CE for Mobile"
9.reconnect or auto reconnect
10.fix some bugs.

May 20,2009
1.Add support DisplayName.
2.Fix AuthName bug.
3.Smaller memory requirements

April 24,2009
PortSIP CE Beta1 release

Download:
PortGo for Windows Mobile (Freeware)
http://www.portsip.com/downloads/portgo/PortGo_m.cab

PortGo Pro for Windows Mobile(15 days trial )
http://www.portsip.com/downloads/por...rtGo_Pro_m.cab

Multiple Language FAQ:
http://forum.xda-developers.com/show...&postcount=364

last updated: Jun 17,2010

Support Device List:
HTC Touch Diamond
HTC Touch Diamond2
HTC Touch HD
HTC Touch HD2
HTC Touch 3G
HTC TyTN II
HTC Touch Diamond 2(looks kinda fuzzy)
HTC sprint mogul/titan 6800
Samsung Omnia i900



If PortGo can support/nonsupport you device, please tell to me, Help create support list,thank you!
Report Type :
Device Name | OS version | Support | False Reasons

sample:
HTC Touch Diamond WM6.1 YES
O2 Atom Exec WM5 NO Earpiece false
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pureheart
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Default Settings



Hi I'm trying to use this software...Could you please help me with the settings?
I've created an account with iptel.org but can i use this account to make calls for free?
how does it work??
which proxy server and port do i have to use on the settings?
 
MaxDan
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With the iptel.org service only pure VoIP calls can be made; iptel.org does not sell minutes into PSTN.
 
kipciaan
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Exclamation Sleeping while working !

Dear MaxDan,

Thank you very much for the time that you spent to find this and sharing with us !

i have tested this on TyTN ii and it succeeded ! I think this softphone is nice than the AgePhone mobile2, it has more codecs including G729 in a freeware, which are not available in AgePhone mobile2 after ($35).

There are few small bugs, which could be taken care in next update:

*It forgets the toggle setting for Earpiece / Speaker (some times toggles by itself)

*It also have a sleeping habit, when the screen goes off or the phone goes into standby mode, you can't hear your call, but its working in background.


Rest i would like to say its great software till date...

Thanks & Best Regards

Kip
 
MaxDan
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Dear kipciaan,
Thank you for the use of PortSIP CE.
Question 1:
*It forgets the toggle setting for Earpiece / Speaker (some times toggles by itself)
Did you mean when selected "Use Earpiece", the phone still play ring tone on speaker when the call is incoming? This is our design.

*It also have a sleeping habit, when the screen goes off or the phone goes into standby mode, you can't hear your call, but its working in background.


This is a trouble, Because i can't disable sleep(Power will shortage), and i don't know connect to internet use WIFI or 3G or EDGE, which device must be activating. We will choose a compromise solution to fix this BUG at last version.

Best Regards!
 
kipciaan
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Dear MaxDan,

thanks for your reply!

* it toggle by it self :

when i generate CALL selecting "USE EARPIECE" mostly the sound comes from the EARPIECE but sometimes from SPEAKER (at back) on TyTN ii.

i think may be the file used for audio routing delay to execute or something... ('m not a technical pro_ )

there must be solution for the backlight and to hold the softphone running without shut off.

in the Menu i see the "OPTIONS" are hidden, don't know why?

*Please change the "History" button to "Contacts/PhoneBook" making others life simple!

#the "CALL History" can be viewed by going into menu.

waiting for the latest updates soon.

Best Regards.
 
yo3gjc
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Finally, a SIP tool. Thank you so much! Tow further enhacements:
-if possible in the future to allow configuration of a dial prefix, some VOIP providers require e.g 001 +phone number, if selected from Addressbook 001 is missing, so a possiblity to dial automatically 001+ addressbook number (or any other prefix) is welcome
-also multiple SIP accounts if possible, to switch between providers,
-how can I replace beep when a number is dialed with a DTMF tone

Thanks for the nice weekend present, and congrats for this very usefull tool
Service Provider: Fido (Canada) IMEIUnlocked
Phone/Model: Samsung Galaxy S IV (SGH-i337M)
Build Version: JDQ39.I337MVLUAMDJ
Modem: I337MVLUAMDJ
AWS Support: Flashed i337_AWS.QCN with QPST

Other phones:
IPhone biiig break (dark side)
HTC 8925 Ultimate-X.V9 build 21054+HardSPL3.34+ Radio:1.71.09.01
HTC Blackstone - Laurentius26 THDV3.55 Radio 1.13
 
MaxDan
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dear kipciaan,
* it toggle by it self
I will test this problem later.

there must be solution for the backlight and to hold the softphone running without shut off.
This BUG will fix shall precede all other.

the Menu "OPTIONS"
This is a reserved Menu. To implemented some specifically function.

"Contacts/PhoneBook"
We have to consider use windows mobile system contacts or PortSIP CE program self contacts. when implemented contacts,will change "History" button to "PhoneBook"

Best Regards.
 
MaxDan
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Dear yo3gjc,
if possible in the future to allow configuration of a dial prefix, some VOIP providers require e.g 001 +phone number, if selected from Addressbook 001 is missing, so a possiblity to dial automatically 001+ addressbook number (or any other prefix) is welcome
We can add dial prefix future, setting at "Options" menu.

-also multiple SIP accounts if possible, to switch between providers
You can login out to change SIP accounts. We don't support use multiple SIP account at the same time.

how can I replace beep when a number is dialed with a DTMF tone
You can replace audio file at sound directory, 8000HZ mono 16bit wav.
 
lover
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i wasnt successful to get my VoIPBuster account registering
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