Need a android application developer - Android Q&A, Help & Troubleshooting

Android app that serves as a SIP-GSM gateway
I need an application that runs on Android phones (4.0 or above) and can send/receive calls through SIP (Can use any sip stack like sipdroid) or IAX2 and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa.
Full description below.
Please review the attached flow diagram and confirm that you can do the job.
I wish to see a demo before accepting this work
Please review and message me with considerations. If any requirements are unable to be met, please let me know before we make a deal.
Summary
The goal of this project is to develop an Android application that can send calls through SIP and forward them to the GSM network AND receive calls from GSM and forward to SIP.
The application should then forward the audio and convert from VoIP to GSM and vice vers
General
Deliveries
The application working in APK format
Full source code
Simple manual for compiling and generating the application from source
A SIP client running on another phone that connects to the server via WIFI and able to demonstrate the functionality in this project.
Features
- Route call from SIP to GSM
- Route call from GSM to SIP (preconfigured SIP client)
- Convert audio from/to SIP and GSM networks
- Able to run on Android 4.0 and up.
Application Configurations
- Enable or disable the app
- Enable to receive calls only on wifi or all networks
- Enable to route calls from GSM to SIP (SIP address is configured through an API)
- The number to send the call should be configured through an API that is supplied with the APP
- Enable to route calls from SIP to GSM
- Must run on background
- Must be very lightweight to run on small memory devices
- Configure SIP Accounts.
Sip Requirements
- Register on Sip Proxy/Gateway
- Receive call authenticated by IP, user/pass or no authentication.
- Make calls with or without authentication
- Forward DTMF digits using RFC2833 or inband
- Use codec G711 and GSM
- Be able to use codec G729
- Receive through GSM one 1 simultaneous call and route it to a predefined SIP client
- Be able to store a CSV with all calls made
Skills required:
Android, VoIP

I join this search for a specialist.
A large budget has been allocated for this development.

Interesting idea, 2014,jesus. Good luck!

Related

[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:Optimized 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/portgo/PortGo_Pro_m.cab
Multiple Language FAQ:
http://forum.xda-developers.com/showpost.php?p=6866807&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
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?
With the iptel.org service only pure VoIP calls can be made; iptel.org does not sell minutes into PSTN.
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
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!
Advance Settings
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.
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
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.
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.
i wasnt successful to get my VoIPBuster account registering
How can i use this software with voipBuster?
hello,
works fine with french freephonie sip provider.
Is it the same than pangolin ?
Can it be skinable?
thanks :.)
Using a VoIPBuster Account
Please try these settings!
Account name : your VoipBuster username
Password : your VoipBuster password
Proxy server : sip.voipbuster.com
SIP port : 5060
Under Advance tab:-
Domain
Auth Name:your VoipBuster username or voipnumber
User Domain: (leave it Blank)
Stun and Firewall OutBond Proxy:
Stunserver (option) : stun.voipbuster.com port 3478
Outbound proxy server : leave empty
You are requested to use G.711 codec for best results
Best Regsrds
Thanks a lot MaxDan, IMHO this is the biggest step ahead since X-pro, not to many apps around. Once again thanks for this initiative
MaxDan said:
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.
Click to expand...
Click to collapse
so what else am I supposed to install on my IMate-8502 other than this software?
I have a SIP account from poivy.com.
Thanks
Handfree
This software works grate with my sip account .
Just 3 requests I have for next version if they are possible :
1) Use windows contacts to dial from the windows phonebook with our sip account.
2) Is it possible for this program to have an option to transfer the voice to bluetooth headset when there is a one connected to the phone ?
3) I also request for multiple sip accounts at the same time. The reason is my sip provider have a different proxy for incoming call and another one for outgoing with 2 different port numbers. Right now I can make calls no problem but I can't receive any calls because of the above reason .
If u can add several or at least 2 proxy at the same time, this software will be my Sipphone for ever.
Tnx
Thunder
hi Tempest69,
Is it the same than pangolin ?
Yes,It's same than pangolin. But pangolin is run at windows 2000/xp/vista, Pangolin is freeware too.
Can it be skinable?
We only support Customization Softphone for company current.
Freeware isn't support skinable.
hi thunder6290.
1.Use windows contact
we will add it at next version(Beta2).
2.bluetooth headset
3.multiple sip accounts
I has add it to todo, but it will add to PortSIP CE latter. maybe Beta3 or Beta4.
I has update todo, you can find it at Page1.
Anyone tried this with gizmo5? If so what settings did you use?
finally a newish voip software for WM6 and it's free.
Will try it out tonight after I get home from work.

[Q] Android SIP server (Gateway)

Dear All
how to make Android work as SIP server the Q like this:
SIP application on Android register in SIP gateway (like softswitch), then softswitch
send SIP request to android to make call from the android using the GSM SIM card.
how it can be done or there any app like this

call forwarding to voip (sip/skype/viber)

hey there,
is it somehow possible to forward an incoming call to voip - if sip, skype, viber or anything else doesnt matter to begin with ...
or might there be an app around which i didnt find yet?
thanks
fm40 said:
hey there,
is it somehow possible to forward an incoming call to voip - if sip, skype, viber or anything else doesnt matter to begin with ...
or might there be an app around which i didnt find yet?
thanks
Click to expand...
Click to collapse
Nope, doesnt work, try finding an app based on call forward.... hit the "Thanks" would be appreciated !
Pseudo forwarding to SIP/VOIP
Well it is very much possible requires double hopinng but it is possible!
Get a rebel.com account and register your forwarding phone number, rebtel calls you press 1.
Get a SIP provider who supports inum (I would recomend callcentric but there are plenty others as well). Get a sip/inum did.
In rebtel add a contact in your local number zone which is your sip inum or your Skype name, should you fancy M$oft proprietary standards.
Forward mobile to the rebtel local number. This should cost you local airtime minutes, or local rate if not under a monthly plan.
Voila, calls forwarded to your sip/skype Voip device, which you can pick from any place on earth without roaming fees.
Hmm did I mentioned that on the voip sphere everything is absolutely FREE!
Callcentric calling features also include, call treatments, virtual fax, voicemail and further forwarding with lowcost high quality sip rates.
Happy forwarding
Viber sip gateway
Viber SIP gateway
Make More Revenue with Viber SIP gateway!!
• Viber SIP Gateway works as proxy server between the originating switch and the terminating number.
• Viber SIP Gateway Filters the dialled in numbers for the availability in viber and then forwards the call to viber.
• In case of non-availability it sends back the call to the originating IP.
• provides 5+ ACD
• Live DEMO available on request
For further information please contact [email protected]
Skype: mobile.sip

[Q] Can I route a call via SIP based on a phone number using native SIP android?

My Android 4.x phone supports SIP.
However, the options when to use the "Internet calling" are very limited: "For all calls", "Only for Internet calls", "Ask for each call".
I would like to use Internet (SIP) calls based on phones number (e.g. all international and long-distance calls should go via SIP, all phone number starting with +395656 xxx and local calls should go via my regular voice account).
I don't want to use any application like csipsimple, I want to use native support.

[Q] Android App Suggestion Needed : Communications Hub

Hello
I will be travelling to another country, however I wish to keep my (home country) current mobile connection (number) active since a lot of local (banking etc ) services and sms alert notifications are received on my home country mobile number.
I plan to keep my mobile switched on and hooked to the charger to let it run :
Hence I am looking for an Android App to :
1. Monitor the incoming calls and selectively based on rules either decide to offer a IVR recording to leave the message (recorded on the SD card in wav or amr format) OR forward the call to another predefined number (presumably my cell number in the visited country). The caller wont have to pay the charges for international calling, since the outgoing call from the cell is originated and the caller and receiver are connected as if being on a conference call.
2. Similarly monitor the incoming SMS and based on rules either store locally or forward to predefined international number.
3. For point number 1, an option to upload the recorded voice messages to an online service like google drive or dropbox, with the filename containing the caller number/name and date/time of the call, and delete from the phone after a successful upload. This way I can retrieve the voice messages via internet instead of using the mobile operator voice recording service.
Back in the days on my Nokia N95 there used to be an excellent symbian application which was able to do most as described above and more, I am looking for a similar type pf application on android.
I have seen that an app exist https://play.google.com/store/apps/details?id=fahrbot.apps.blacklist&hl=en but it is more focused on the security and blocking aspect .
I am looking for an app which let you run a mini PABX on the mobile to offer similar functionality.
Does anything like this exists...Need your guidance and suggestion to find the correct app.
P.S :
The company making the symbian app, is maybe out of business but you can read about this excellent app (Interactive Voice Call Master) over here http://my-symbian.com/s60v3/software/applications.php?faq=1&fldAuto=156
Thanks.

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