SIP Dialer Software for PC, Mobile Phones & Web

SIP Dialer Software for PC, Mobile Phones & Web

We are looking for SIP dialer software for PC, Mobile Phone & Web with full source code, more detail are as under:

1. User friendly interface

2. Support SIP/H323 for signaling.

3. Supports G729, G723.1 and [url removed, login to view] codec for sending audio stream

4. Can run behind NAT or on private IP with VPN Tunneling embedded

5. This software will run on the phone having Symbian, Inroide, Iphone, ipad, blackburry, and windows Mobile.

6. Client can hear remaining balance by listening to IVR.

7. The ability to change VoIP provider IP and logo, (fully customizable)

8. This software should run with any kind of Softswitch, VoIP Billing Platform which support SIP/H323.

9. Must be running from the application layer. The most important part is to add G729 and G723.1 codec in it.

10. Very important part of this project is to add tunneling feature so call can place even where VOIP protocol (SIP/H323) is blocked.

11. If network is not present and user wants to use VoIP calls from provider then application will automatically dial access number of provider through GSM network to connect provider calling card or callback system.


• Connection over Wifi / 3G / Local Access or Calling Card access number

• Language support - customizable

• Native Contact Phonebook

• Dial, Hang up, Redial automatically

• Call Hold

• Call Mute

• Call Forwarding

• Call Waiting

• Speaker Phone

• Keypad Display

Call History: All, Recent, Dialed, Missed

• DTMF: Inband / Outband / SIP INFO

• Account Setup

• Brandable Client (changing some images) like themes

• Codec Support G729, AMR

• IVR number setting

• Provider can set IVR extension .

• Access number, if network not present

• Encryption Integrated

• Encryption supported for all blocked region

• Work with all standard SIP servers

• Supports all standard SIP supported switches based on RFC3261

• Remote end Dialer setting Panel

How it should work and affiliated with the server and communicate please see details below:



Registers with a voip Gatekeeper/ Softswitch by VPN Tunnel Service,


(Sends an Invite)

Request to Establish a call with another SIP phone or requests this to the sip server. Using codecs G723.1 and G729, (G711 and GSM Optional)


Sends DTMF via RFC2833 or inband


Sends Hold/Resume command

Answer Call

Accepts and invite, and establishes a call.

Reject Call

Rejects and invite


requests to refer the call to another sip address.

Play Wav File

Plays a wav file to a remote user

Stop Wav file

Stop the file being played


Invite Incoming

A call is coming in, someone invites us via SIP

Invite Answered

An invite made by me has been accepted

Invite ringing

My outgoing call is ringing the remote party

DTMF Received

a DTMF has been received

Register Success

Server accepted our registration

Remote Hold/Resume

Server accepted our hold/Resume command

Receives Sound

We are detecting a stream of sound coming through the line.


List of calls with Redialing

List of all currently open calls, each call must be an object which handles/has all calls details.

Delivery Package Contain

Control panel

To set operator code

Sip proxy

Codec Selection

Header/footer message

Balance URL

Support contact information


1 please refrains from any biding if you do not qualify for VoIP platform.

2. Must provide demo

3. We need full source code.

4. Time is important so don’t waste y& mine.

5. Must fix all bugs before delivering the entire project, Train our personnel and sign NDA with company.

6. Buyer Get all the rights of the Sour

Thanks & Regards

Skills: VoIP

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About the Employer:
( 0 reviews ) Manchester, United Kingdom

Project ID: #1102731

6 freelancers are bidding on average $5092 for this job


Thanks for detail. We have gone through all the description and are comfortable with implementation. We have submitted further detail over PMB. Thanks

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