We are looking for a mobile dialer(Full source code) SIP
Os: Symbian S60/Symbian UIQ3/Iphone/Java/Android/windows mobiles/Blackberry
Supported services should be:
1) Voice + instant voice message chatting
3) Video Calls (optional)
1) User friendly interface
2) Mobile Dialer should support SIP/H323 for signalling.
3) Mobile Dialer supports G729, G723.1 and G.711 , GSM codec for sending audio stream
4) Mobile Dialer can run behind NAT or on private IP with VPN Tunneling embedded
5) This software will run on the phone having Symbian or Windows Mobile 5 and 6 having internet via GPRS/EDGE/Wi-Fi/3G and allother available OS and Phone models.
6) Mobile Dialer should be fully customizable.
7) Able to use Mobile Dialer with any kind of Softswitch,Voip Billing Platform which support SIP/H323.
The dialer must be runing from the application layer. The most important part is to add G729 and G723.1 codec in it.
Second important part of this project is to add tunneling feature so call can placed even where VOIP protocol (SIP/H323) is blocked.
. Multitasking background support
. Transfer and attended transfer
. Push Notifications, a reliable way to receive calls when the Softphone is closed
. Multi line
. Call waiting
. Call Grouping
. Conference calling
. Call Log
. Address Book Matching to automatically format your contacts into the proper international format
. TLS support for encrypted SIP
. Bluetooth support
. audio codec manipulation
. call recorder and player, seamlessly integrated into the call history
. simultaneous registration of multiple SIP accounts, have multiple accounts registered to receive incoming
. calls and switch the account used for outgoing calls without leaving the keypad
.Phonebook contacts integration. Easy to call anyone in your contacts via SIP
. Contacts search function, search your contacts by name or number
. add new contacts directly from the softphone
. Quickdial, your 12 favorite contacts are one touch away
. ability to generate DTMF tones while in call, to control various PBX features or automated systems (use
. audio, rfc 2833 or SIP INFO)
. speakerphone support
. detailed call history, with intelligent call grouping for an easy overview
. support for sip:username URLs in phonebook
. configurable RTP port range
. SIP Proxy support, VPN support
. STUN server support
. Call Waiting and Conferencing
. Transfer and Attended Transfer
use API and get data of user balance and other billing details
new user can creat account and get listed on switch using APi