AstirVoIP

AstirVoIP Features :-
AstirVoIP uses SIP protocol based for signalling.
AstirVoIP Supports all codecs.
AstirVoIP run Behind NAT or private IP.
User Friendly Interface.
Auto Sync of Balance.
Real Time Sip status messages.
Call History.
Loud Speaker.
Hold.
Address book integration.
Compatible with all sip standard switches.
It has very efficient implementation of jitter buffer to play voice smoothly.
Silent suppression and comfort noise generation has been used to reduce the bandwidth usage, and this makes the dialer more efficient in making VoIP calls.
Integrates with mobiles phonebook and auto detects (+) sign when contact is picked from phonebook