ASTRI

Session Initiation Protocol (SIP) is a call control protocol that enables Voice over IP (VoIP) applications of the next generation. ASTRI offers a series of VoIP-related solutions based on this technology, including a desktop SIP Phone, WiFi Phone, video IP phone, WiFi/Cellular handoff solution, etc. The cores of these products are based on a SIP platform, one of the fundamental technologies offered in ASTRI.

The SIP Development Platform offers a complete solution to build VoIP client applications. It offers two levels of APIs, at the SIP Stack's level and the User Agent's level, for application development on the Linux platform. The following list presents some typical call features enabled by this platform.

•	Proxy Numeric Call
•	Direct IP Call
•	Call Hold
•	Call Waiting
•	Call Transfer
•	Client-Based Call Forward

The key to success for vendors and manufacturers of IP telephony devices is time to market. ASTRI's SIP Development Platform is the perfect suite for rapid prototyping and short development cycle of such products. This platform is available for licensing in the form of a well-documented development and testing toolkit.

Specifications

SIP-related Protocols

•	SIP (RFC 3261)
•	PRACK (RFC 3262)
•	Locating SIP Servers (RFC 3263)
•	Auto Media Negotiation (RFC 3264)
•	Subscribe/Notify (RFC 3265)
•	Digest Authentication (INVITE)
•	SIP Alert-Info
•	RTP Payload for DTMF (RFC 2833)

•	SIP INFO Method (RFC 2976)
•	SIP Reason Header (RFC 3326)
•	SIP Refer Method (RFC 3515)
•	Rport (RFC 3581)
•	MWI (RFC 3842)
•	SIP 'Replaces' Header (RFC 3891)
•	SIP Referred-By Mechanism (RFC 3892)

Other Relevant Protocols

•	SDP
•	RTP

•	NTP
•	STUN