Video – supported codecs: H.261, H263, H263+, H264

Call Recording – records your incoming and outgoing calls

Built-In Voice Mail – sends calls o the voicemail on predefined situations (no answer, unconditional, busy or user select)

Multiple Conferencing – combines up to 10 connections simultaneously

QoS – configures Media and Signaling TOS/DiffSrv values

SRTP – Encrypted RTP streams for secure communication

Internal Address Book

Call Logs – Incoming, Outgoing and Missed

Multiple Lines – supports up to 10 concurrent incoming and outgoing calls

Multiple SIP Accounts – supports multiple SIP Provider definitions

Built-In NAT Solution – Supports STUN, TURN, ICE, Comedia

IM and Presence – enables instant connection from your buddy list

Call Control Features (SIP)

• Fully compatible with SIP (RFC 3261)
• Incoming and Outgoing calls
• Multiple SIP Proxy server support
• Registration support
• Attended Redirect support
• Caller ID and Caller ID Block
• Hold

• Proxy (Local/Remote) Authentication Support
• URL dialing
• IP-to-IP Calls
• Transfer
• Multiple Line Conference
• Forward incoming calls
• Click-2-Dial from any web page

Sound Control (Sound Card)

• Selection of the audio device used by the softphone
• Headset support
• USB headset devices support
• Volume and Gain control
• Mute and speaker
• Local signalization files: ringing-tone, ring back- tone, Busy tone, DTMF tones

Media Control

SRTP: Secure Real Time Protocol

Provides confidentiality to the media stream

Audio Codecs supported:

• G.711 PCMU/PCMA
• GSM-FR
• G.729a/b
• G.723.1
• iLBC
• Speex

Call Control Features (SIP):

• Fully complies with SIP (RFC 3261)
  
• Incoming and Outgoing calls
  
• Multiple SIP Proxy server support
  
• Registration support
  
• Attended Redirect support
  
• Caller ID and Caller ID block
  
• Hold
  
• Proxy (Local/Remote) Authentication Support.
  
• URL dialing
  
• IP-to-IP Calls
  
• Transfer
  
• Multiple line Conference
  
• Forward incoming calls
  
• Clik-2-Dial from any web page

Sound Control (Sound Card)

• Selection of the audio device used by the soft-phone
  
• Headset support
  
• USB Headset devices support
  
• Volume and Gain control
  
• Mute and Speaker
  
• Local signalization files: Ringing-tone, Ring back-tone, Busy-tone, DTMF-tones

User Interface

• Easy branded GUI.
  
• Modern user interface.
  
• Easy Localizable messages (English, Spanish, Hebrew and Russian included.

Minimum PC Requirements:

• Pentium III 1GHz PC

• Windows 2000, XP

• 25 MB available disk space

• 64 MB memory for Windows 2000

• 128 MB memory for windows XP

• Available USB port for video device

Standards & RFCs

Signaling

(RFC3261) SIP

(RFC2976) SIP INFO Method

(RFC3262) Reliability of Provisional Responses

(RFC3263) Locating SIP Servers

(RFC3264) An Offer answer Model with SDP

(RFC3265) SIP-Specific Event Notification

(RFC3311) UPDATE

(RFC3361)DHCP support for SIP Servers

(RFC3428) SIP Extension for IM

(RFC3515) SIP REFER Method

(RFC3581) Symmetric Response Routing

(RFC3725) Best Current Practices for 3PCC

(RFC3842)Message Waiting Indication

Media

(RFC3550, RFC3551) RTP/RTCP

(RFC3711) Secure RTP

(RFC2327) SDP

(RFC2833)RTP Payload for DTMF digits