9. MV Softphone

MV-SoftPhone

SIP Communication Center

The combination of IP phone, video, IM and presence inspires users to communicate and enjoy the benefits VoIP communications.

MV SIP communication Center is a complete java based PC software package that addresses residential users, corporate users and Telecom Infrastructure vendors.

IP telephony is a strong complement and substiute to traditional telephony.

IP video telephony enables the users to view the called party – person to person.

Advanced features
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
Attended Redirect support

Caller ID and Caller ID block

Caller ID and Caller ID block

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

Easy branded GUI.

Modern user interface.

Easy Localizable messages (English, Spanish, Hebrew and Russian includ

Video codecs information:

	H.263+	H.263	H.261	H.264
CPU consumption	High	Average	Low	Very High
Compress quality	High	Average	Reasonable	Very High
Picture quality	High	Average	Reasonable	Very High