Multi-threaded, server-side summing/mixing engine with integrated SIP support.
Client-side control panel for non-blocking, multi-channel/multi-access professional communications.
Software interface to easily bridge disparate communication systems including hardware intercoms, two-way radios, and public/private telephone networks.
Administrative GUI control for dynamic configuration and monitoring of VCOM from any workstation on the network.
Full multi channel/multi access, non-blocking professional communications functionality
Point to Point (private channel between any two operators)
Group Call (one caller to many listeners), and Party Line (group conference)
Client Control Panel Selectors (keys) can be configured for Talk, Listen, or Talk/Listen
Selectors can be programmed for latch mode, momentary mode, or both
Dynamic “Answer Back” selectors indicate incoming calls from clients for whom a
dedicated key is not programmed
IFB function serves as a voice cueing mechanism for on-air talent
ISO allows for private communications between users
Patching/Dispatch function enables users to add/remove 4-wire or SIP ports from
Party Lines from Client
LDAP (Lightweight Directory Access Protocol) integration to provide Active Directory
Voice Activity indicator causes Control Panel selector name to modulate when audio is
detected on a given channel
Silence Suppression stops the transmission of audio after a period of no detected voice
activity
Support Codec G711, G722, Speex Wideband; MPEG-4 HE AACv2; MPEG Layer 2, Layer 3
AES (Advanced Encryption Standard) to encrypt audio to prevent network snooping
CDR (Call Data Records) for generating a report to determine who has the ability to
access the party lines (aka hoots)
QOS (Quality of Service) to flag all audio packets to and from the client using IP
differentiated services allowing for packet prioritization when traversing a congested
network
Client Control Panels can be oriented horizontal or vertical
Individual selector volumes can be adjusted by +/- 18dB in 6dB increments
System wide automatic microphone gain control provides unity audio levels
Dual audio output capability enables monitoring on both headset and speakers
simultaneously
Device Interface supports connection with two way radios, hardware based
communications
systems, or public/private telephone networks via standard hardware interfaces
Integrated SIP server allows connection of SIP enabled VoIP peripherals directly into the
VCOM Virtual Matrix
Support for integrated multi-line Phone Interface using “off-the-shelf’ multi-line PCI card
to connect to POTS lines