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Bandwidth Management (Prioritization)
The SatVox system sits upstream from the local network, and makes sure that LAN traffic such as FTP, web browsing, email etc. does not interfere with the placing, receiving or conducting of a telephone call.
Network Address Translation (NAT)
A static public IP address is required at both the remote site and the host site, although the SatVox system functions normally behind a firewall or NAT router and does not need its own public IP address.
Digital Signal Processing (DSP)
Special microprocessors designed to execute repetitive math-intensive algorithms, DSPs are 2 -3 times faster than general purpose processors. High quality VoIP requires extreemly fast processing, therefore a hardware based approach using digital signal processing allows a much higher voice quality to be achieved as compared to a system that operates on PC.
Echo Cancellation
Provides the effective removal of hybrid and acoustic echo inherent within digital voice transmission. Echo Cancellation reduces background noise and removes hybrid and acoustic echo before any transcoder processing. With Echo Cancellation enabled, the overall efficiency of the coding is enhanced, significantly improving the quality of voice.
Voice Compression
Typical voice telephone calls consume 64 kbps of bandwidth. SatVox uses standards based H.323v2/v3/v4 to compress speech to a point where high quality voice can be transmitted using as little as 10 kbps.
Silence Suppression
SatVox adds "white noise" during periods of time during a VoIP conversation where no one is speaking, to give the digital call familiar analog charecteristics. It removes the "are you still there?" dead air from the call.
Call Progress Tone Detection (CPTD)
Call progress tone signals provide information regarding the status or progress of a call to customers and connected equipment. These audible tones are transmitted over the voice path within the frequency limits of the voice band including dial tone, busy tone, audible ringback and reorder tone (fast busy).
Packet / Jitter Processing
Provides a buffer to resolve inconsistant delivery of IP voice packets to improve the quality of a digital telephone call. SatVox Jitter Control can compensate for up to 15% packet loss.
General
SatVox brings to the satellite market carrier class and subscriber VoIP Gateways, IP Telephones, and embedded Gatekeeper software. By utilizing industry standard protocols such as SIP, H.323 and MGCP, SatVox is interoperable with other VoIP products and is designed to seamlessly integrate with customer's existing phones and phone systems.
To ensure the highest possible voice quality, the SatVox product line has incorporated 16 levels of programming to dynamically preserve required satellite bandwidth and provide precise control over voice associated resources. SatVox offers scalability by allowing the flexibility to stack gateways and add ports as you grow. This scalability feature also provides the ability to maintain one IP address and deliver features such as line hunting, caller-id, auto fax detection, and remote programming updates.
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