Comrex VH2, Руководство по эксплуатации продукта

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G.729a
Because G.711 is a bit old and primitive, an encoder has been developed to deliver equivalent audio quality while using a
fraction of the network bandwidth. G.729a implements a more aggressive compression algorithm, resulting in network usage
of around 8 Kb/s per channel, or about 1/8th the data of G.711. This can be very helpful for avoiding excessive network
congestion. Of course, equivalent audio means the same limited fidelity as G.711.
This encoder is sometimes simply referred to as G.729 (without the a), but is equivalent to the user. Another variant, G.729ab,
is sometimes available that can detect when voice is present and squelch the data stream during periods of silence, further
conserving network bandwidth. Comrex STAC VIP supports G.729a.
G.722
Familiar to ISDN broadcasters, G.722 is an encoder designed to increase the audio fidelity of phone calls. Using the same
network bandwidth as G.711 (64 Kb/s each way), G.722 more than doubles the audio spectrum conveyed by the call, making
the caller sound much more natural and identifiable. The 7 KHz spectrum carried by G.722 covers the majority of human
voice energy, excluding only the most sibilant sounds in speech.
G.722 is the most common encoder for calls that are classified as “HD Voice” in the VoIP world. All Comrex codecs and VoIP
devices support G.722.
Opus
Efforts are increasing at combining the worlds of VoIP and web services. Many web audio services have standardized on
Opus, an encoder that delivers near-CD quality audio with low delay. As these efforts continue, users can expect to find more
support for the Opus codec in VoIP devices and networks. All Comrex codecs and the STAC VIP phone system support Opus.
Other encoders
A large spectrum of VoIP-ready encoders have been introduced in the past decades, each having proponents and particular
advantages for certain applications. These include iLBC, iSAC, G.722.1, G.722.2, G.726, VMR-WB, SILK and AMR-WB+.
For the most part, we expect the industry to support only the four encoders outlined above in most equipment and networks.
0
50 15 7 3 300
Hz kHz kHz kHz Hz
FM Audio Bandwidth
G.722 (HD Voice)
Telephone
20
kHz 10
kHz
OPUS
Session Initialization Protocol
The piece that ties RTP sessions and encoders together, and gives VoIP its telephone-like qualities, is another completely
separate connection between devices called the SIP. You’ll see the term SIP thrown around in place of VoIP in many places
(SIP Phones, SIP PBXs). It’s a very powerful specification and is being used for an increasing number of applications besides
VoIP, like compatibility standards between broadcast IP hardware codecs, studio-style AoIP installations, and real-time web
audio and video. It’s becoming such a vital element of so much new technology, it’s a very valuable thing to be expert in.