Home' RTCA Documents for Review : DO-343B Contents 67
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4.2.33 V60 - Voice Grade of Service
Verification technique: OT and A
1. Call Data Records from both Inmarsat and CNPs will be processed to
determine the GOS during operational trials.
2. An analysis will be carried out to show that the 2012 voice traffic carried over
Inmarsat’s Classic Aero network can be carried over the SBB network. The
analysis will include spectrum, channel unit, voice codec and voice switching
systems. The analysis will also determine approximately how much additional
capacity is available both in terms of what is currently operational and in terms
of growth capacity.
4.2.34 V70 - Voice Quality
Verification technique: BT and A
The overwhelming contributor to voice quality degradation is the low-bitrate speech
codec employed on the link between the AES and Inmarsat’s terrestrial earth station.
Therefore it will be demonstrated that the speech codec algorithm employed meets the
requirement when DRT tested with trained human listeners in accordance with
ANSI/ASA S3.2-2009 and in realistic over-the-air bit error rate conditions. This will be
demonstrated separately for each of the codec types in use for circuit-switched and
packet-switched (VoIP) speech services.
In addition, it will be shown by analysis that further significant speech quality
degradation does not occur between the speech encoder/decoder at Inmarsat’s
terrestrial earth station and the boundary of the CNP’s network, for example by showing
that the G.711 42 standard is adhered to throughout this segment without additional
4.2.35 V80 - Voice Latency
Verification technique: BT
Simultaneous acoustic recordings will be made of speech-like sounds transmitted
between the handset of an AES and a second audio device (e.g. a handset or
microphone/loudspeaker connected to an E1/T1 break-out tester) connected to the
terrestrial interface under test. These tests will be run with an AES in the laboratory
but operating over the satellite. Alternatively, this test may be conducted remotely from
the terrestrial interface under test if the latency of the intervening link is known or, if this
latency is not known, the requirement is met anyway (including this unknown latency).
For convenience, the AES will be co-located with this second audio device for this test.
Using these recordings, the relative timing of occurrence of particular sounds at the
sending and receiving ends of the call will be measured, and hence the latency
determined. The average latencies of at least ten test calls in each direction will be
4.2.36 V90 - Voice Availability
Verification technique: OT
Voice availability will be verified in the same manner as data availability which is
described in Section 4.2.13.
4.2.37 V100 - Voice Number Unplanned Outages
4.2.38 V110 - Voice Unplanned Notification Time
4.2.39 V120 - Voice Addressing
Verification technique: OT
This requirement will be verified by observing that voice calls are delivered to/from the
18 G711 is the most commonly found voice companding system in voice trunk networks and is an ITU standard.
G.711 uses a sampling rate of 8,000 samples per second, and uses non-uniform (logarithmic) quantization with 8
bits used to represent each sample, resulting in a 64 kbit/s bit rate. G711 delivers toll quality voice.
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