Alcatel-Lucent omniacces-voip gateway Manual Do Utilizador
H.323 VoIP Gateway Voice and Convergence Features
Page 1-12
Payload Packetization and Digital Signal Processing
Payload packetization is responsible for conversion between time-continuous telephony
(analog or digital payload) at the telephony interface and Real Time Protocol (RTP) packets
on the data network interface. It supports voice compression, echo cancellation, Fax and
DTMF Relay (demodulation/modulation), modem data transport (up to 14400 baud), voice
activity detection and comfort noise generation, as well as packet arrival de-jittering.
(analog or digital payload) at the telephony interface and Real Time Protocol (RTP) packets
on the data network interface. It supports voice compression, echo cancellation, Fax and
DTMF Relay (demodulation/modulation), modem data transport (up to 14400 baud), voice
activity detection and comfort noise generation, as well as packet arrival de-jittering.
Physically, the payload packetization function is implemented on the DSPs (DIMMs), with
control and configuration on the Motorola MPC860 processor. Configuration is performed
through the
control and configuration on the Motorola MPC860 processor. Configuration is performed
through the
vsmboot.asc
file on the switch. Upon VoIP daughtercard activation, the configura-
tion is transferred from the switch to the daughtercard. See Chapter 4, “Setup and Installa-
tion,” for more information.
tion,” for more information.
The controls for voice interoperability provided by the payload packetization functions
include the following:
include the following:
•
Codecs (see also Coding Profiles -- H.323 Call Capabilities) — provides encoding/
decoding of H.323 packets.
decoding of H.323 packets.
•
Voice Echo Cancellers — reduces echo on voice conversations.
•
Fax or Modem over IP — allows fax/modem calls to be transmitted via H.323.
•
Voice Activity and Silence Detection — detects voice conversation (or lack thereof) to
reduce H.323 bandwidth requirements.
reduce H.323 bandwidth requirements.
•
Comfort Noise and Jitter Buffer — generates slight background noise (white noise) on
the voice conversation, so callers do not think the connection has failed.
the voice conversation, so callers do not think the connection has failed.
Digital Signal Processors, or DSPs as they are more commonly known, are math-intensive
coprocessors used to convert and manipulate information, especially in telecommunications
systems (systems that transmit all types of data including voice and video). They are also
programmable chips, well-suited for VoIP as DSPs have the ability not only to convert but to
compress analog signals into various digital formats, i.e., perform digital signal processing.
coprocessors used to convert and manipulate information, especially in telecommunications
systems (systems that transmit all types of data including voice and video). They are also
programmable chips, well-suited for VoIP as DSPs have the ability not only to convert but to
compress analog signals into various digital formats, i.e., perform digital signal processing.
Although DSPs do not have any direct analog input/output since they are actually digital
devices, they can accept digitized analog data rather than raw analog signals. As a result,
DSPs are used in the digital and analog VoIP daughtercards developed by Alcatel to bring
switch-enabled VoIP to enterprises; however, before the digitized and compressed voice
signals can be delivered as “voice data” in a VoIP network, they must be packetized into
H.323 packets.
devices, they can accept digitized analog data rather than raw analog signals. As a result,
DSPs are used in the digital and analog VoIP daughtercards developed by Alcatel to bring
switch-enabled VoIP to enterprises; however, before the digitized and compressed voice
signals can be delivered as “voice data” in a VoIP network, they must be packetized into
H.323 packets.
Packetized voice is digitized voice compressed into finite bit stream of IP packets, that carry
the “voice payload” between remote and distant locations, across the IP network and make
processing VoIP calls in IP networks possible. Once compressed and packetized, periodic
delays (jitter) to make the call sound smoother must be imposed on the transmission of these
packetized “voice” conversations to mimic “real time voice” (resonating by nature in continu-
ous “analog” waveform). DSPs are used further to reduce the delays from conversion and
compression to ensure quality voice communications without affecting the real time voice
processing and compression that occurs simultaneously.
the “voice payload” between remote and distant locations, across the IP network and make
processing VoIP calls in IP networks possible. Once compressed and packetized, periodic
delays (jitter) to make the call sound smoother must be imposed on the transmission of these
packetized “voice” conversations to mimic “real time voice” (resonating by nature in continu-
ous “analog” waveform). DSPs are used further to reduce the delays from conversion and
compression to ensure quality voice communications without affecting the real time voice
processing and compression that occurs simultaneously.
To transmit the compressed data (digitized voice) across the IP network, the Real Time Proto-
col (RTP) is used. RTP streamlines and then transports voice packets, including interactive
multimedia packets over IP, although it does so without any guarantees or quality of service
provisioning.
col (RTP) is used. RTP streamlines and then transports voice packets, including interactive
multimedia packets over IP, although it does so without any guarantees or quality of service
provisioning.
♦
Note
♦
H.323 VoIP telephone calls automatically receive the
highest priority in the VoIP network via the Quality of
Service ToS bit. For more information, see the switch
manual.
highest priority in the VoIP network via the Quality of
Service ToS bit. For more information, see the switch
manual.