Alcatel-Lucent omniacces-voip gateway Manuel D’Utilisation

 

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Voice packet transmissions, or the “payload,” are expedited by engaging the User Datagram 
Protocol (UDP) for faster delivery, packets which by necessity include the IP network call 
transport header information. Resultant jitter caused by delays imposed on the payload pack-
ets upon arrival to their destinations is also handled by the DSPs.

 

 

UDP is needed by RTP to keep pace with “Real Time Voice” but lacks controls and error 
checking capability.

DSPs can monitor calls in progress, detect voice activity and handle echo cancellation (the 
filtering of unwanted transmission signals as specified in ITU algorithm standards G.160 and 
G.126); comfort level background (white) noise can also be generated on either the transmit-
ting or receiving end.

Since digital signal processing affects nearly every operation in VoIP, numerous DSPs are 
incorporated adjacent to the supporting MPC860 CPU signaling controller in the voice switch-
ing daughtercards (normally used with voice switching modules), comprising the core of 
Alcatel’s enterprise VoIP on the call processing end. The DSPs and the Motorola MPC860 
controlling processor work in unison to support the various protocols and interfaces that 
implement the enterprise VoIP telephony functions contained in software on the voice switch-
ing daughtercards. In a nutshell, the DSPs are the voice processors, and the MPC860 control-
ler is the data communications processor on the daughtercards. Altogether, the above 
components provide T1, E1 and ISDN voice and data synthesis processing, with scalable 
versions of each bringing enterprises any-to-any switching functionality that now, with enter-
prise VoIP, includes least-cost call routing for VoIP Virtual Private Networks (VPNs).

 

Initially, digital signal processing involves DSP detection of an array of voice signaling types 
using Channel Associated Signaling (CAS) repetitive circuit-state signaling protocols (for T1 
and E1 lines). Many forms of call signaling exist to set up and end calls, most of which result 
in the ringing of a phone or connection of a fax machine. These forms entail newer line 
signaling methods that use digital pulses (PCM, or Pulse Code Modulation), analog touch-
tones such as DTMF (Dual Tone Multiple Frequency), and other much older analog signals in 
all their assortments, including but not limited to: Ear & Mouth (E&M), Loop Start, Ground 
Start, Foreign Exchange Subscriber (FXS) and Wink Start. Each signaling method was devel-
oped through the years by the telephone industry to provide Plain Old Telephone Service 
(POTS).

E&M signaling, of which there are five interface types, is the most widely used method for 
connecting calls to PBXs, telephone switching systems which use channelized T1 or E1 lines 
to transmit signals and multiplex digitized voice. T1 robbed bit signaling is an example of 
narrow or in-band signaling — where signaling tones are passed along the same circuit as 
someone’s voice.

ISDN (Integrated Services Digital Network), on the other hand, is another type of signaling 
wherein voice transmissions are digitized then placed on separate broad or out-of-band chan-
nels (so signaling tones are not passed along the same circuit as someone’s voice). This 
prevents signaling or other intrusions into the calls, and usually provides faster transmission.

ISDN is a common protocol in the Common Channel Signaling (CCS) network architecture 
used for exchanging information between out-of-band signaling networks and telecommuni-