Cisco SPA112 Administration Guide - Page 38

Adaptive Jitter Buffer, Secure Calls, Adjustable Audio Frames Per Packet, DTMF Relay - t 38

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Configuring Voice Getting Started with Voice Services 4 • Adaptive Jitter Buffer The ATA can buffer incoming voice packets to minimize the impact of variable network delays. This process is known as jitter buffering. The size of the jitter buffer adjusts to changing network conditions. The ATA has a Network Jitter Level control setting for each line of service. The jitter level determines how aggressively the ATA tries to shrink the jitter buffer over time to achieve a lower overall delay. If the jitter level is higher, it shrinks more gradually. If jitter level is lower, it shrinks more quickly. For more information, see the Network Settings section in Line 1 and Line 2 Settings (PHONE1 and PHONE2), page 90. • Secure Calls When secure calling is enabled on a line interface, a user can make an outbound call secure by entering a star code. When this feature is activated, the audio packets in both directions are encrypted. For more information, see Line 1 and Line 2 Settings (PHONE1 and PHONE2), page 90 and User Settings (User 1 and User 2), page 108. • Adjustable Audio Frames Per Packet This feature allows the user to set the number of audio frames contained in one RTP packet. Packets can be adjusted to contain from 1-10 audio frames. Increasing the number of packets decreases the bandwidth utilized, but it also increases delay and may affect voice quality. For more information see the RTP Parameters section in SIP Settings, page 63. • DTMF Relay The ATA may relay DTMF digits as out-of-band events to preserve the fidelity of the digits. This can enhance the reliability of DTMF transmission required by many IVR applications such as dial-up banking and airline information. For more information see the RTP Parameters section in SIP Settings, page 63. • Call Progress Tones The ATA has configurable call progress tones. Call progress tones are generated locally on the ATA so that an end user is advised of status (such as ringback) Parameters for each type of tone (for instance a dial tone played back to an end user) may include frequency and amplitude of each component, and cadence information. For more information, see the Call Progress Tones section in Voice System Regional Settings, page 73. • Call Progress Tone Pass Through This feature allows the user to hear the call progress tones (such as ringing) that are generated from the far-end network. Cisco SPA100 Series Phone Adapters Administration Guide 38

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Configuring Voice
Getting Started with Voice Services
Cisco SPA100 Series Phone Adapters Administration Guide
38
4
•
Adaptive Jitter Buffer
The ATA can buffer incoming voice packets to minimize the impact of
variable network delays. This process is known as jitter buffering. The size
of the jitter buffer adjusts to changing network conditions. The ATA has a
Network Jitter Level control setting for each line of service. The jitter level
determines how aggressively the ATA tries to shrink the jitter buffer over
time to achieve a lower overall delay. If the jitter level is higher, it shrinks
more gradually. If jitter level is lower, it shrinks more quickly. For more
information, see the
Network Settings
section in
Line 1 and Line 2
Settings (PHONE1 and PHONE2), page 90
.
•
Secure Calls
When secure calling is enabled on a line interface, a user can make an
outbound call secure by entering a star code. When this feature is activated,
the audio packets in both directions are encrypted. For more information,
see
Line 1 and Line 2 Settings (PHONE1 and PHONE2), page 90
and
User Settings (User 1 and User 2), page 108
.
•
Adjustable Audio Frames Per Packet
This feature allows the user to set the number of audio frames contained in
one RTP packet. Packets can be adjusted to contain from 1–10 audio
frames. Increasing the number of packets decreases the bandwidth utilized,
but it also increases delay and may affect voice quality. For more
information see the
RTP Parameters
section in
SIP Settings, page 63
.
•
DTMF Relay
The ATA may relay DTMF digits as out-of-band events to preserve the
fidelity of the digits. This can enhance the reliability of DTMF transmission
required by many IVR applications such as dial-up banking and airline
information. For more information see the
RTP Parameters
section in
SIP
Settings, page 63
.
•
Call Progress Tones
The ATA has configurable call progress tones. Call progress tones are
generated locally on the ATA so that an end user is advised of status (such
as ringback) Parameters for each type of tone (for instance a dial tone
played back to an end user) may include frequency and amplitude of each
component, and cadence information. For more information, see the
Call
Progress Tones
section in
Voice System Regional Settings, page 73
.
•
Call Progress Tone Pass Through
This feature allows the user to hear the call progress tones (such as ringing)
that are generated from the far-end network.