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Introduction
to TCLw and Echo Response |
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| Echo in VoIP
telephone calls |
Echo is
an undesirable characteristic of a telephone call where the user of
a telephone hears a copy of the sound transmitted by their own telephone
that is delayed in time. Echo is a greater concern for VoIP telephones
than it was in analog telephony. This section provides an introduction
to echo and how it is generated in a telephone call.
When sound is applied to the handset transmitter of a VoIP telephone,
it is digitized, processed, encoded and transmitted from the network
port of the telephone as an IP packet. In addition to being sent to
the network, the sound appears at the receiver of the same telephone.
There can exist several different paths for the transmitted sound
to reach the telephone’s receiver. Some of these paths are intentional
and desirable, while others are not desired and produce objectionable
signal components. The sound that is heard in the receiver is the
composite of the send signal being returned to the receiver by the
various paths. Each path may have a different frequency response and
delay characteristic.
The following figure shows a simple VoIP telephone call between two
telephones. This example shows some of the possible signal paths for
the send signal to be returned to the receiver of the same telephone. |
Figure 1: Simple VoIP call
between two telephones
(Click to enlarge) |
Direct acoustic coupling exists through the plastic and cavities of
the near-end handset itself. In addition, the near-end telephone intentionally
sends some of the electrical signal from the transmitter to the receiver.
These signal components make up the Sidetone of the telephone. Sidetone
is desirable because the user perceives that the telephone is active.
It is natural for the talker to hear their own voice when speaking.
The sidetone simulates this.
The sidetone signal has only a small amount of delay because it is
generated locally in the telephone. The delay of the sidetone is too
short to be perceived by the user of the telephone.
For the simple VoIP connection between two telephones shown above,
transmit and receive data travel in separate packets on the IP network.
There is no possibility of echo being generated until the transmit
packets arrive at the far-end telephone.
When the transmit packets reach the far-end telephone, they are decoded
and converted to an electrical signal that is used to drive the receiver
of the far-end telephone. Electrical crosstalk in the far-end telephone
couples some of the receive signal back into the transmit path. In
some cases, electrical crosstalk may also occur in the handset cord.
These signal paths can generally be avoided by good circuit design
and layout techniques.
The receiver in the handset of the of the far-end telephone generates
an acoustic output that is coupled back to the transmitter. This coupling
can occur through the air, and also through the handset itself.
The combined signal from both the electrical and acoustic paths of
the far-end telephone is digitized, encoded, and transmitted back
to the near-end telephone where it appears at the receiver. These
components of the received signal have longer delays than those generated
in the near-end telephone because they have passed through A to D
and D to A converters, encoders / decoders, and the IP network.
When the signal delay exceeds about 15 ms, the user perceives the
delay between the transmit signal and the signal in the receiver.
These delayed components of the received signal are referred to as
the echo. The presence of echo is objectionable and lowers the perceived
quality of the connection.
The signal paths that cause echo exist for both Analog and VoIP telephones.
For the case of POTS telephones, an additional component of the receive
signal is generated by imperfect matching whenever a 2 to 4 wire hybrid
is used. Fortunately, the delay in the analog telephone system is
low enough that these signals are not generally perceived as echo.
For VoIP telephony, significant delays are present in telephones and
the network. A VoIP telephone will typically capture a 20 ms frame
of audio from the transmitter, encode the audio using a codec, and
then transmit a packet to the network. The send latency of the telephone
is the time from the start of capture of an audio frame, to the time
the transmit packet appears at the network port of the telephone.
A telephone should have a send latency of less than 35 ms.
The send latency can be improved by capturing smaller frames of audio
and by increasing the speed of encoding in the telephone, but the
potential to reduce frame size is limited by efficiency concerns in
the IP network, and the smallest packet size in common usage is 10
ms.
For the receive direction, latency is the time from when a packet
arrives at the network port of the telephone to the time when the
sound begins to play at the receiver of the telephone. A good telephone
should have a receive delay of less than 65 ms (per TIA-810-A).
The receive latency is longer than the transmit latency because the
telephone is required to maintain a buffer of packets to account for
jitter in the timing of received packets, and to re-order packets
that traveled to the telephone via different network paths. As with
Send latency, the potential to reduce the receive latency is limited.
From the typical send and receive latency delays in VoIP, it is apparent
that any component of the transmit signal that is returned to the
receiver via a path outside the near-end telephone will have enough
delay that it is clearly perceptible to the talker as being an echo.
The round trip delay time from the near-end telephone to the far-end
telephone and back again is typically on the order of two hundred
milliseconds.
For the case of more complex networks of VoIP telephones, there are
other potential sources of echo. For example, a network of VoIP telephones
may connect through a gateway to the PSTN network. The gateway will
have a 2 wire to 4-wire hybrid. Since the hybrid will not perfectly
match the impedance of the PSTN network line, an echo is generated,
and because the VoIP telephone has delay, the echo will be perceptible
to the user of the telephone.
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| Echo
control for VoIP telephones |
The Send
and Receive Latencies of VoIP telephones introduce enough delay that
any portion of the transmit signal that returns to the receiver of
the same telephone will be perceived as echo. As a result, echo control
is required.
For centralized telephone systems such as digital cellular networks,
it is possible to locate echo cancellers at a central location in
the network. For VoIP, calls are established directly between telephones
without any central location where echo control can be implemented.
As a result, VoIP telephones are required to control the amount of
echo that they generate. Simply, this means that a signal applied
to the network receive port of the telephone should not be transmitted
back to the network on the transmit port.
Note that a telephone controls echo that it generates; it does not
attempt to reduce echo caused by other telephones or gateways in the
network; it is the responsibility of each network device to control
the echo that it generates
One consequence of strategy is illustrated in the following figure:
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Figure 2: Comparison of echo
signal between poorly and well designed telephones.
(Click to enlarge)
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When a telephone
call is established between two telephones, one of which has a good
design that generates little echo, and another of poor design that
generates a large echo signal, it will be the user of the well designed
telephone that hears echo. This occurs because the poorly designed
telephone couples the signal it receives from the well designed telephone
back into the transmit path and sends it back to the well designed
telephone, where the user hears a delayed copy of their own voice.
In the other direction, the well designed telephone does not couple
the signal it receives into the transmit path, and so the use of the
poorly designed telephone does not hear echo.
This situation can mislead the user of the poorly design telephone
to believe that their telephone is working well, and mislead the user
of the well designed telephone to believe that their telephone has
an echo problem when in fact the opposite is true.
For Handset telephones, echo control is relatively simple to achieve
because the receiver is intended to be close-coupled to an ear and
so its output amplitude is low. Similarly, the transmitter microphone
is relatively close to the talker’s mouth, and so its sensitivity
is not high. This, combined with the relative positions of the two
transducers, tends to result in only a small amount of acoustic coupling
of the received signal back to the transmitter.
For Handsfree telephones, echo control is much more complex. Compared
to a handset telephone, the speaker of the handsfree telephone is
much louder because it is not closely coupled to the user’s ear, and
the microphone of the handsfree telephone is more sensitive because
it is further from the user’s mouth. As a result, the acoustic coupling
of the speaker to the microphone is high.
Various techniques have been developed to control echo. Handsfree
telephones frequently employ more than one method to achieve echo
control. Methods of echo control include the following:
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• Echo
Cancellers
An echo canceller is a software algorithm that filters
the transmit signal of the telephone and attempts to remove the portion
of the signal that is due to the transmitter picking
up the receive output of the telephone.
When the telephone receives a signal from the network,
the echo canceller will dynamically adjust its parameters to try and
minimize the amount of echo being transmitted. Typically
this adjustment is only done when there is no transmit signal being
applied to the telephone (single-talk mode).
The cancellation of the echo is never perfect.
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• Non
Linear Processing
When the user of the telephone is not speaking,
it is possible to apply attenuation to the microphone path. The amplitude
of
the echo signal is reduced by the amount of the
attenuation. This is a relatively simple method of reducing echo,
but is only
applicable for single-talk conditions.
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Non Duplex
/ Partial Duplex Operation
Handsfree telephones can also be designed to switch
between distinct sending and receiving states. Software in the
telephone determines which state the telephone should
operate in based on the relative signal levels in the transmit and
receive directions. Signals in the opposite direction
to that in which the telephone is operating are attenuated. Telephones
that attenuate by up to 20 dB are said to be partial
duplex, and those that attenuate by more than 20 dB are said to be
non duplex. A telephone with less than 3 dB
attenuation is considered full duplex.
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Location of Transducers
The placement of the speaker and microphone on the
telephone affect the acoustic coupling between them. Echo
performance can be improved by the careful design
and placement of these transducers.
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| Measurements
of Echo Performance |
Echo Response
is a measurement used to quantify the echo performance of a telephone.
It is a measurement of the ratio of the transmit output signal to
the applied receive signal, measured at the network port of the telephone.
Echo response is measured over a range of frequencies.
Weighted Terminal Coupling Loss (TCLw) is an average of the echo response
taken over a range of frequencies. TCLw is a single number that indicates
how well the telephone attenuates it’s echo signal. TCLw is expressed
in dB. A higher TCLw indicates more attenuation of the echo.
Telephony standards typically specify echo control requirements in
terms of TCLw. Often, the TCLw measurements are to be adjusted to
normalize the result based on the nominal Send Loudness Rating (SLR)
and Receive Loudness Rating (RLR) of the applicable standard. For
example, the TIA-810-A standard for narrowband IP telephones requires
a normalized TCLw of 52 dB for handset, and a normalized TCLw of 45
dB for handsfree.
As can be seen from the above limits, the echo signal path requires
a large amount of attenuation in order for the telephone to provide
acceptable performance. Due to the high attenuation, the measurement
of TCLw involves the measurement of low amplitude signals. The maximum
value of TCLw that can be measured is limited by noise.
TCLw tests must be performed in a high quality anechoic chamber to
eliminate the effects of background noise. The telephone itself also
generates noise. Usually it is the send noise of the telephone that
is the limiting factor on the maximum TCLw that can be measured. It
is important to be aware that the maximum TCLw that can be measured
may be relatively close to the required limits for TCLw.
If a telephone generates excessive noise, either high send noise,
or extreme receive noise, then it may not be possible to obtain a
TCLw measurement that passes the required limits. Noise problems should
be corrected before attempting to improve TCLw performance unless
it can be verified that the noise is not limiting the TCLw reading.
When the normalized TCLw is being measured, it is important that the
send and receive loudness ratings of the telephone are not significantly
quieter than nominal. If the telephone is too quiet, the normalization
will subtract from the TCLw reading. Since the maximum TCLw is limited
by noise, it may become impossible to achieve a TCLw that passes the
required limits. If a telephone has problems with send and receive
loudness, these should be corrected before making measurements of
normalized TCLw.
The type of test signal used also affects the maximum TCLw that can
be measured. In terms of noise immunity, a sine wave is the best test
signal to use. The sine wave has the lowest peak amplitude relative
to it’s RMS level. This allows high stimulus levels to be applied
to the telephone without clipping, and all the energy of the test
signal is applied at a single frequency at any given time. Measurement
can be performed through a narrow bandpass filter to remove much of
the noise from the measurement. As a result, a sine wave stimulus
signal is least affected by noise.
White noise or pink noise can also be used as test signals for TCLw.
These signals have higher peak amplitudes, which limits the maximum
receive level that can be used for testing. The energy in these signals
is spread across a range of frequencies. Send noise of the telephone
has a greater effect on these test signals than sine wave signals.
Both the noise and sine wave signals can be modulated or pulsed to
reduce that chance that the telephone will interpret the test signal
as a steady state background noise. Some telephones will change gains,
switch states, or attempt to mute such signals.
Voice or artificial voice can also be used as test signals. These
signals are the least desirable in terms of noise immunity. They typically
have peak amplitudes approximately 20 dB about the RMS level. With
a telephone having high noise it may be difficult to pass TCLw limits
when using these test signals. Even send noise that is within the
limits permitted by standards may be too much to allow TCLw measurements
using voice like signals.
The measured value of TCLw will vary depending on the test conditions.
Telephones that use Non Linear Processing will typically have better
TCLw when tested in single-talk mode, that is, when sound is applied
only at the network receive port. In this case, the telephone is able
to apply attenuation to the transmit path that reduces the amplitude
of the echo signal. When these telephones are tested in double-talk
mode, where a second sound source is applied at the transmitter, these
telephones will disable the transmit path attenuation, resulting is
a lower TCLw.
The ability of an echo canceller to remove the receive signal from
the transmit path depends on the characteristics of the receive signal.
Typically, an echo canceller will work better with sine wave test
signals than with a more complex signal such as noise, voice or artificial
voice. TCLw testing using sine wave signals is not recommended when
an echo canceller is being used because the results obtained will
likely not be representative of the performance in actual use. It
is necessary to use a different test signal, which is less desirable
in terms of immunity to noise, to test these telephones.
Echo Response and TCLw are not the only measurements that can be used
to characterize the echo performance of a telephone. Temporally Weighted
Terminal Coupling Loss (TCLT) can also be used. This method, describes
in IEEE 1329-1999, was designed to take into account time dependent
behavior and psycho-acoustic effects of echo, and may also help to
overcome the limitations on TCLw measurement due to a telephone’s
sending noise level.
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For information
on how to upgrade the IP Phone Test System to perform the TCLw and
Echo Test Kit contact us.
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