Figure 6.1 (a) represents functional PSTN - based voice conversation. In
the fi gure, a voice conversation is happening between two phones — A and B. In a PSTN call, the telephone interface is terminated at the central offi ce (CO) or digital loop carrier (DLC) through a two - wire TIP - RING interface. The DLC hybrid as part of a subscriber line interface circuit (SLIC) creates a two - to - four - wire conversion. Analog - to - digital converter (ADC) and digital - to - analog converter (DAC) are referred to as a subscriber line access circuit (SLAC), also known as hardware CODEC. The DLC - to - PSTN CO and CO - to - CO interfaces work as four - wire conversions with separate send and receive
paths. In the local PSTN call, the CO completes the call directly or through another DLC. In the long - distance call, the PSTN CO will send voice and sig- naling bits to the destination CO. The destination CO completes the call through its local DLC and telephone interfaces. The interface between the DLC and the telephone is a two - wire TIP - RING in most countries. Inside the telephone, a handset works equivalent to a four - wire interface, but the telephone interface going to the DLC is of two wires. A hybrid inside the telephone creates a two - to - four - wire conversion. Echo is created at every
two - to - four - wire hybrid of DLC and telephone. A telephone hybrid echo
level of 6 to 14 dB is more dominant than a DLC electronic circuit hybrid echo of 18 to 24 dB. In Fig. 6.1 (a), echo is created at the telephone hybrid. For simplifi cation in representation, the echo path through the DLC hybrid path is not shown in Fig. 6.1 (a), but it is marked in Fig. 6.1 (b) on the phone - B side.
Talker echo is explained with an example call between A and B. In the example shown in Fig. 6.1 (a), the person at phone - A is speaking and is called the “ talker. ” The person at phone - B is the “ listener. ” Voice from A goes up to B, and part of A comes back to A along with voice originating from B. Person at phone - A will be hearing his own voice called “ talker echo ” with round - trip delay. The talker listening to his own voice after delay is the talker echo. Round - trip delay is the transmission time for voice to travel from A to B and returning from B to A as echo. The delay from either A - to - B or B - to - A is called one - way delay. Average one - way delay is calculated as half of round - trip delay. While A is speaking, the presence of echo with increased round - trip delay limits the conversation comfort for person - A. The person at A slows
Figure 6.1. Echo representations in PSTN. (a) Talker and listener echo functional representa- tion. (b) End - to - end losses and echo representation.
down conversation with long pause periods and lowers the speaking sound level to minimize the echo annoyance. The same interpretation is applicable to B. When B is speaking, B will also be getting his own voice (talker echo) with round - trip delay.
Echo cancellation requirements are given in G.168 [ITU - T - G.168 (2004) ]. G.131 [ITU - T - G.131 (2003) ] provides recommendations for acceptable talker echo in relation to one - way delays [ITU - T - G.114 (2003a) , ITU - T - G.114
(2003b) ]. At lower delays, some amount of talker echo is not a problem.
The annoyance of echo keeps growing with an increase in the end - to - end delay.
It is also possible to see multiple circulations of echo, creating listener echo. Listener echo characteristics are given in the G.126 recommendation [ITU - T - G.126 (1993) ]. Similar to talker echo, listener echo is also sensitive to one - way and round - trip delays. A ’ s voice after returning back from B as talker echo may leak back to the phone - A hybrid and go back up to B. This process makes B to hear the same speech of person - A ’ s voice a second time or multiple times. The second time voice of A is the “ listener echo ” — the echo of the talker ’ s voice heard by the listener a second time. In both listener and talker echo, B is listening to A ’ s voice. Listener echo is mainly created when both A and B are not keeping any echo cancellers or minimum losses in the system. VoIP systems use an echo canceller as part of voice processing algorithms. Hence, listener echo is not applicable in VoIP voice calls with minimal echo cancella- tion operation. Listener echo is more delayed than talker echo and more annoying than talker echo. The presence of listener echo indicates instability in overall transmission that may be resulting from the improper gain/loss plan- ning in the overall system and the characteristics of phones used. The incor- poration of talker echo cancellation can prevent listener echo.
6.1.1 Echo and Loudness Ratings
In this section, SRL, RLR, CLR, transmit (Tx)/receive (Rx) losses, LR, TELR, ERL, TCLw, mean one - way delay, and sidetone are presented in relation to markings made in Fig. 6.1 (b). These parameters play a major role in echo characterization and perception. The values in this section with dB scale are loss parameters, and a high value indicates a higher amount of loss.
SLR is the send loudness rating. It is the ratio represented in dB scale of the sound pressure produced by a talking person to the voltage produced by the telephone on the TIP - RING interface. Analog telephones give an SLR of about 7 dB [ITU - T - P.79 (1999) ] and vary by country requirements. In Chapter 1 under TR - 57 tests, another parameter SRL is given. This SRL is the singing return loss, which is not the same as this SLR. This SRL is an electrical refl ec- tion loss that is closely associated with ERL.
RLR is the receive loudness rating. It is the ratio represented in dB scale of the telephone electrical signal to the acoustic signal conversion [ITU - T - P.79 (1999) ]. Analog telephones give an RLR of about 3 dB and vary by country
requirements. RLR and SLR vary based on the country - specifi c phone. The combined value of SLR and RLR should be 10 dB for a handset of nominal performance. The SLR minimum is 3 dB, and the RLR minimum is 1 dB. The minimum values will make the voice too loud and create degradation of voice quality.
CLR is the circuit loudness rating decided by the two - wire lines from PSTN DLC to the telephone. In PSTN, long lines will go from DLC/CO to the end - user telephone. CLR will also contribute to the losses, but it is not signifi cant. In VoIP service, CLR is not accounted for because of the short lines from the VoIP system to the user telephone.
Tx and Rx losses are the intentional losses kept in the system by design. Tx loss is associated with the sender. Rx loss is at the receiving end. The losses are marked as Tx(A) and Rx(B) in Fig. 6.1 (b). These losses are also called padding (meaning attenuation) losses. These losses are called send (top) path and receive (bottom) path losses. The same losses are also applicable in each side of the VoIP system. There are no losses in four - wire digital PSTN and on IP network.
ERL is the echo return loss. ERL [ITU - T - G.168 (2004) ] is attenuation of a
signal from the receive - out (R out ) port to the send - in port (S in ) of an echo
canceller. Hybrids inside SLICs and telephones create echo. In practical systems, phone ERL (6 to 14 dB) is the more dominant contributor than SLIC ERL (18 to 24 dB). For echo cancellation, the resultant ERL marked as passing through a SLIC and telephone hybrid in Fig. 6.1 (b) is considered. The singing return loss explained in Chapter 1 is related to ERL. ERL has to be as high as possible. Good phones with matching impedance of phones and interfaces offer an ERL of 24 dB.
TCLw is the weighted terminal coupling loss of the telephone sets. TCLw is applicable to digital handsets and to IP phones in non - speakerphone mode. Digital phones are used with PBX and integrated services digital network (ISDN) service, and IP phones directly use a four - wire interface. Electrical echo is not present on four - wire interface phones. A small part of acoustic echo couples through handsets, but the coupling loss in TCLw is of 45 dB [ITU - T - G.131 (2003) ]. This loss is suffi cient to use four - wire equivalent phones without echo cancellation. TCLw is measured as attenuation from the digital input to the digital output at the 14 1/3 - octave bands between 200 Hz and 4 kHz [ITU - T - P.341 (2005) ]. TCLw is measured when no signal occurs from the local user speech in the send path.
LR is the loudness rating. It is the loss from the mouth of A to the ear of B, similar to end - to - end transmission loss taking into account both end phones [ITU - T - P.79 (1999) , URL (Cisco - EC) , TIA/EIA - 912 (2002) ]. The loudness rating value consists of losses in the system — namely the SLR, CLR of lines, RLR of the receiving system, and end - to - end any padding losses. For sending voice from A to B, EC at A will not introduce any loss. CLR is usually a small value. In relation to markings made in Fig. 6.1 (b), LR from A to B is given as follows.
LR A to B SLR A CLR A Tx A Rx B CLR B RLR B
SLR A se
( ) = ( ) + ( ) + ( ) + ( ) + ( ) + ( )
= ( ) + nnd path loss RLR B+ ( )
LR B to A( ) =SLR B( ) +CLR B( ) + ( ) +Tx B Rx A( ) +CLR A( ) +RLR A( ) The LR suggested value is from 8 to 12 dB [ITU - T - G.111 (1993) ], and the preferred value is 10 dB. LR outside the range of 8 to 12 dB degrades the quality. A higher value of LR is attenuated voice, and a lower value is loud voice. Attenuated or loud voice degrades voice quality. This parameter is also called the overall loudness rating (OLR).
TELR is the talker echo loudness rating. It is the level difference between original acoustic levels of voice at microphone to the received acoustic echo power [ITU - T - G.131 (2003) ] on the same phone speaker. TELR is the round - trip acoustic and electrical combined loss. A higher value of TELR is desired, and it improves voice quality by decreasing the echo level. With reference to Fig. 6.1 (b), TELR is signal loss from the mouth of A to the ear of A after returning as echo. In TELR calculations, voice makes one round trip by going up to B and coming back to A. It is not sidetone. By considering CLR and digital network (PSTN or VoIP) losses as insignifi cant, TELR(A) as seen at phone - A and TELR(B) as seen from B are given below. All parameters associ- ated here are expressed in dB scale. When echo cancellers are used, echo canceller rejection to echo in dB also contributes to TELR. TELR has to be as high as possible. The TELR requirements are given in relation to mean one - way delay. For higher delays, TELR has to be more than 65 to 75 dB. A higher value of TELR is achieved by employing an echo canceller.
TELR A SLR A Tx A Rx B ERL B Tx B EC rejection at B R ( ) = ( ) + ( ) + ( ) + ( ) + ( ) + ( ) + xx A( ) +RLR A( ) TELR B SLR B Tx B Rx A ERL A Tx A EC rejection at A R ( ) = ( ) + ( ) + ( ) + ( ) + ( ) + ( ) + xx B( ) +RLR B( )
In the above formulation, ERL(A) and ERL(B) include the combined infl u- ence of the SLIC hybrid and the phone. While viewing from phone - A, Tx(A) + Rx(B) is considered as send path loss, and Tx(B) + (EC rejection) + Rx(A) is considered as receive path loss. Receive path includes echo canceller rejec- tion. TELR (A) is also expressed as
TELR A( ) =SLR A( ) +send path loss ERL B+ ( ) +receive path loss RLR A+ ( ) TELR(A) will not go through SLR(B), RLR(B) of phone - B. It will only go through an equivalent ERL of hybrids at B. Additional details on TELR are given in G.122, G.131, P.310, G.107 [ITU - T - G.122 (1993) , ITU - T - G.131 (2003) , ITU - T - P.310 (2003) , ITU - T - G.107 (2005) ], and TIA/EIA - 116A (2006) .
Sidetone is a small part of the microphone electrical signal that is fed back immediately as leakage to the same phone speaker. Sidetone is also similar to
talker echo, but sidetone arrives within imperceptible delay of 1 ms [ITU - T - G.107 (2005) ], and for delays above 2 to 4 ms, sidetone may be perceived as echo. Sidetone creation happens as part of the phone circuitry as intentionally created leakage or from the phone hybrid. Sidetone marked in Fig. 6.1 (b) on the phone - A side is essential to create a balanced sound level from the talker. In the absence of sidetone or very low - level sidetone, the talker will get the perception of hollowness (silence) and start speaking loudly. Too much of sidetone creates annoyance, and the talker will have tendency of reducing the speaking sound level. Sidetone mask rating (STMR) levels of 7 to 12 dB and a maximum of 20 dB [ITU - T - G.121 (1993) ] is considered in the early evalua- tions as required for VoIP communication. In recent revisions of G.107 [ITU - T - G.107 (2005) ], sidetone is considered as 10 to 20 dB with a default value of 15 dB.
Mean one - way delay is the half of the round - trip delay. Round - trip delay considers a talker ’ s echo reaching the ear after completing a round trip from the destination. This delay has to be as low as possible. One - way delays up to 150 ms are treated as comfortable for VoIP voice communication [ITU - T - G.114 (2003b) , ITU - T - Y.1541 (2006) (2006)], even though lower delay improves voice quality. In inter - regional PSTN and VoIP calls, one - way delay increases up to 300 ms [ITU - T - G.114 (2003b) ]. In usage of terminology, end - to - end delays and one - ways delays are used without much distinction.