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This section summarizes dimensions contributing to QoE for conversational voice in a Triple-play implementation. At the voice Service Layer, quality of experience dimensions include:

• Control Plane:

– Interactive responsiveness (call set-up, control and teardown) • Data Plane:

– Voice Intelligibility

• Many potential impact points on voice intelligibility in an end-to- end system

• Distortion, loss, delay, echo, and transcoding are common impairments

• Usability

– Service UI (set-up, directories, caller ID, configuration, etc.…) • Reliability / Availability

• Security / Privacy

– for user, and Telco

– security impacts on other dimensions (ex. encryption / decryption delay) A general overview of VoIP customer premises deployment options may be found in TR- 110 DSLHome™ Reference Models for VoIP Configurations in the DSL Home53. The voice service QoE guidelines that follow apply to both voice sessions from pure VoIP phones connected directly to the IP network and from traditional analog phones sitting behind terminal adaptors (ATA).

There are four key factors affecting QoE for VoIP:

o delay (including delay variation or jitter),

o the speech codec,

o cell/packet loss,

o echo.

These factors are included in the ITU G.107 E-model R measurement for predicting conversational speech quality as shown in Figure 15. A fifth factor: signal level, is not

affected by IP transport, but it is important to establish proper settings at the point where an IP network connects to another type of network.

Figure 15 Summary of the voice QoE impairments and the impact on the E-Model “R”

Speech codec

The speech codec chosen will have a strong influence on the final obtained quality, both because of the baseline quality of the codec (that is, the quality of the codec without other impairments) as well as the response of the codec to other factors, such as presence of background noise, packet loss, and transcoding with itself or another codec. The choice of codec is an important determinant of the overall performance of VoIP.

End-to-end delay

The end-to-end delay of a voice signal is the time taken for the sound to enter the transmitter at one end of the call, be encoded into a digital signal, travel through the network, and be regenerated by the receiver at the other end. Delay is sometimes called latency. When delay is too long, it may cause disruptions in conversation dynamics. As well, increasing delay makes echo more noticeable.

Jitter

Variation in delay, caused by differences in the time taken for packets to cross the network, is called jitter. Jitter is a concern because the decoding of the digital signal is a synchronous process and must proceed at the same constant pace that was used during

encoding. The data must be fed to the decoder at a constant rate. Variation in packet arrival times is smoothed out by the jitter buffer, which adds to the end-to-end (mouth-to-ear) delay. Jitter is not considered a separate impairment because the effects of jitter in the packet network are realized in the output either as delay (added as the jitter buffer wait time) or as distortion from packet loss (because packets arriving after the expiration of the jitter buffer wait time are not included in the output signal).

Packet loss

As in other packet-based services, packets may be dropped during their journey across the network. Packets may also be lost if they are late in arriving at the destination codec buffer and miss their turn to be played out. The missing information degrades the voice quality, and a Packet Loss Concealment (PLC) algorithm may be needed to smooth over the gaps in the signal.

Echo control

Because of the longer delay introduced by VoIP (compared to POTs or TDM voice networks), echo control is a major concern. A given level of echo sounds much worse when the delay is longer. Echo control at the appropriate places in the connection will protect the users at both ends. Echo control relies on the correct signal levels (see Signal Level, below) as well as on echo cancellers and other techniques that prevent or remove echo from the connection.

Signal level

The level or amplitude of the transmitted speech signal is determined by amplitude gains and loss across the network. There are a number of contributors to the final signal level, and most are defined in the loss plan (sometimes called the loss/level plan) of the network. The loss plan for TDM ensures that the output speech is heard at the proper level and contributes to the control of echo. The loss plan for VoIP is reasonably simple; the sensitivities of the sending device (say, an IP phone) and the receiving device (say, a media gateway) are defined by standards, and there is no gain or loss in the packet portion of the network.

Things are more complicated when a packet network is connected to another network with a different loss plan. When the other network is a traditional network with analog access, it may be necessary to adjust the level of each signal path (the signal sent to the other

network and the signal coming from the other network) to account for the loss plan of that network. The required loss for each path must be determined and set accordingly. Errors in the loss settings can cause incorrect speech level or audible echo at one or both ends of a connection.