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B13 End-to-End Delay and Delay Variation
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A key component of the overall voice quality experienced by the user is the end-to-end delay As a guideline, ITU G114 specifies that wherever possible, the one-way transmission delay for an international reference connection should be limited to 150 ms It is important to stress that the international delay budget is under pressure and that the 150-ms target is already broken if, for example, satellite links or cellular access systems are deployed In a packet-based network, the end-to-end delay is made up of fixed and variable delays; the fixed delays include the packetization delay and the transmission delay, whereas the variable delay is imposed by statistical multiplexing (and hence queuing) at each (MPLS) router For voice and other real-time media, the variable delay must be filtered at the receiving terminal by an appropriate jitter buffer to reconstitute the original constant rate stream This process imposes an additional connection delay that is equal to the maximum packet delay variation (that is, this fixed delay is set by the worst statistical delay irrespective of its rate of occurrence) Thus, packet delay variation should be minimized within the VoIPoMPLS network to minimize the overall one-way delay as well as reduce costs in the end equipment by reducing the memory requirements for the jitter buffer The MPLS network should be able to create an LSP with a controlled delay variation
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Motivations, Drivers, Approaches, and Advantages of VoMPLS
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B14 Packet Loss Ratio (PLR)
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Packet loss is a key voice impairment factor For voiceband connections, ITU-T G821 specifies the overall requirements for error performance in terms of errored seconds and severely errored seconds Under this definition, for the majority of voice-encoding schemes, the loss of a single VoIPoMPLS packet will cause at least a single severely errored second ITU-T G821 specifies an end-to-end severely errored seconds (SESs) requirement of 1 in 10-3 this requirement is predominately driven by the demands of voiceband data (fax and modem) Speech impairment in packetized voice networks, on the other hand, can be unnoticeable with fairly high packet loss (as high as 5 percent in some cases) The relationship between SES and packet loss is not well known In networks where it is important to pass voice, modem, and/or fax data without degradation, techniques such as controlling packet loss may be employed Alternatively, demodulation, data pass-through, and remodulation of fax/modem calls may be employed to achieve such a goal
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When determining the timing accuracy for VoIPoMPLS domains, the following types of traffic must be considered: speech, voiceband data, and circuit-mode data All speech traffic is obtained by the equivalent of sampling the analog speech signal at a nominal 8 kHz and generating linear PCM This can be companded to 64 Kbps in accordance with ITU-T G711, or it can be compressed to a lower bit rate either on a sample-by-sample basis (such as ADPCM G726/7) or on a multiple-sample basis to produce packets (such as various forms of code-excited linear prediction [CELP]) Voiceband data traffic is obtained by sampling the analog modem signal, that is, the low-rate data modulated onto defined frequency carrier signals, in the same way as for speech and companding to 64 Kbps using G711 Except for very low data rates, compression is not possible In all cases, provided the traffic can be carried by the VoIPoMPLS packet network directly from encoder to decoder and the decoder can work on the sample rate determined from the received traffic, the encoder only needs to have a frequency tolerance sufficient to achieve the required analog frequency response and constrain the traffic data bandwidth; thus, the VoIPoMPLS packet network would have no particular frequency tolerance requirements (Packet jitter including delay variation would still have to be constrained within buffer sizes, and measures such as sequence numbers would still be needed to maintain accurate determination of the transmitter sample rate under circumstances of packet loss)
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