sensitivity of applications to IP im- pairments. Clearly, entries below the vertical axis do not tolerate any type of packet lost; e.g. command/control actions in Telnet or interactive games, on-line-banking, e- commerce, etc. Which means that reliable service transmission will imperatively in- clude both delay control and packet transfer integrity. Service Components in UMTS Controlling delay implies keeping end-to-end one way delay below 250 ms, otherwise this impairment will annoy users and service quality perception will diminish. When packets get lost due to late arrival or discarded as result of congestion, the missing in- formation degrades multimedia transmission, demanding Packet Loss Concealment (PLC) techniques in voice type transmission and error correction or re-send on data transmission. Although PS or IP networks have flexibility when using codecs, we still need to add encoding time to the end-to-end delay. While the delay for different types of codecs illustrated in Figure 6.6 vary depending on their physical parameters, e.g., type, bit rate, and frame size as noted in Table 6.5; all must allow normal CS voice quality. Table 6.5 Selected Codec Parameters Codec Bit rate (kbps) Frame size (ms) Total delay (ms)* G.711 Based on packet size G.726 ADPCM Based on packet size G.729/A CS-ACELP G.732.1 MP-MLQ GSM-EFR ACELP *Total delay assumes one frame per packet Voice quality obtained through test methods, e.g. Mean Opinion Score (MSO) de- scribed in ITU recommendations P.800, rate the GSM-EFR codec quite acceptable (Fi- gure 6.6). This codec corresponds to the AMR family selected for UMTS. Hence, when it comes to delay limits for future VoIP services, e.g. 3G networks will not add unnec- essary delays. Codec Codec Types with Clean Speech Quality Normal CS Voice Quality Figure 6.6 Codec quality perception. Finally for completeness, we list the ITU G.114 recommendations on delay limits. De- pending on the applications delay ranges can be noted as follows: 50 ms limit for processing delay, will vary based o