QoS in Integrated 3G Networks Robert Lloyd Evans (2)

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.g., to 1/3) to give more error correction for the optimum practical result.UMTS_AMRSince UMTS_AMR is the main codec for UMTS, the way in which ituses the radio bearers and the acceptable levels of error have been thoroughly investigated.Like the earlier GSM algorithms, this has three RAB subflowscorresponding to the three classes of bits, each of which leads to a dedicated transport channel.AMR operates at a rate of 50 20-ms frames per second, but the numberof bits per frame depends on which of the eight possible rates is in use.These frame sizes are mandatory, but the subdivision into the three classes, A, Band C, is only optional with suggested values as in Table 10.4 [8]A call will not necessarily permit use of all of these rates, and an activesubset of those available for use is specified.For each member of the active subset a RAB format combination indicator (RFCI) is defined; togetherTable 10.4AMR Frame SizesRate (Kbps) Frame SizeA BitsB BitsC Bits12.2244811036010.22046599407.9515975847.414861876.713458765.911855635.1510349544.75954253254QoS in Integrated 3G Networksthese form the RAB format combination set (RFCS).The permitted sub-frame sizes for each quality class also define the block sizes in the TFS for the mapping of transport channels to the physical channels of the radio link (see Section 3.3.4).For example, if the active subset consisted of only 12.2- and 7.4-Kbps rates, plus 0 Kbps (for silences) and a silence descriptor rate fordescribing the characteristics (e.g., insert background noise) of silences, the RFCS would be as shown in Table 10.5; if all possible rates were permitted,the TFS for each channel would include all the entries in each of columns A,B, and C in Table 10.4, plus entries associated with silences.The figures in the columns in Table 10.5 provide the SDU sizes for theSDU format indicators for each subflow in the RAB parameters (see Section9.2).The RAB parameter IE also needs the bit rates (maximum 12.2 Kbps,guaranteed 7.4 in this case), error rates, and delivery behavior for erroneous frames.The values for these fields are left to the operator implementation,but with plausible values as indicated in Table 10.6.The SDU error ratio is only applicable to subflow 1 as it is the only onewhose frames, sent as Iu PDU type 0, are provided with a CRC frame qualityindication set by the level 1, the others being PDU type 1 (see Section 8.1.3.).Table 10.5Example of an RFCS for AMRRFCISubflow 1Subflow 2Subflow 3TotalSource Rate8110360244116187148223339394Table 10.6Possible Error Parameters for AMR SubflowsSubflowResidual BERSDU Error RatioError Delivery11 × 1067 × 103y21 × 10335 × 103Specific Applications255The AMR subflows are passed down to level 2 under RRC controlwhere they all use RLC transparent mode, but are passed to the MAC layerfor separate DCHs.In this mode RLC only performs segmentation and reas-sembly of the Iu PDUs so that they fit into suitable sized RLC PDUs fortransmission at the correct rate.These are passed to the MAC layer wherethey are again handled transparently.Data is passed between the MAC layerand the physical layer via transport blocks.In the case of AMR, the blocksizes for each of the three DCH are simply those for the numbers of bits forthe corresponding class in Table 10.5.The transport format also indicatesthe TTI of 20 ms for each, the type (here convolutional) and rate of thechannel coding to use, and the CRC size to use for each.The channel codingrate depends on which AMR bit rate is used, as the purpose of the variablerates is to allow the code rate to be increased if signal reception is poor.For the class A bits at 12.2 Kbps, the code rate is 1/2, while B and C bits areunencoded, but at 7.4 Kbps the A bits use a rate 1/3 code, and B uses 1/2.Only the class A bits use a CRC, and this is an 8-bit field.The end-to-end delay target for AMR speech is the 100-ms limitneeded to facilitate the speakers from not interrupting each other.UMTS speech normally uses the CS core network with signaling forATM AAL2, but when a conversation is held with an IP phone then eitherthe PS domain or the Internet multimedia core might be involved instead.Ifso, the call will entail use of a media gateway for transcoding, while signaling is handled by the MGCF (see Chapter 8) through SIP commands.The session is established by a SIP INVITE that details the call features throughSDP elements, followed by SIP series 1xx call progress signals and a SIP OKTEAMFLY200, with eventual release by a SIP BYE [ Pobierz całość w formacie PDF ]

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