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d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 11.2 Framing Pattern Substitution in A-TRAU frame | The Framing Pattern Substitution is used in each of the eight 36 bit data fields of the A-TRAU frame (see Figure 5) to avoid transmitting a sequence of eight zeroes (called Z sequence in the following).
The purposes of FPS is to avoid erroneous synchronisation to the A-TRAU due to sixteen zeroes occurring accidentally in the data bits and to avoid erroneous synchronisation to V.110. The synchronisation pattern of two consecutive V.110 frames cannot be found within a stream of A TRAU frames. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 11.2.1 FPS encoding | A Zero Sequence Position (ZSP) field is used to account for the occurrence of eight zeroes in the 36 bit data field.
NOTE: A sequence of eight zeroes is considered as a block (e.g. a stream of eleven consecutive zeroes produces only one ZSP and not four ZSPs).
The ZSP field is defined as follows:
Table 5
1
2
3
4
5
6
7
8
1
C
A0
A1
A2
A3
A4
1
The meaning of the different bits of the ZSP field is :
C : Continuation bit. '0' means that there is another ZSP in the data field. '1' means that there is no other ZSP.
A0-A4 :address of the next Z sequence (eight zeroes) to be inserted. The address ‘00001’ corresponds to the bit D1, the value ‘11101’ to the bit D29, (A0 is the msb, A4 is the lsb).
NOTE: a Z sequence substitution cannot occur at bit D30..D36 (as it is 8 bit long)
1 : locking bit prevent the false occurrence of a Z sequence.
The Framing Pattern Substitution is applied in each of the eight 36 bit data field (see Figure 5).
Bit Zi indicates whether FPS is used in the ith 36 bit data field (i=1 to 8). The coding of the Zi bit is the following:
Table 6
Zi (i=1..8)
meaning
1
no substitution
0
at least one substitution
If Zi bit indicates no substitution, the output data bits of FPS are equal to the input data bits.
If Zi indicates at least one substitution, the bits D1-D8 contain the first ZSP.
The following description indicates the general operating procedures for FPS. It is not meant to indicate a required implementation of the encoding procedure.
Figure 1
Step 1 :
The input 36 bit sub frame is considered as a bit stream in which the bits are numbered from 1 to 36.
This bit stream contains 0, 1 or several Z sequences, (Zseq1 to Zseq3 on the figure)
The Z sequence is a sequence of 8 consecutive zeroes : '0000 0000'
Step 2 :
Starting from this bit stream, two lists are built up :
2-a : the 'a' list which contains the address of the first bit of each Z sequences.
2-d : the 'd' list which contains all the data blocks which do not have the Z sequence.
Step 3 :
The 'a' list is transformed so as to build the ZSP list. Each ZSP element is used to indicate:
at which address is the next Z sequence of the message
if yet another ZSP element is found at this address (link element)
Step 4 :
The output 37 bit sub frame is built from:
the Zi field which indicates whether the original message has been transformed or not with this technique. In the example given in Figure 1, Zi shall be set to '0' to indicate that at least one FPS has occurred.
the ZSP and D elements interleaved.
As the ZSP elements have exactly the same length as the Z sequence, the sub frame length is only increased by one (the Zi bit), whatever the number of frame pattern substitutions may be.
For special cases, refer to annex A. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 11.3 A-TRAU Synchronisation Pattern | The frame synchronisation is obtained by means of the first two octets in each frame, with all bits coded binary "0" and the first bit in octet no 2 coded binary "1". The following 17 bit alignment pattern is used to achieve frame synchronisation :
00000000 00000000 1XXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX
XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX
XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX
XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX
XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 12 THE RAA'' FUNCTION | On the IWF side of the A interface, the RAA" function shall convert between the A-TRAU format and a synchronous stream. FPS shall be performed by this function as well, see subclause 11.2. In transparent operation, the RAA" function shall handle the M1 and M2 bits as specified for the RA1' function in 3GPP TS 04.21.
In non-transparent operation, the RAA" function shall map between the A-TRAU format and 290 bit blocks consisting of M1, M2 and 288 bits making up half of an RLP frame, see subclause 15.2 of this GSM TS. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 13 The RA2 Function | Described in 3GPP TS 04.21. The RA2 function shall be applied only for single slot operations. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 14 The A-interface Multiplexing Function | The multiplexing function shall be applied only for AIUR up to and including 57.6 kbit/s for multislot operations.
The multiplexing function is based on the ITU-T I.460. The multiplexing function is used to combine n (n=2 to 4) substreams of multislot intermediate rate of 8 kbit/s or n substreams of multislot intermediate rate of 16 kbit/s on one 64 kbit/s stream by using subcircuits in each octet to each substream such that:
i) An 8 kbit/s substream is allowed to occupy subcircuits with positions 1,3,5 or 7 of each octet of the 64 kbit/s stream; a 16 kbit/s stream occupies bit positions (1,2) or (3,4) or (5,6) or (7,8).
ii) The order of the bits at each substream is identical before and after multiplexing.
iii) All unused bit positions shall be set to binary “1".
iv) For transparent multislot configurations the lowest allowed subcircuits are always used.
v) For non-transparent multislot configurations, the lowest allowed subcircuits shall be used at call set up and after change of channel configuration except at downgrading. At downgrading any of the used subcircuits may be released in uplink direction. Always, the released subcircuit(s) in downlink direction shall be the same as the released subcircuit(s) in uplink direction. At a possible subsequent upgrading, the lowest available bit positions shall be used for the added substreams.
NOTE: The rules given here are almost identical to those of I.460, Section ‘Fixed format multiplexing’, except for the rule i) is stricter in that 8 kbit/s substreams cannot occupy any positions, iv) and v) are added. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 15 Support of non-transparent bearer services | |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 15.1 TCH/F9.6 and TCH/F4.8 kbit/s channel codings | In the case of non-transparent services the RA1/RA1' function shall perform the same mapping as that described for transparent services, using 12 and 6 kbit/s radio interface data rates, with the following modification.
The E2 and E3 bits in the modified ITU-T V.110 80 bit frames shown in Figure 3 (derived from the standard ITU-T V.110 frame shown in Figure 2) are used to indicate each consecutive sequence of ITU-T V.110 80 bit frames corresponding to the four modified ITU-T V.110 60 bit frames (Figure 4) received/transmitted in one radio interface frame. This allows 240 bit Radio Link Protocol frames to/from the MSC to be aligned with the 4x60 bit frames encoded by the radio subsystem channel coder as a single unit (see 3GPP TS 05.03). The 8 bits consisting of the E2 and E3 bits in one of the above sequences is referred to as the Frame Start Identifier. The FSI value is 00 01 10 11. This value is assigned to the E2 and E3 bits as shown in Table7.
Table 7
E2
E3
First Modified ITU-T V.110 80 bit frame
0
0
Second
0
1
Third
1
0
Fourth
1
1
As each RLP frame is transported between the BSS and MSC in four modified ITU-T V.110 80 bit frames, it is necessary following a transmission break and at start up, to determine which modified ITU-T V.110 80 bit frame of the stream is the first for a particular RLP frame. This is needed so that correct alignment with the radio subsystem can be achieved.
Modified V.110 80 bit frames can slip in time during re-routing, and whilst sync exists within the modified ITU-T V.110 80 bit frame to determine the modified ITU-T V.110 80 bit frame boundaries, the FSI is required to determine which quarter of an RLP frame each modified ITU-T V.110 80 bit frame contains.
Table 8 : Relationship between FNUR, AIUR, substream rate, number of substreams and intermediate rate
FNUR
AIUR
Number of Channels x Substream Rate
Channel Coding
Multislot Intermediate Rate
2,4 kbit/s
2,4 kbit/s
2-8 times duplication of each bit to reach 2,4 kbit/s
TCH/F4.8
8 kbit/s
4,8 kbit/s
4,8 kbit/s
4,8 kbit/s
TCH/F4.8
8 kbit/s
4,8 kbit/s
9,6 kbit/s
9,6 kbit/s
TCH/F9.6
16 kbit/s
9,6 kbit/s
9,6 kbit/s
2x4,8 kbit/s
2XTCH/F4.8
8 kbit/s
9,6 kbit/s
9,6 kbit/s
9,6 kbit/s
TCH/F9.6
16 kbit/s
14,4 kbit/s
14,4 kbit/s
3X4,8 kbit/s
3XTCH/F4.8
8 kbit/s
14,4 kbit/s
19,2 kbit/s
2X9,6 kbit/s
2XTCH/F9.6
16 kbit/s
19,2 kbit/s
19,2 kbit/s
4X4,8 kbit/s
4XTCH/F4.8
8 kbit/s
19,2 kbit/s
19,2 kbit/s
2X9,6 kbit/s
2XTCH/F9.6
16 kbit/s
28,8 kbit/s
28,8 kbit/s
3X9,6 kbit/s
3XTCH/F9.6
16 kbit/s
38,4
38,4 kbit/s
4X9,6 kbit/s
4XTCH/F9.6
16 kbit/s
NOTE: The table gives the relation between the FNUR, AIUR, Substream Rate, Channel Coding and Intermediate Rate. As an example: the wanted FNUR is 14,4 kbit/s and the selected channel coding is TCH/F9.6. The data stream is split into two substreams of 9,6 kbit/s yielding an AIUR of 19,2 kbit/s. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 15.1.1 Alignment | An alignment window spanning four modified ITU-T V.110 80 bit frames shall be used to search for the pattern of 8 bits described above in order to identify alignment with an RLP frame.
In the event of failure to detect the 8 bit pattern, the alignment window is shifted one complete modified V.110 80 bit frame, discarding the contents of the most historical frame and then checking the new 8 bit pattern. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 15.1.2 Support of Discontinuous Transmission (DTX) | The E1 bit in the modified ITU-T V.110 80 bit frame shown in Figure 3 shall be used in the direction MSC-BSS to indicate that DTX may be invoked (see 3GPP TS 24.022). The E1 bit in all of the four consecutive frames relating to the RLP frame to which DTX may be applied shall be set to 1. If DTX is not to be applied, the E1 bit shall be set to 0.
In the direction BSS-MSC the E1 bit shall always be set to 0. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 15.1.3 Order of Transmission | The first bit of each quarter of an RLP frame to be transmitted shall correspond to bit D1 of a modified V.110 frame (figures 3 and 4). The remaining 59 bits of each quarter of an RLP frame shall correspond to the D and D' bits , D2 - D'12, in order left to right and top to bottom as shown in figures 3 and 4.
The first quarter of an RLP frame to be transmitted shall contain the E2 and E3 bit code 00 as shown in Table 1. The second quarter contains the code 01, etc. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 15.2 TCH/F14.4, TCH/F28.8, and TCH/F43.2 channel codings | In case of non-transparent service, a 576 bit RLP frame shall be mapped over two consecutive A-TRAU frames.
Because of that mapping, it is required, following a transmission break and at start up, to determine which A-TRAU frame of the stream is the first for a particular RLP frame. This is needed so that correct alignment with the radio subsystem can be achieved.
The two consecutive M1 bits are referred to as the Frame Start Identifier. The FSI value is 01. This value is assigned to the M1 bits as shown in Table 9.
Table 9
M1 bit
First A-TRAU frame
0
Second A-TRAU frame
1
A-TRAU frames can slip in time during re-routing, and whilst A-TRAU frame synchronisation exists, the FSI is required to determine which half of an RLP frame each A-TRAU frame contains.
Table 10 : Relationship between AIUR, substream rate, number of substreams and intermediate rate
AIUR
Number of substreams x
AIUR per substream
Channel Coding
Multislot intermediate Rate
14,4 kbit/s
14,4 kbit/s
TCH/F14.4
16 kbit/s
28,8 kbit/s
2X14,4 kbit/s
2XTCH/F14.4
1XTCH/F28,8
16 kbit/s
43,2 kbit/s
3X14,4 kbit/s
3XTCH/F14.4
1XTCH/F43,2
16 kbit/s
57,6 kbit/s
4X14,4 kbit/s
4XTCH/F14.4
16 kbit/s
57,6 kbit/s
4X14,4 kbit/s
4XTCH/F14.4
2XTCH/F28,8
16 kbit/s
NOTE: The table gives the relation between AIUR, Substream Rate, Channel Coding and Intermediate Rate. As an example: the AIUR is 28,8 kbit/s and the selected channel coding is 14,5 kbit/s. The data stream is split into two substreams of 14,5 kbit/s yielding an AIUR of 28,8 kbit/s
The same number of substreams is used in each direction, even if the AIURs in each direction differ. Superfluous substreams are filled with idle frames. These are inserted at the BTS or IWF and are discarded at the IWFor BTS respectively. At the IWF, the down link AIUR is determined by the out of band signalling (Assignment Complete, Handover Performed), whereas the up link AIUR is determined inband by examining the possible substream positions on the A interface.
. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 15.2.1 Alignment | An alignment window spanning two 290 bit blocks in case of TCH/F14.4 channel shall be used to search for the pattern of 2 bits '01' described in subclause 15.2, in order to identify alignment with an RLP frame.
In the event of failure to detect the 2 bits pattern the alignment window is shifted one 290 bit block, discarding the contents of the most historical frame and then checking the new 2 bits pattern. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 15.2.2 Support of Discontinuous Transmission (DTX) | The M2 bit in the A-TRAU frame shown in Figure 5 shall be used in the direction MSC to BSS to indicate that DTX may be invoked (see 3GPP TS 24.022). The M2 bit in all of the two consecutive A-TRAU frames relating to the RLP frame to which DTX may be applied shall be set to 1. If DTX is not to be applied, the M2 bit shall be set to 0.
In the direction BSS to MSC the M2 bit shall always be set to 0. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16 Support of transparent bearer services | |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.1 TCH/F9.6 and TCH/F4.8 channel codings | |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.1.1 User rate adaptation on the A interface, AIUR less than or equal to 38,4 kbit/s | The ITU-T V.110 80 bit frame shall be used for transparent data on the A interface. These frames are transmitted on up to four substreams multiplexed into one stream sent over the A interface. The split/combine function is applied on the substreams as specified in clause 5 of this GSM TS. The relation between the AIUR and the number of channels is specified in table11.
The 64 kbit/s consists of octets, bits 1 through 8, with bit 1 transmitted first.
For a 9 600 bit/s radio interface user rate the V.110 frame is carried with a 16 kbits/s stream which occupies bit positions (1,2).
For radio interface user rates of either 4 800 bit/s, 2 400 bit/s, 1 200 bit/s or 300 bit/s the V.110 frame is carried with a 8 kbits/s stream which occupies bit position (1). For user rates < 1 200bit/s asynchronous characters are padded with additional stop elements by the RA0 function (in the MSC/IWF) to fit into 600 bit/s synchronous RA1 rate prior to rate adaptation to 64 kbits/s.
No use of 4 kbit/s stream is foreseen.
In a given V.110 frame on the A interface:
- for 9 600 bit/s there is no repetition of bits D within the 16 kbit/s stream ;
- for 4 800 bit/s there is no repetition of bits D within the 8 kbit/s stream ;
- for 2 400 bit/s each bit D is repeated twice within the 8 kbit/s stream (D1 D1 D2 D2 etc) ;
- for 1 200 bit/s each bit D is repeated four times within the 8 kbit/s stream (D1 D1 D1 D1 D2 D2 D2 D2 etc) ;
- for 600 bit/s each bit D is repeated eight times within the 8kbit/s stream (D1 D1 D1 D1 D1 D1 D1 D1 D2 D2 D2 D2 D2 D2 D2 D2 etc); |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.1.2 User rate Adaptation on the A-interface, AIUR greater than 38,4 kbit/s | For AIUR of 48 kbit/s, 56 kbit/s and 64 kbit/s one stream consisting of ITU-T V.110 32 bit frames or 64 bit frames, as specified in 3GPP TS 04.21 shall be transmitted over the A-interface. Splitting/Combining which occurs in the BSS, is as specified in 3GPP TS 04.21.
Table 11 gives the relation between the User Rate, Substream Rate Channel Coding and the Intermediate Rate. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.1.3 Relation between AIUR and the number of channels | Table11: Relationship between the AIUR, substream rate, channel coding, intermediate rate and number of channels
AIUR
Number of channels x Substream Rate
Channel Coding
(Multislot) intermediate Rate (Note1)
2,4 kbit/s
2-8 times duplication of each bit to reach 4,8 kbit/s
TCH/F4.8
8 kbit/s
4,8 kbit/s
4,8 kbit/s
TCH/F4.8
8 kbit/s
9,6 kbit/s
2X4,8 kbit/s
2XTCH/F4.8
8 kbit/s
9,6 kbit/s
9,6 kbit/s
TCH/F9.6
16 kbit/s
14,4 kbit/s
3X4,8 kbit/s
3XTCH/F4.8
8 kbit/s
14,4 kbit/s
2X9,6 kbit/s w/ padding
2XTCH/F9.6
16 kbit/s
19,2 kbit/s
4X4,8 kbit/s
4XTCH/F4.8
8 kbit/s
19,2 kbit/s
2X9,6 kbit/s
2XTCH/F9.6
16 kbit/s
28,8 kbit/s
3x9,6 kbit/s
3XTCH/F9.6
16 kbit/s
38,4 kbit/s
4X9,6 kbit/s
4XTCH/F9.6
16 kbit/s
48 kbit/s
5X9,6 kbit/s
5XTCH/F9.6
64 kbit/s
56 kbit/s
5X11,2 kbit/s
5XTCH/F9.6
64 kbit/s
64 kbit/s
66x11,2 kbit/s w/padd.
6XTCH/F9.6
64 kbit/s
NOTE: For AIURs 38,4 kbit/s this column indicates the multislot intermediate rate: for higher AIURs it indicates the intermediate rate. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.1.4 Handling of status bits X, SA, SB | In the single slot case, status bit SA shall be coded repeatedly as S1, S3, S6, S8, and SB is coded repeatedly as S4 and S9 in Figure 2. In the multislot case, status bit SA is coded repeatedly as S6, S8 and SB is coded as S9 in figures 2, 5 and 6.
The handling of the status bits shall comply with the synchronisation procedures for transparent services which are as described in 3GPP TS 29.007 (MSC), 3GPP TS 04.21 (BSS), 3GPP TS 27.001 (MS). |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.1.5 Handling of bits E1 to E7 | Bits E1 to E3 shall be used according to 04.21.
Bits E4 to E7 may be used for network independent clocking as indicated in 3GPP TS 04.21. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.2 TCH/F14.4, TCH/F28.8, and TCH/F32.0 channel codings | |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.2.1 User rate adaptation on the A interface, AIUR less than or equal to 56 kbit/s | The A-TRAU frame shall be used for transparent user data rates other than 32 kbit/s on the A interface. The A-TRAU frames are transmitted on up to four substreams multiplexed into one stream sent over the A interface. The split/combine function is applied on the substreams as specified in clause 7 of this TS. The relation between the AIUR and the number of channels is specified in table 12.
In a given A-TRAU frame on the A interface:
- for 14 400 bit/s there is no repetition of bits D within the 16 kbit/s stream in a given A-TRAU frame on the A interface.
The ITU-T I.460 rate adaptation is used for the transparent 32 kbit/s user rate on the A interface, i.e. four bits of each octet in the 64 kbit/s time slot are used for transporting the 32 kbit/s user data. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.2.2 User Rate Adaptation on the A-interface, AIUR greater than 56 kbit/s | For AIUR of 64 kbit/s one stream consisting of ITU-T V.110 32 bit frames or 64 bit frames, as specified in 3GPP TS 04.21 shall be transmitted over the A-interface. Splitting/Combining which occurs in the BSS, shall be as specified in 3GPP TS 04.21.
Table 12 gives the relation between the User Rate, Substream Rate Channel Coding and the Intermediate Rate. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.2.3 Relation between AIUR and the number of channels | Table 12: Relationship between the AIUR, AIUR per substream, channel coding, intermediate rate and number of substreams
AIUR
Number of substreams x
AIUR per substream
Channel Coding
Multislot intermediate Rate (note 1)
14,4 kbit/s
14,4 kbit/s
TCH/F14.4
16 kbit/s
28,8 kbit/s
2X14,4 kbit/s
TCH/F14.4
TCH/F28.8
16 kbit/s
32 kbit/s
1x32 kbit/s
TCH/F32.0
32 kbit/s
38,4 kbit/s
3X14,4 kbit/s w/padding
TCH/F14.4
16 kbit/s
48 kbit/s
4X14,4 kbit/s w/padding
TCH/F14.4
16 kbit/s
56 kbit/s
4X14,4 kbit/s w/padding
1x64.0 kbit/s (Note 2)
TCH/F14.4
TCH/F32.0
16 kbit/s
64 kbit/s
64kbit/s
5X14,4 kbit/s w/padding
1x64.0 kbit/s (Note 2)
TCH/F14.4
TCH/F32.0
64 kbit/s
NOTE 1: For AIURs 56 kbit/s this column indicates the multislot intermediate rate: for higher AIURs it indicates the intermediate rate.
NOTE 2: One substream over two air interface timeslots. No multislot intermediate rate. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.2.4 Handling of status bits X and SB | The X and SB bits shall be carried over the A interface in a multiframe structure as described in subclause 8.1.1.1 of 3GPP TS 04.21. SA bit is not carried over the A interface.
The handling of the status bits shall comply with the synchronisation procedures for transparent services which are as described in 3GPP TS 29.007 (MSC), 3GPP TS 04.21 (BSS), 3GPP TS 27.001 (MS). |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 17 Frame Formats | Octet
Bit number
No.
0
1
2
3
4
5
6
7
0
0
0
0
0
0
0
0
0
1
1
D1
D2
D3
D4
D5
D6
S1
2
1
D7
D8
D9
D10
D11
D12
X
3
1
D13
D14
D15
D16
D17
D18
S3
4
1
D19
D20
D21
D22
D23
D24
S4
5
1
E1
E2
E3
E4
E5
E6
E7
6
1
D25
D26
D27
D28
D29
D30
S6
7
1
D31
D32
D33
D34
D35
D36
X
8
1
D37
D38
D39
D40
D41
D42
S8
9
1
D43
D44
D45
D46
D47
D48
S9
Figure 2: The ITU-T V.110 80 bit frame for Transparent Data
octet
bit number
no.
0
1
2
3
4
5
6
7
0
0
0
0
0
0
0
0
0
1
1
D1
D2
D3
D4
D5
D6
D'1
2
1
D7
D8
D9
D10
D11
D12
D'2
3
1
D13
D14
D15
D16
D17
D18
D'3
4
1
D19
D20
D21
D22
D23
D24
D'4
5
1
E1
E2
E3
D'5
D'6
D'7
D'8
6
1
D25
D26
D27
D28
D29
D30
D'9
7
1
D31
D32
D33
D34
D35
D36
D'10
8
1
D37
D38
D39
D40
D41
D42
D'11
9
1
D43
D44
D45
D46
D47
D48
D'12
Figure 3: The modified ITU-T V.110 80 bit frame for Non-Transparent Data
D1
D2
D3
D4
D5
D6
D'1
D7
D8
D9
D10
D11
D12
D'2
D13
D14
D15
D16
D17
D18
D'3
D19
D20
D21
D22
D23
D24
D'4
D'5
D'6
D'7
D'8
D25
D26
D27
D28
D29
D30
D'9
D31
D32
D33
D34
D35
D36
D'10
D37
D38
D39
D40
D41
D42
D'11
D43
D44
D45
D46
D47
D48
D'12
Figure 4: Modified ITU-T V.110 60 bit frame for Non-Transparent Data
bit number
octet number
0
1
2
3
4
5
6
7
0
0
0
0
0
0
0
0
0
1
0
0
0
0
0
0
0
0
2
1
C1
C2
C3
C4
C5
M1
M2
3
Z1
D1
D2
D3
D4
D5
D6
D7
4
D8
D9
D10
D11
D12
D13
D14
D15
36 bit data field 1
5
D16
D17
D18
D19
D20
D21
D22
D23
6
D24
D25
D26
D27
D28
D29
D30
D31
7
D32
D33
D34
D35
D36
Z2
D1
D2
8
D3
D4
D5
D6
D7
D8
D9
D10
9
D11
D12
D13
D14
D15
D16
D17
D18
36 bit data field 2
10
D19
D20
D21
D22
D23
D24
D25
D26
11
D27
D28
D29
D30
D31
D32
D33
D34
12
D35
D36
Z3
D1
D2
D3
D4
D5
13
D6
D7
D8
D9
D10
D11
D12
D13
14
D14
D15
D16
D17
D18
D19
D20
D21
36 bit data field 3
15
D22
D23
D24
D25
D26
D27
D28
D29
16
D30
D31
D32
D33
D34
D35
D36
Z4
17
D1
D2
D3
D4
D5
D6
D7
D8
18
D9
D10
D11
D12
D13
D14
D15
D16
36 bit data field 4
19
D17
D18
D19
D20
D21
D22
D23
D24
20
D25
D26
D27
D28
D29
D30
D31
D32
21
D33
D34
D35
D36
Z5
D1
D2
D3
22
D4
D5
D6
D7
D8
D9
D10
D11
23
D12
D13
D14
D15
D16
D17
D18
D19
36 bit data field 5
24
D20
D21
D22
D23
D24
D25
D26
D27
25
D28
D29
D30
D31
D32
D33
D34
D35
26
D36
Z6
D1
D2
D3
D4
D5
D6
27
D7
D8
D9
D10
D11
D12
D13
D14
28
D15
D16
D17
D18
D19
D20
D21
D22
36 bit data field 6
29
D23
D24
D25
D26
D27
D28
D29
D30
30
D31
D32
D33
D34
D35
D36
Z7
D1
31
D2
D3
D4
D5
D6
D7
D8
D9
32
D10
D11
D12
D13
D14
D15
D16
D17
33
D18
D19
D20
D21
D22
D23
D24
D25
36 bit data field 7
34
D26
D27
D28
D29
D30
D31
D32
D33
35
D34
D35
D36
Z8
D1
D2
D3
D4
36
D5
D6
D7
D8
D9
D10
D11
D12
37
D13
D14
D15
D16
D17
D18
D19
D20
36 bit data field 8
38
D21
D22
D23
D24
D25
D26
D27
D28
39
D29
D30
D31
D32
D33
D34
D35
D36
Figure 5: A-TRAU 320 bit frame
octet
bit number
no.
0
1
2
3
4
5
6
7
0
0
0
0
0
0
0
0
0
1
1
D1
D2
D3
D4
D5
D6
S1
2
1
D7
D8
D9
D10
D11
D12
X
3
1
D13
D14
D15
D16
D17
D18
S3
4
1
D19
D20
D21
D22
D23
D24
S4
5
1
E1
E2
E3
E4
E5
E6
E7
6
1
1
1
1
1
1
1
S6
7
1
1
1
1
1
1
1
X
8
1
1
1
1
1
1
1
S8
9
1
1
1
1
1
1
1
S9
Figure 6: The modified ITU-T V.110 80 bit frame padded for 4,8 kbit/s transparent data at intermediate rate 16 kbit/s
Annex A (informative):
Frame Pattern Substitution
A.1 Special cases
If the sub frame starts with a Zseq, D1 is empty. With the above example, the resulting input and output sub frames are the following :
In the same case as above but with only one ZSP, the resulting input and output sub frames are the following:
A.2 False Z sequence detection
The Framing Pattern Substitution algorithm presented in subclause 10.2 ensures sure that all the Z sequences found in the original sub frame are removed, but it shall be checked that the transformations performed do not introduce new unwanted Z sequences.
The goal of this subclause is to show that the transformed sub frame does not contain new Z sequences introduced by the algorithm itself.
The coding of the ZSP is the key point to avoid such an emulation. The different cases are considered below.
1 : Sequence ZSP
The worst case is when the address is equal to 1 :
1
C
A0
A1
A2
A3
A4
1
1
0
0
0
0
0
1
1
There is a maximum of 5 zeroes.
2 : Sequence Di / ZSP.
By definition, a data block always ends up with a one (except the last one of the message) and the ZSP always starts with a 1.
3 : Sequence ZSP / Di
ZSP always ends up with a 1 and Di has a maximum of 7 zeroes : it is not possible to find 16 zeroes in a row.
4 : Sequence Di / Dj
Di is not the last data block of the message.
As already mentioned, Di ends up with a one (except the last one) : this is the same case as 3.
5 : Sequence Zi / D or D / Zi
This case only occurs when there is no substitution. In this case, the Zi bit close to the D field is always a one: this does not change the number of zeroes in sequence.
6 : Sequence last Di / new framing pattern
The last D sequence can end up with up to 7 zeroes, followed by the 16 zeroes of the next frame.
There is anyhow no ambiguity, when considering that the framing pattern is made up of 16 zeroes followed by a one.
7 : Sequence last Di / Z bit of the next sub frame
The last D sequence can end up with up to 7 zeroes, followed in the worst case by Z=0 and then a ZSP. As a ZSP starts with a one, this makes a maximum of 8 zeroes in a row.
8 : Sequence ZSP / ZSP (not shown on the figure)
This case arrives when the original message has at least 16 zeroes in a row.
As the ZSP element always starts and ends up with a one, this always induces two consecutive ones.
Annex B (informative):
Change History
Change history
Date
TSG #
TSG Doc.
CR
Rev
Subject/Comment
Old
New
s27
A005
Synchronisation
5.3.0
7.0.0
s29
A006
Introduction of EDGE channel codings into the specifications
7.0.0
8.0.0
s30
A007
Asymmetric channel coding
8.0.0
8.1.0
09-2000
TSG#09
NP-000551
A008
1
32 kbit/s UDI/RDI multimedia in GSM
8.1.0
8.2.0
12-2000
TSG#10
NP-000604
A009
Removal of 1200/75 bit/s data rate and clean-up
8.2.0
8.3.0
03-2001
TSG#11
NP-010040
A013
Correction of downgrading procedure for HSCSD
8.3.0
8.4.0 |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.1 Bearer capabilities | |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.1.1 Bearer capabilities | The EDGE radio interface shall be designed to work in all typical GSM radio environments like rural area (RA), typical urban (TU) and an indoor environment. EDGE shall also work in a Hilly Terrain (HT) environment however the main focus is on channels with lower delay spread than HT, as specified in GSM05.05.
The peak rates mentioned below may not be available in the full cell area. The radio interface should however be optimised to provide as much coverage/availability as possible.
In addition to peak data rates, the average throughput and the area where 384 kbps can be achieved are important measures and should be optimized. |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.1.1.1 Enhanced GPRS | EGPRS shall provide a range of bearer capabilities that depend upon the environment and user’s speed. The peak rate shall at least be:
EGPRS
Indoor/Low range outdoor
Urban/Suburban outdoor
Rural outdoor
EGPRS
384 kbps (48 kbps/timeslot)
384 kbps (48 kbps/timeslot)
144 kbps (18kbps/timeslot)
Speed
up to 10 km/h
up to 100 km/h
up to 250 km/h
Propagation conditions
Indoor, TU3
TU50
HT100
850/900MHz: RA250
1800/1900MHz: RA130
HT100 |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.1.1.2 Enhanced CSD | ECSD shall provide a range of bearer capabilities (per time slot) that depend upon the environment and user’s speed. The peak rates should at least be:
ECSD/T
Indoor/Low range outdoor
Urban/Suburban outdoor
Rural outdoor
ECSD/T
32 kbps/timeslot *)
32 kbps/timeslot *)
-
Speed
up to 10 km/h
up to 100 km/h
-
Propagation conditions
Indoor, TU3
TU50, HT100
-
ECSD/NT
Indoor/Low range outdoor
Urban/Suburban outdoor
Rural outdoor
ECSD/T
28.8 kbps/timeslot *)
28.8 kbps/timeslot *)
-
Speed
up to 10 km/h
up to 100 km/h
-
Propagation conditions
Indoor, TU3
TU50, HT100
-
*)The data rates are not defined yet by SMG2/4 and giving a rough indication
The maximum transfer delay including channel coding and decoding for ECSD/T shall be the same as for CSD/T.
Due to the current limitations of the core network, transmission of circuit switched data shall be limited to 64 kbps per user. |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.1.2 Bearer service attributes | The same bearer service attributes used for GPRS and CSD should be used for EGPRS and ECSD. Some new bearer service parameters relevant to the radio interface may be needed. For EGPRS the same QoS classes should apply. |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.1.3 Hand over/cell re-selection | The same hand over/cell re-selection mechanisms as for CSD/GPRS apply. Re-selection methods should allow the operator to optimise the service availability for EDGE users.
Seamless transition ECSD-> CSD -> ECSD and EGPRS classic->GPRS -> EGPRS classic.
Seamless transition from EGPRS Classic->EGPRS Compact->EGPRS Classic is possible when the Classic and Compact systems are time-synchronised.
EGPRS Compact mobile stations shall be able to re-select to a neighboring synchronized EGPRS Classic cell, and vice-versa. Re-selection between EGPRS Classic and EGPRS Compact at different frequency bands is desirable.
EDGE shall allow multi-band operation, i.e 850/900/1800/1900 MHz including E- and R-band.
Hand over should be supported for GSM850/900/1800/1900 and between GSM850/900/1800/1900.
14.1.4 Mobile Stations
EGPRS Compact mobile stations shall support EGPRS Classic.
UWCC operators require that EGPRS-Capable handsets shall support operation at both 850 and 1900 MHz, and shall support both EGPRS Classic and EGPRS Compact.
14.1.5 Link adaptation
Link adaptation should be provided to adapt the modulation and coding scheme to the radio channel conditions. This includes a reconfiguration of time slots (i.e. transparent 28.8kbps -> 2x14.4kbps) as well as fall-back to GMSK.
Measurements should be provided for efficient link adaptation for services and applications provided by EDGE. |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.2 Operational requirements | |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.2.1 Compatibility with services provided by present core networks | EDGE will enhance the GPRS and CSD service by providing higher data rates. That means, that EDGE will rely on underlying GSM functionality. |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.2.2 Operating environments | The operational scenario for EDGE includes international operation across various EDGE radio operating environments (850/900/1800/1900 MHz bands). Further, EDGE will support a variety of services with a range of bit rates. |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.2.3 Radio Access network planning | EDGE should not require a modification of actual frequency/coverage planning defined for GSM air interface when introduced. Frequency/coverage re-planning may be used to maximise the throughput in the system.
An EDGE network shall support at least 4/12, 3/9 and 1/3 frequency reuse patterns. EGPRS Compact shall support at least 1/3 frequency re-use. |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.2.4 Operators | All GSM900/1800/1900 operators should be able to deploy EDGE without licensing problems. |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.3 Efficient spectrum usage | |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.3.1 Spectral Efficiency | The spectral efficiency of EDGE should be significantly higher than in GSM.
The radio interface should be designed to maximise spectral efficiency.
EGPRS Compact shall be possible to implement in less than 1 MHz initial spectrum. |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.3.2 Spectrum utilisation | It should be possible to use EDGE in all GSM bands.
It should be possible to mix EDGE and non EDGE timeslots on the same carrier.
If simultaneous EDGE and non EDGE operation in the same time slot is required or not is for further investigation.
If EDGE is used on the BCCH carrier, it should not have any impact on BCCH power measurements. |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.3.3 Coverage/capacity | The peak service rates may be provided only in a limited coverage area. Link adaptation shall provide a mechanism to have a smooth degradation of the service rates for the outer cell areas.
EDGE should be designed to maximise the area where high data rates can be achieved.
SMG2 should define appropriate evaluation criteria. The throughput should be at least 384 kbps over 25% of the cell in both coverage and interference limited systems, with the simulation assumptions as described in the EDGE Feasibility Study”.
EDGE should be flexible to support a variety of initial coverage/capacity configurations, e.g. cell by cell deployment, and facilitate coverage/capacity evolution.
EGPRS Compact shall support more than one carrier in a given sector.
EGPRS Compact shall be able to grow non-uniformly, such that sectors may have a different number of EGPRS Compact carriers. |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.3.4 Evolution requirements | With EDGE several 3rd generation services can be provided in GSM. The technical parameters for EDGE should allow an evolution for coverage and capacity, as well as provisioning of future 3rd generation services. |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.3.4.1 Coverage evolution | The radio coverage for the EDGE may be:
- contiguous coverage;
- island coverage;
- spot coverage.
EDGE should be sufficiently flexible to support a variety of initial coverage configurations and facilitate coverage evolution. Coverage can be increased by deploying cell planning parameters optimised for EDGE usage and/or techniques like e.g. adaptive antennas, advanced power control, efficient resource allocation etc. |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.3.4.2 Capacity evolution | EDGE should facilitate the implementation and use of appropriate capacity improvement techniques, if applicable, in the various radio operating environments.
EDGE should not prevent capacity improvements, e.g. adaptive antennas, advanced power control, efficient resource allocation etc. It is desirable that the EDGE does not depend on the implementation of these techniques, but that they are capacity improvement options. It is desirable that they do not significantly add complexity or cost to the infrastructure or MSs. |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.4 Complexity / Cost | |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.4.1 Mobile complexity and cost | Hand portable and PCMCIA card sized EDGE terminals should be optimised in terms of size, weight, operating time, range, effective radiated power and cost/performance ratio. |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.4.2 Network complexity and cost | The cost/performance ratio of development and equipment should be kept at a reasonable level. |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.4.3 Mobile Station / Base Station types | The EDGE standard should support multislot operation for ECSD and EGPRS. It should be possible to provide a variety of Mobile Station as well as Base Station types of varying complexity, cost and capabilities in order to satisfy the needs of different types of users. The number of mobile classes should though be minimised. (The number and classes are for further study). |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.5 Requirements from bodies outside SMG | |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.5.1 Electromagnetic compatibility | The modulation characteristics have to be such that the degree of interference caused to other equipment is not higher than in today's systems. |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.5.2 RF Radiation effects | EDGE shall be operative at RF emission power levels which are in line with the recommendations related to electromagnetic radiation. |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.5.3 Security | The EDGE radio interface should be able to accommodate at least the same level of security as the GSM radio interface does. |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.6 Co-existence with other systems | A GSM system with EDGE should be capable to co-exist with a GSM system without EDGE deployment within the same or neighbouring band as well on neighbouring time slots. Furthermore should the performance of GSM channels without EDGE not be worsened by the fact that the neighbouring channel is a GSM channel with EDGE and vice versa.
A GSM or ANSI-136 system with EGPRS Compact should be able to co-exist with a GSM or ANSI-136 system without EGPRS Compact deployed within the same or neighbouring band. |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.7 Further Work Areas | |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.7.1 Services coordination | Possibility to have same or similar services in EDGE as in UMTS needs to be investigated. Investigation of hand-over between GSM and UMTS is necessary. |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.7.3 Measurements in existing GSM | It should be studied if there is a need to modify existing measurements for EDGE or other work items.
History
Document history
16th February 1998
First draft
20th February 1998
Update after Joint SMG1,2,3 & 4 EDGE workshop in Helsinki
27th April 1998
Update after SMG1 and SMG2 WPB#4
25th May 1998
Update after SMG4 and EDGE SMG2 working session
27th May 1998
Update after smg2 #26 decision, V1.0.0
3rd July 1998
Update after SMG4 workshop , V1.1.0
7th August 1998
Update after SMG1 plenary, V1.2.0
28th August 1998
Update after SMG2 EDGE workshop and SMG3WPA, V1.3.0, removal of open questions. expanded concept section
12th of October 1998
Update after SMG2WPA and SMG2 WPB, V1.4.0
2nd of November 1998
Update after EDGE WS, V1.5.0
2nd of December 1998
Update for EDGE WS #6, V1.6.0
14th December 1998
Update for SMG 7, V1.7.0
7th of January 1999
Update for SMG2 WPB, V1.8.0
25th of January 1999
Update for SMG2 plenary, V1.9.0
4th February 1999
Update for SMG plenary, V1.10.0
2nd March 1999
Update for SMG2 EDGE WS #7, V1.11.0
12th March 1999
Update for SMG2 #30, V1.12.0
31st May 1999
Updated for SMG2 #31, V1.13.0
18th June 1999
Updated for SMG EDGE WS #9, V1.14.0
24th August 1999
Updated for SMG EDGE WS #10, V1.15.0
19th September 1999
CR list for EDGE compact included, updated for SMG2 #32, V1.16.0
22th September 1999
COMPACT requirements included, V1.17.0
14th October 1999
Update with results from SMG2#32, V1.18.0
22th November 1999
Updated with results from workshop and other STCs, V1.19.0
10th January 2000
Updated with results from workshop and SMG #30bis, V1.20.0
23rd February 2000
Updated with the results from SMG2 and SMG#31, V1.21.0
2nd April 2000
Updated with results from EDGE WS #13, V1.22.0
22nd May 2000
Updated with results from SMG #31bis, V1.23.0
June 2000
Approved at SMG#32 Plenary
Rapporteur EDGE BSS: Frank Mueller, Ericsson
Email: [email protected]
Tel: +46-8-7570287
Fax:+46-8-7575550
Rapporteur EDGE NSS: Tommi Kokkola, Nokia
Email: [email protected]
Tel: +358-9-511 38358
Fax:+358-40-504 0734
Rapporteur GPRS 136 HS EDGE: Tommi Ljunggren, AT&T
Email: [email protected]
Tel: +46 8 754 0001
Fax: +46 8 754 0002 |
14ce710049a3c611ebb26f3972649957 | 21.978 | 1 Scope | The purpose of this Technical Report is to study the feasibility of a core network CAMEL server controlling voice services carried by VoIP within a GPRS PDP Context using:
1) An architecture based on ITU-T, H.323 family of recommendations.
2) An architecture based on IETF SIP specifications. |
14ce710049a3c611ebb26f3972649957 | 21.978 | 2 References | The following documents contain provisions, which through reference in this text, constitute provisions of the present document.
- References are either specific (identified by date of publication, edition number, version number, etc.) or non‑specific.
- For a specific reference, subsequent revisions do not apply.
- For a non-specific reference, the latest version applies.
- A non-specific reference to an ETS shall also be taken to refer to later versions published as an EN with the same number.
[1] IETF RFC 1889, A Transport Protocol for Real-Time Applications.
[2] ITU-T Q.931, Digital Subscriber Signalling System No. 1 (DSS 1) – ISDN User-Network Interface Layer 3 Specification For Basic Call Control.
[3] GSM 04.08
[4] IETF RFC 2543, SIP: Session Initiation Protocol.
[5] N2-99826, “Feasibility Technical Report on CAMEL control of VoIP Services”, 19th-23rd July 1999, Visby, Sweden.
[6] “Location of the SSF entity in GPRS network supporting real-time applications”, Contribution from Lucent Technologies circulated to supporters of CAMEL control over VOIP on 26th July 1999.
[7] S2-99528, “The inclusion of a ‘Service Switching Function’ in the reference multimedia architecture for the support IN based service features”, 26th – 20th July 1999, New Jersey, USA
[8] Tdoc65, “Comments to DTR 02004, concerning IP, fixed wireless and cellular mobile roaming scenarios and the conceptual IP cellular network model”, Motorola, ETSI TIPHON Tel Aviv 26-30th October 1998.
[9] TS 101 441 Vx.4.0 Draft A, GSM 03.78, CAMEL Phase 3, stage 2
[10] N2-99826, “Feasibility Technical Report on CAMEL control over VoIP Services”, 19th-23rd July 1999, Visby, Sweden.
[11] TS 101 441 Vx.4.0 Draft A, GSM 03.78, CAMEL Phase 3, stage 2.
[12] ETSI TIPHON 10, Temporary Document 65, Tel Aviv, 26-30 Oct.1998 |
14ce710049a3c611ebb26f3972649957 | 21.978 | 3 Introduction | The GSM CAMEL feature for control of operator specific services when roaming has evolved over a number of years and work on the latest Phase (Phase 3) is planned for completion as part of Release '99. Work in other 3GPP groups, particularly S2, is focused on developing a reference architecture for an all IP PLMN. One of the many requirements for 3G is to support roaming between 2G and 3G networks. Another requirement is to support a minimum set of 2G standardised supplementary services for roamers, such as call forwarding. The all IP PLMN is based on an evolved GPRS that supports voice over IP (VoIP).
VoIP calls may be established between 3G terminals or between a 3G terminal and legacy network terminal, such as 2G mobile, ISDN or PSTN. In the IP world there are currently two solutions that support VoIP; the H.323 family of recommendations defined by the ITU, and the Session Initiation Protocol (SIP) defined by the Internet Engineering Task Force (IETF). This Technical Report introduces these solutions and examines the feasibility of CAMEL controlling VoIP services in an all IP network with a view to supporting some 2G Supplementary Services and operator specific services or similar. Thus it is envisaged that CAMEL service control may be employed across 2G and 3G networks providing consistent services when roaming.
The information in the report is introduced below:
Section 5: briefly describes the proposal and functional model.
Section 6: addresses services that may be employed across 2G and 3G.
Section 7: describes the H.323 solution, architecture, message flows, state models, impact on standards, impact on services, multimedia evolution, advantages and disadvantages.
Section 8: describes the SIP solution, message flows, state models, impact on standards, impact on services, multimedia evolution, advantages and disadvantages.
Section 9: introduces work in other standards groups covering similar or related issues.
Section 10: records questions and answers from CN2a meeting presentations of earlier drafts
Section 11: describes conclusions and recommendations. |
14ce710049a3c611ebb26f3972649957 | 21.978 | 4 Definitions and Abbreviations | BCSM Basic Call State Model
CAMEL Customised Application for Mobile networks Enhanced Logic
CAP CAMEL Application Protocol
CSI CAMEL Subscriber Information
GGSN Gateway GPRS Support Node
GPRS General Packet Radio Service
HLR Home Location Register
IPSSF Internet Protocol Service Switching Function
MAP Mobile Application Part
MSC Mobile Switching Centre
MO Mobile Originating
MS Mobile Station
MT Mobile Terminating
O-CSI Originating CSI
PDP Packet Data Protocol
SGSN Serving GPRS Support Node
SIP Session Initiation Protocol
SS-CSI Supplementary Service CSI
T-CSI Terminating CSI
VLR Visitor Location Register |
14ce710049a3c611ebb26f3972649957 | 21.978 | 5 Proposition | An overall objective for this feasibility study is to demonstrate that CAMEL control of VoIP services in 3G networks can be readily specified and implemented by adapting standards and software used in 2G networks. This approach leads to services that function the same when a user roams between 2G and 3G networks, simplifies service evolution from 2G to 3G, and leads to more rapid implementation.
Figure 1: Functional Model for CAMEL Control of VoIP Services
Figure 1 illustrates the proposed functional model of the core network. It is derived from existing CAMEL specifications and 3GPP 23.121. In the latter technical specification, the Gatekeeper/SIP Proxy Server has been placed on the Gi interface to allow the Internet call control signalling to be transparent to the GPRS nodes, (SGSN and GGSN). Furthermore, if the signalling were to be non-transparent, and the SGSN were to handle the VoIP call control, the existing GPRS procedures, including the current CAMEL Phase 3 work would require changes. The reason is that if a subscriber roams outside the area served by the SGSN, the relationship between the gsmSCF and the SGSN in the old location is terminated and a new dialogue between the SGSN in the new location area and the gsmSCF is established. Such an approach would not allow existing CAMEL services to be re-used without substantial changes, unless techniques such as anchoring original SGSN are introduced.
The functions represented in the model are:
HLR: Release ’99 or later HLR
gsmSCF: Release ’99 or later CAMEL Service Control Function (gsmSCF)
gsmSSF: Release ’99 or later Service Switching Function
MSC Emulator and VLR: Modified Release ’99 GMSC, VMSC and VLR functions. Many processes, such as call control, database and billing are retained or enhanced. Circuit switching and ancillary processes are removed. H.323 or SIP server inter-working functions are added. This combination of functionality is referred to in this report as the 'IPSSF'.
The interface between the H.323 Gatekeeper/SIP Server and the 'IPSSF' call control processes must:
1 Carry sufficient call data for the gsmSSF to function correctly and to deliver the necessary information to the gsmSCF so that service logic decisions can be made.
2 Allow the gsmSCF (in combination with gsmSSF and MSC Emulator) to control VoIP calls (e.g. change ‘B’ party address) and manipulate call information (such as presentation number) similar to a GSM Release ’99 GMSC or VMSC.
H.323 Gatekeeper/SIP Proxy Server: – Either a H.323 Gatekeeper or a SIP Proxy server.
Gateway: - Provides interworking between packet network and external circuit switched networks such as PSTN or ISDN.
GGSN: – Release '99 or later Gateway GPRS Support Node
Notes:
Mobile Terminated (MT) Speech Calls: MT VoIP call states are modelled using a CAMEL Phase 3 or later T_BCSM as used for circuit switched MT calls.
Mobile Originated (MT) Speech Calls: MO VoIP call states are modelled using a CAMEL Phase 3 or later O_BCSM as for circuit switched MO calls.
Physical Location: The H.323 Gatekeeper/SIP Server and 'IPSSF' may be located in a home network or a visited network as both MAP signalling and CAP signalling are standardised for international use.
Physical Realisation: From the control of VoIP viewpoint the IPSSF and the H.323 Gatekeeper/SIP Server may be combined in one network entity or may be separate in separate network entities. If they are separate, standardisation of the interface may be required.
IP PDU Routing: Routing of IP call control packets to/from the H.323 Gatekeeper/SIP Proxy Server is not fully addressed in this work. It is simply assumed that appropriate addressing and routing takes place. |
14ce710049a3c611ebb26f3972649957 | 21.978 | 5.1 Relationship with Architecture work in S2 | Ongoing architecture work in S2 for 3G is documented in TS 23.121 and in the Technical Report on 'Architecture for an all IP network' (TR 23.922 V1.0.0). The work in this CAMEL control of VoIP services FTR is intended to aid/complement S2 work by analysing feasibility at a somewhat lower level of abstraction, focusing on functional/protocol capabilities. The IPSSF in the functional model described above may form part of a Service Capability Server (SCS) in the Multimedia architecture PS Domain' described in TS 23.121. Similarly, it may form part of the Call State Control Function (CSCF) described in the all IP TR. The exact mapping between the Functional Model used in this report and the architectures developed by S2 requires further study and is work probably most appropriate to S2. |
14ce710049a3c611ebb26f3972649957 | 21.978 | 6 Service Requirements | In 3GPP, SA1 has responsibility for the definition of service requirements. Requirements for Release '00 are at a very early stage of development. When considering feasibility it is very useful to have in mind the services that a 3G network may be required to support. This section lists many example 2G network voice services that may be required by 3G VoIP customers, including some new 3G network possibilities, such as ‘high quality audio’. Other services are being considered but not included here because the focus of this report is mainly on how existing 2G services may be support on an IP only network. Many of the services listed below are not supported by H.323 or SIP. They may be supported in the future. Services that early implementations may be required to support are show in italics. Detailed study is necessary to identify the services that require CAMEL control as it may be possible to re-use the VLR and MSC Emulator capabilities for some services without the need for CAMEL control, such as Operator Determined Barring. Some services may be more appropriate to implementation in the Terminal not the network, such as Call Hold. |
14ce710049a3c611ebb26f3972649957 | 21.978 | 6.1 Basic Services | Example basic services that may require CAMEL control in a 3G network:
- Speech
- Emergency calls
- Low bit rate data
- Medium bit rate data
- High bit rate data
- High quality audio
- Low bandwidth video
- High bandwidth video |
14ce710049a3c611ebb26f3972649957 | 21.978 | 6.2 Supplementary Services | Example 2G supplementary services that may require CAMEL control in a 3G network:
- Operator determined barring
- User defined barring
- Call screening
- Call deflection
- Call forwarding unconditional
- Call forwarding on busy, no reply and not reachable
- Call waiting
- Call hold
- Call transfer
- Calling number identification presentation/restriction
- Connected number identification presentation/restriction
- Multiple Subscriber Profile
- Multi-party
- Call Completion Services (e.g. CCBS)
- Closed user group
- Advise of charge
- Calling name presentation |
14ce710049a3c611ebb26f3972649957 | 21.978 | 6.3 Operator Specific Services | Example 2G operator specific services that may require CAMEL control in a 3G network:
- Short number dialling
- Prepay
- VPN |
14ce710049a3c611ebb26f3972649957 | 21.978 | 6.4 Other Services | Example services not listed in the ‘Basic Services’, ‘Supplementary Services’ or ‘Operator Specific Services categories that may require CAMEL control in a 3G network:
- Lawful interception
- Voice group-call service
- Voice broadcast service
- SMS
- Fax
- ASCI
- MExE
- Location Services
- SoLSA |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7 H.323 Solution | |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.1 Description | ITU-T defines a family of recommendations known as H.323 for multimedia communications over packet based networks. The majority of VoIP products available today support H.323.
Figure 2: H.323 Environment
The H.323 environment is illustrated in Figure 2. A 'terminal' is a network endpoint that provides real-time, voice or multimedia communications with another Terminal or a Gateway. A Gateway is a network endpoint that provides the necessary translation services to allow H.323 terminals to communicate with terminals on other networks, such as the ISDN. A Gatekeeper is a network entity that provides address translation and access control. It may also provide bandwidth management. Gatekeepers are optional. A Multipoint Control Unit (MCU) is a network endpoint that provides the capability for VoIP conference calls. All terminals participating in a conference establish a connection with the MCU. It manages resources and negotiates between terminals to determine which video and audio codec to use.
Figure 3: H.323 Protocol Architecture
Figure 4: H.323 Protocol Summary
Figure 5: Simple Example H.225 Gatekeeper Routed Call Setup
Addressing: It is common for Internet addresses to be assigned dynamically which means that an IP addresses cannot be used to identify a user. One way of overcoming this problem is for users to register with a Gatekeeper . Once registered, other users can reach them by sending a location request message to the Gatekeeper. To accomplish roaming and IN service convergence a temporary ETSI TIPHON document reference[12] identifies a special type of Gatekeeper called a 'Mediation Gatekeeper' or 'MGK'. The user is assigned a unique ID (or 'alias') in the MGK, such as an MSISDN or an IP address. However, there are other methods that can be used. In this TR, H.323 roaming users are assumed to register with a Gatekeeper in a visited network and the alias is registered in the HLR.
H.225, GSM and IN Integration: H.225 messages are based on Q.931. GSM (04.08) call control processes are similar to Q.931. ITU-T IN Capability Sets (e.g. IN CS-1) are based around ISDN call control. Q.931 is the ISDN user-network interface layer 3 specification for basic call control. This common ancestry in ISDN and Q.931 especially, leads to a view that integration based on a common call control model may be feasible. H.323 supports an architecture where all call control signalling messages between terminals are routed via a Gatekeeper. A Gatekeeper seems to be an obvious choice when attempting to integrate H.225, GSM and IN (i.e. CAMEL). |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.2 Architecture | |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.2.1 Introduction | This section of the report provides information flows that illustrate simple MO and MT calls with CAMEL interactions. It is based upon an architecture that proposed in references [6] and [7]. The objective is to provide further understanding of the proposed architecture and to further progress the work in the feasibility study. |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.2.2 Assumptions | a. The call flows presented are based on using the ITU-T H.323 protocol between the Mobile Station (MS) and the Gatekeeper.
b. The gatekeeper and the IPSSF have been co-located in order to avoid any showing information flows between the two entities. Standardisation of the information flows between these entities is for further study.
c. The attach/detach and establishment of (MS initiated or network initiated) PDP context is based on existing GPRS procedures as defined in UMTS 24.008 and UMTS 29.060. No changes are assumed.
d. The information flows make no consideration for interworking with other networks such as the PSTN/ISDN and no media gateways or signalling gateways are shown. This is considered to be outside the scope of the feasibility study.
e. For this work, it is assumed that roaming users register with a gatekeeper in the visited network. It is acknowledged that there are several proposals in other standard bodies investigating other alternatives involving for example registration with home gatekeeper whilst roaming.
f. The addressing mechanism used to identify called parties provides a means to identify the home GPRS network. An incoming call request can be forwarded to the mediation gatekeeper in the home network of the called user. The mediation gatekeeper can query the HLR to identify the gatekeeper that the called used is registered with. It is acknowledged that there are other alternatives being investigated by the various standards groups. |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.2.3 Functional Architecture | The proposed functional architecture is discussed in [6] and is shown in Figure 6. The main idea behind this functional architecture is to be able to reuse existing (circuit switched) CAMEL services for VoIP.
Figure 6: Proposed functional architecture |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.3 Message Flows | |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.3.1 Registration | It is not within the scope of this feasibility study to investigate or select the registration and location update process that must exist within a multimedia service network. However this section outlines a possible registration process that allows CAMEL Subscriber Information (CSI) to be stored in the Gatekeeper/IPSSF. Two distinct levels of registration exist; one at the PDP transport capability (Attach/Detach and PDP context establishment) and another level at the VoIP/multimedia level. Within H.323, this latter registration takes places with a Gatekeeper and roaming users register with a gatekeeper in the visited network. Gatekeeper discovery can be achieved via H.323 gatekeeper discovery procedure or via the resolution of ras://gk via DNS.
Figure 7 outlines a proposed registration process. MS is a mobile station in a visited network.
{1} The MS attaches to the network via existing GPRS procedures. This involves an attach request to the SGSN and a location update sequence from the SGSN to the HLR.
{2} The MS activates a PDP context to establish an IP session with the gatekeeper.
{3} The MS checks that it can register with that particular gatekeeper using a RAS GatekeeperRQ message. Assuming that the request is accepted, the gatekeeper confirms that registration can take place.
{4} Upon confirmation that the MS is allowed to register with the gatekeeper, a RAS Registration Request is sent.
{5} At this point, the gatekeeper sends a location update to the HLR in the user’s home network to register the alias address. The HLR responds with an InsertSubscriberData message that contains the CAMEL subscription Information (CSI). At this stage it is assumed that the Originating CSI (O-CSI) and the Terminating CSI (T-CSI) is sent to the gatekeeper. Supplementary Service CSI (SS-CSI) applicability is for further study. The HLR keeps a record of the address of the gatekeeper that the MS is registered with.
{6} Once the registration process is complete, the PDP session with the gatekeeper may be terminated.
Figure 7: MS registration with gatekeeper |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.3.2 MO call requiring CAMEL interaction | The call flows for a mobile originated (MO) calls are shown in Figure 8 and are further explained below:
{1} The MS wishes to place a VoIP call. A PDP context must be established to allow an IP session to be set up.
{2} The MS checks that it is allowed to place a call via RAS Admission Request message.
{3} If the MS is allowed to place the call (gatekeeper routed) an H.225 Setup message is sent to the gatekeeper.
{4} When the Setup message is received at the gatekeeper, if the O-CSI for the calling user is present, it would be possible to analyse its contents and if necessary invoke the IPSSF. The IPSSF is now assumed capable of implementing the CAMEL O-BCSM. Later sections in this report cover the mapping of the O-BCSM points in call and H.323 call state model. The IPSSF starts a dialogue with the gsmSCF. The gsmSCF address and service key to invoke are obtained from the triggering criteria in the O-CSI.
{5} The gsmSCF sends the instructions to the IPSSF according to the service logic invoked
{6} The gatekeeper forwards the call according to the instructions received from the gsmSCF. H.225 Setup message is sent to the destination address.
{7} Dialogue between the gsmSCF and the IPSSF may continue according to the service logic. The remainder of the information flows will vary according to the service logic and are not shown.
Figure 8: MO call with CAMEL interaction |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.3.3 MT call requiring CAMEL interaction | The call flows for mobile terminated (MT) calls are shown in Figure 9 and are further explained below:
{1} An incoming H225 Setup message is received by the mediation gatekeeper in the home network of the called subscriber. (The incoming call could be from an other H.323 gatekeeper or H.323 signalling gateway representing a call originating from external networks such as the PSTN).
{2} The mediation gatekeeper queries the HLR to discover the IP address of the gatekeeper that the called user is registered with. SendRoutingInformation/Ack messages can be used for this purpose.
{3} The mediation gatekeeper forwards the call to the required gatekeeper by sending a Setup message.
{4} The gatekeeper receives the H225 Setup message and checks the T-CSI of the called user to determine whether any CAMEL services should be invoked.
{5} If analysis of the T-CSI shows that the triggering criteria are met, the IPSSF is invoked to create a T-BCSM and the IPSSF initiates a dialogue with the gsmSCF in the home network of the called user. Instructions are received from the gsmSCF on how the call is to proceed.
{6} The gatekeeper will route the call according to the instructions received from the gsmSCF and will send an H225 Setup message to the destination party
{7} Dialogue between the gsmSCF and the IPSSF may continue according to the service logic. The remainder of the information flows will vary according tot he service logic and are not shown.
Figure 9: MT Call with CAMEL interaction |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.4 State Models | |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.4.1 H.323 and CAMEL O-BCSM | Figures 10, 11 and 12 show the relationship between the H.323 protocol state model and the points in call (PICs) defined in the CAMEL phase 3 Originating Basic Call State Model (O-BCSM) for successful and unsuccessful call attempts. The O-BCSM represented is based on CAMEL phase 3 state models as described in reference [9]. For the sake of simplicity, the information flows between the GPRS network nodes (GGSN and SGSN), the gsmSCF and Gatekeeper/SSF are not shown in any of the Figures. The sequence of messages flows relating to the SGSN, GGSN and the gsmSCF are as covered in section 7.3.2 and Figure 8. An O-BCSM object can be created upon receipt of an H.225 Setup message and the analysis of the O-CSI.
The figures assume that the MS has previously registered with a gatekeeper in the network. The destination entity shown can be an other MS, gatekeeper, mediation gatekeeper or H.323 signalling gateway. As the call control protocol, H225 used in an H.323 environment is based on Q931, the mapping between the messages and the PICs are similar to existing GSM mechanisms. |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.4.2 Successful call establishment | Figure 10 shows the PICs and relationship to H.323 messages for a successful call establishment of MO call with CAMEL interaction.
{1} The H.225 Setup message indicates that a dialled number is received from the MS and detects that an O-CSI exists for this subscriber.
{2} Analysis of the O-CSI takes place and gsmSCF is invoked.
{3} Instructions are received from the gsmSCF to route the call to the destination address.
{4} Upon receipt of the Connect message from the called party, a transition to the PIC O_Active takes place. O_Answer DP can be reported.
{5} Either called or calling party can release the call with a ReleaseComplete message, upon which a transition to the PIC O_Null&Authorise_Origination_Attempt_Collect_Info takes place and the DP O_Disconnect may be reported.
Figure 10: H.323 message flows and corresponding CAMEL O-BCSM for successful call establishment |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.4.3 Unsuccessful call establishment due to gatekeeper admission reject | Figure 11 shows the PICs and relationship to H.323 messages for the situation where the destination is refused permission to accept the incoming call by the gatekeeper. An AdmissionReject message may be sent by the gatekeeper to the destination is registered with for reasons specified in RAS procedures. In order to keep the Figures relatively simple, the PICs O_Null&Authorise.. and Analyse_Info and corresponding message flows are not shown.
{1} Instructions received from the gsmSCF on routing of the call and an H.225 Setup message is sent to the destination.
{2} The destination is not given permission by the gatekeeper to accept the incoming call. A number of reasons could exist, such as insufficient bandwidth, permission expired etc. This information is carried in the rejection reason of the AdmissionReject and transported into the ReleaseComplete message back to the gatekeeper.
{3} ReleaseComplete message arrives at gatekeeper indicating that the call could not be completed. DPs Route_Select_Failure or O_Routing_and_Alerting_Failure may be reported according to the rejection reason provided in the RelaseComplete message.
Figure 11: H.323 messages flows and corresponding CAMEL O-BCSM for the case where the destination is refused permission to accept the call by the gatekeeper |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.4.4 Unsuccessful call establishment due to refusal by destination party | Figure 12 shows the PICs and relationship to H.323 messages for the situation where the destination does not accept the incoming call request. Once again, in order to keep the Figures relatively simple, the PICs O_Null&Authorise.. and Analyse_Info and corresponding message flows are not shown.
{1} Instructions received from the gsmSCF on routing of call and an H225 Setup message is sent to the destination.
{2} The gatekeeper receives a ReleaseComplete message with a release reason. Potentially, depending on the release reason, the DPs O_No_Answer, O_Busy, O_Abandon, O_Routing_and_Alerting_Failure or Route_Select_Failure may be reported.
Figure 12: H.323 messages flows and corresponding CAMEL O-BCSM for the unsuccessful establishment (other than admission refusal) |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.4.5 H.323 and CAMEL T-BCSM | Figures 13 and 14 show the relationship between the H.323 protocol state model and the points in call (PICs) defined in the CAMEL phase 3 Terminating Basic Call State Model (T-BCSM) for successful and unsuccessful MT call attempts. The T-BCSM represented is based on CAMEL phase 3 state models as described in reference [9]. Once again, for the sake of simplicity, the information flows between the GPRS network nodes (GGSN and SGSN), the gsmSCF and Gatekeeper/SSF are not shown in any of the Figures. The sequence of messages flows relating to the SGSN, GGSN and the gsmSCF are as covered in section 7.3.3 and Figure 9. A T-BCSM object can be created upon receipt of an H.225 Setup message and the analysis of the T-CSI. |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.4.6 Successful MT call delivery | Figure 13 shows the PICS and relationship to the H.323 protocol state model for a successfully established MT call requiring CAMEL interaction.
{1} The H.225 Setup message arrives at the gatekeeper the destination MS is register with. If a T-CSI exists for the subscriber, a T-BCSM instance is created.
{2} Analysis of the T-CSI takes place and gsmSCF is invoked
{3} Instructions from the gsmSCF are received and upon receipt of the Connect message from the called MS a transition to T_Answer takes place. The DP T_Answer may be reported.
{4} Either calling or called party can release the call with a ReleaseComplete message, upon which a transition to PIC T_Null takes place with potentially the DP T_Disconnect being reported.
Figure 13: H.323 message flows and corresponding CAMEL T-BCSM for successful call establishment |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.4.7 Unsuccessful MT call delivery | Figure 14 shows the PICs and relationship to H.323 message for the situation where the terminating MS does not accept the incoming call.
{1} The H.225 Setup message arrives at the gatekeeper the destination MS is register with. If a T-CSI exists for the subscriber, a T-BCSM instance is created.
{2} Analysis of the T-CSI takes place and gsmSCF is invoked
{3} The called party refuses to accept the call and sends a ReleaseComplete message to the gatekeeper. The ReleaseComplete message may contain release reason. Potentially, depending on the release reason, the DPs T_No_Answer, T_Busy, T_Abandon or T_call_handling_Failure may be reported.
Figure 14: H.323 message flows and corresponding CAMEL T-BCSM for unsuccessful call establishment |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.5 CAMEL Integration | Integration of CAMEL functions with H.323 Gatekeeper functions may require enhancements to H.323 standards or CAMEL standards. Some initial work has been done to identify the standards that may need to be enhanced and to assess the extent of the changes required. |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.5.1 Impact on H.323 Standards | Section 7.4 describes and illustrates how H.225 messages may be mapped to O_BCSM and T_BCSM Detection Points (DPs) at the Gatekeeper/SSF. When studying the Figures it can be seen that some CAMEL Phase 3 DPs (such as O_Active) have good correspondence with H.225 messages at the Gatekeeper, while other DPs require more detailed analysis. At this level of analysis it seems likely that changes to H.323 may not be needed.
As the H.323 standards are already well established it seems likely that changes would be unwelcome in and it seems more likely that, if necessary, some reduction in CAMEL functionality may be a more acceptable compromise. |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.5.2 Impact on CAMEL and other UMTS Standards | It is likely the some functional modelling additions and some CAP message information element additions will be needed. It is also likely that some HLR/VLR data and MAP protocol additions may be needed. It is of course very difficult to assess and describe exactly how much effort and meeting time is required to effect these changes. In comparison to previous CAMEL work on Phase 1, Phase 2 and Phase 3, in the author's opinion the effort required is probably more than was necessary to complete GPRS inter-working in CAMEL Phase 3, but probably less than the work necessary to complete CAMEL Phase 1.
The major impact of these changes concerns documents 22.078, 23.018, 23.078, 29.078 and 29.002. |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.6 Service Impacts | The combination of CAMEL and H.323 (as described in this report) may support some GSM supplementary services, such as unconditional call forwarding (subject to some re-engineering) and perhaps most operator specific services (depends on the extent of any reduced functionality necessary for integration). Every service needs to be studied in detail to determine exactly what can be supported. Interworking with H.323 based supplementary services is another aspect. Again every service needs to be studied in detail, however, it seems likely that a comparable level of inter-working may be possible to that achieved with a combination of an MSC and ISDN, i.e. low-level services like CLI probably are feasible but more complex services may not be feasible. |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.7 Multimedia Evolution | The proposal in this report is focused on voice not multimedia. Control of multimedia services requires further study. However, the proposal does not in any way preclude CAMEL evolution to control multimedia services supported by the H.323 family of recommendations. It is likely that further enhancements to the protocols and functions listed in section 7.5.2 (e.g.. MAP, CAP, HLR, VLR, CSE) may be necessary, depending on the nature of the multimedia service control required. |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.8 Advantages | The following advantages have been identified:
Maximises the re-use of existing functional entities, protocols and services. Such reuse decreases the development and ownership costs allowing existing familiar 2G services to be provided to 3G subscribers at an early stage.
Minimum changes to the CSE for the support of legacy services. There are several IN/CAMEL services already deployed such as PrePaid, VPN, Mobile Number Portability etc., which may be required in a voice over IP network.
This approach is in line with the work currently underway in ETSI SPAN 3 (Services and Protocols), in particular a work item addressing IN support for voice over IP in the H.323 architecture and associated protocols in association with the TIPHON project. The study will investigate how an H.323 gatekeeper can act as a virtual Service Switching Point (SSP). It is worth noting that ETSI plan to harmonise the fixed line IN protocol (ETSI Core INAP CS3.1) and mobile equivalent (CAP Phase 3) into a common protocol targeted for ETSI Core INAP CS4. |
14ce710049a3c611ebb26f3972649957 | 21.978 | 7.9 Disadvantages | The following disadvantages have been identified:
Introduces new functional entity ‘IPSSF’, which provides the necessary mapping between the Gatekeeper and the CSE. However, this functional entity is based on the functions already provided by a VMSC/GMSC, where already standardised process such as the gsmSSF can be reused.
The interface between the SCS/CSCF and the IPSSF requires further study. |
14ce710049a3c611ebb26f3972649957 | 21.978 | 8 SIP Solution | |
14ce710049a3c611ebb26f3972649957 | 21.978 | 8.1 Description | The IETF Multiparty Multimedia Session Control (MMUSIC) working group specifies an IP telephony architecture (Figure 15). The architecture is seen as an alternative to ITU-T H.323 family of recommendations that is simpler to implement. It allows for the possibility of interworking with H.323 terminals. However, H.323 currently has the greatest industry backing.
The Session Description Protocol (SDP) can be likened to H.245 (channel open/close and terminal capability handling). Session descriptions have a list format containing information about the session (see RFC 2327). The Session Initiation Protocol (SIP), reference [4], can be likened to H.225 (Registration, Admission, Status (RAS), and Q.931 messages). Message headers are in plain text and look similar to Email headers.
SIP uses a client server model similar to the Hypertext Transfer Protocol (HTTP) and many others (Figure 16 (a)). It is used in conjunction with other protocols such as SDP, RTP and RSVP. SIP can establish connections via TCP or UDP (Figure 16 (b)).
Figure 15: IETF IP Telephony Architecture
Figure 16: (a) Client/Server Model using TCP, (b) Protocol Stack using UDP/TCP
SIP Messages: To initiate a session a client sends an INVITE message to a server. An INVITE message typically contains a session description in SDP sufficient to establish communication. SIP Request and Response messages are listed below.
SIP Request Messages
SIP Response Messages
INVITE
ACK
BYE
CANCEL
OPTIONS
REGISTER
1xx Informational
2xx Success
3xx Redirection
4xx Client Error
5xx Server Error
6xx Global Failure
INVITE - Invites client or server to establish a session.
ACK - Confirmation reception of a final response to an INVITE message.
BYE - The sender wishes to close the session.
CANCEL - Cancels pending requests.
OPTIONS - Asks for information about capabilities before establishing a session.
REGISTER - Informs a Location Server of the client's IP address.
SIP response messages use a 3 digit number, e.g. 1xx. The first digit defines the category. The next two digits allow up to 100 variations, e.g. 200 OK (successful invitation).
Session Establishment: Sessions may be established using direct point-to-point communication or by using a SIP Server for personal mobility. A SIP server may be a Redirection Server or a Proxy Server. To establish a session using a SIP server the originator sends an INVITE message to the server. The server communicates with a Location Server to retrieve the IP address of the terminating party. When a Redirection Server is employed (Figure 17) the IP address is passed to the originator. The originator sends a new INVITE message and a session is established with the terminating party. When a Proxy Server is employed (Figure 18) the address of the terminating party is not passed to the originator. A Session is established between the originator and terminating party via the Proxy Server. One or more intermediate Proxy Servers may take part in the session. It is envisaged that a Proxy Server may be a network entity where CAMEL service control could be applied. This possibility is investigated further in this report.
Figure 17: Session Establishment Using a Redirection Server
Figure 18: Session Establishment Using a Proxy Server
Interworking with H.323: RFC 2543 states (reference [4]) that SIP could be used to determine that a party can be reached via H.323, obtain the H.245 gateway and user address and then use H.225.0 to establish the call. |
14ce710049a3c611ebb26f3972649957 | 21.978 | 8.2 Architecture | |
14ce710049a3c611ebb26f3972649957 | 21.978 | 8.2.1 Introduction | This section of the report provides information flows that illustrate the possible interaction of CAMEL and SIP. In particular it provides a proposal for the triggering of CAMEL services as well as a mapping between the CAMEL call states and the call states of the Session Initiation Protocol (SIP).
An overall objective for the feasibility study is to demonstrate that CAMEL control of VoIP services in 3G networks can be readily specified and implemented by adapting standards and software used in 2G networks. This approach leads to services that function the same when a user roams between 2G and 3G networks, simplifies service evolution from 2G to 3G, and leads to more rapid implementation. This section of the report investigates the possibility of CAMEL service control based on the SIP proxy Server approach, as described in companion contributions (or sections that will form part of the technical report). This means that a locally configured proxy server is required for outgoing calls that require legacy service support based on existing CAMEL services.
The section of the report is organised as follows: Section 8.2.2 outlines the proposed functional architecture for the support of CAMEL/SIP interaction. Section 8.2.3 briefly describes the concepts for IN service triggering based on CAMEL Subscription Information. Section 8.3.1 describes a registration process, section 8.3.2 deals with the detail of triggering services for Mobile Originated Calls, section 8.3.3 deals with the details for triggering Mobile Terminated calls. Section 8.4 describes the mapping of the SIP protocol state to the CAMEL basic call state model for the Originating and the Terminating sides. |
14ce710049a3c611ebb26f3972649957 | 21.978 | 8.2.2 Functional Architecture | The proposed functional architecture is derived from various companion contributions relating to the technical report. It is provided here for completeness. The concept of the ‘IPSSF’ is introduced which acts as an overlay between the IP telephony call control and the Intelligent Network layer provided by the CAMEL Service Environment (CSE) or the GSM Service Control Function (gsmSCF). This ‘IPSSF’ provides the necessary mapping between the SIP protocol state machine and the CAMEL Basic Call State Model (BCSM). Figure 19 outlines the proposed functional architecture at the network level.
Figure 19: Proposed functional Architecture to support CAMEL control of VoIP based on SIP call control |
14ce710049a3c611ebb26f3972649957 | 21.978 | 8.2.3 Basic concept of the proposal | Subscribers may register in the visited SIP network allowing the subscriber to receive incoming calls. A subscriber may use the MSISDN as an additional identifier in the registration process. Upon registration with the server, the CAMEL subscription information for the subscriber is sent to the IPSSF by the HLR in the subscriber’s home network. As incoming calls made to the subscriber terminate at the server the subscriber is registered with, the Terminating CAMEL Subscription Information (T-CSI) may be examined and if necessary the gsmSCF may be invoked on a per incoming call basis. Similarly, calls made by a subscriber already registered with a proxy server allow the Originating CAMEL (O-CSI) subscription information to be examined and potentially allow the gsmSCF to be invoked. Callers not registered will not have any O-CSI information in the proxy server they are using to place the call. The proposal here is as follows: when the initial call request message (or the INVITE method) is received by the SIP proxy server, the IPSSF/VLR establishes a dialogue with the HLR of the home subscribers network to allow the CAMEL subscription information to sent. The O-CSI may then be examined and if necessary the gsmSCF may be invoked. |
14ce710049a3c611ebb26f3972649957 | 21.978 | 8.2.4 Assumptions | a. All the call flows show that the SIP Proxy Server and the IPSSF have been co-located in order to avoid showing an information flows between the two entities. Standardisation of the messages for this interface is for further study.
b. The attach/detach and establishment of the Packet Data Protocol (PDP) session are based on existing GPRS procedures found in UMTS 24.080 and UMTS 29.060.
c. When a subscriber registers or places a call via a SIP Proxy server, the home network of the subscriber can be identified.
d. A subscriber may use the MSISDN as an additional identifier in the registration process.
e. Originating and terminating SIP Proxy servers must operate in a call-state aware mode.
f. As registration with a SIP Proxy server is not mandatory, it shall be possible to determine whether a registration exists for that particular subscriber when an incoming call is placed by a subscriber. This allows the CSI information to be fetched from the HLR is the subscriber is not registered. (Note: Absence of the O-CSI does not necessarily mean that the user is not registered, merely that the O-CSI may not exist for that subscriber).
g. The information flows make no consideration for interworking with other networks (e.g. PSTN via gateways as this is considered to be outside to scope of the technical report. |
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