This document provides guidelines for core network design and planning. It discusses key capacity concepts, emerging technologies, network topologies, and an approach for engineering network elements. The document contains two case studies applying these guidelines. It aims to help network operators effectively plan their core networks.
This document provides guidelines for core network design and planning. It discusses key capacity concepts, emerging technologies, network topologies, and an approach for engineering network elements. The document contains two case studies applying these guidelines. It aims to help network operators effectively plan their core networks.
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Completion of the following Table is the agreement and authorization to implement the contents of this document.
Name Department Title Signature Date
Revision History
Date Issue Author Reason Description Of Change Nov 05 S Ramesh V 1.0 First Draft Nov 06 S Ramesh UMTS change incorporated, M2UA, M3UA Incl. addtl. Case Study V 2.0 Second Draft
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1. SETTING 1.1. AUDIENCE 1.1.1. PRIMARY Regional engineering teams who will be required to Design the Core Networks Solution for the purpose of Bidding and raising Proposals catering to customer specific needs 1.2. CONTENT INITIAL VERSION 1.2.1 First draft version of Core network designing guidelines. It includes guidelines that can be applied on excel sheet used to dimension the core network. SECOND VERSION 1.2.2 - Detailed description of Design guidelines. Includes processes and deliverables. The most detailed document around the service in question. Multiple-page document in black and white a technical paper. Covers UMTS core dimensioning details with MSC server and Media Gateway architecture. 1.3. ROLES Development Champion: Initial content creation according to template below. Final technical approval on finished document. Services Product Manager: Edit and contribute as required to fulfill content requirements for defined audience. This includes editing for the Development Champion and final approval on finished proof.
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2. INTRODUCTION TO CORE NETWORK DESIGN AND PLANNING 2.1. NEED FOR PLANNING Core network Design and Planning is a complex process particularly if the size of the networks become large. This is so as all the networked elements are almost fully meshed logically and there is an end-to-end dependency causing any change in the core network at a single element impact many other elements as well. Thus core network is always to be regarded as a whole which is dynamically changing. A core planning could theoretically be done without a previous access network planning. This is often the case as core and access planning is de-coupled for many operators. In this case core planing is done based on the expected subscriber figures and the forecasted traffic for the MSC area in consideration. The purpose and the basic idea of this document is to show how to sequentially proceed with traffic payload planning.
Core network has well defined capacity of service requests it can handle. These service attempts may be in the form of Voice calls, SMS, Data calls or other type of service requirement that is supported by the system. There are many network elements involved in a typical call scenario and each network element has a definite call handling capacity. Again the call handling capacity of each network element is dependent on a number of factors such as caller behaviour, software features, hardware capability etc. In addition to planning the capacity of the network element itself that is involved in handling a particular type of call, there are links (signalling and traffic bearing) that carry traffic between these network elements based on the type and nature of the call that needs to be planned to efficiently deliver a minimum assured Quality of Service to the end customer.
All of these concepts are defined and explained in detail later in this chapter. 2.2. STAGES WHEN PLANNING NEEDED Core Network Solution system is a huge investment incurred by the Service providing operator and contributes significantly to his CAPEX costs. Service providers profitability therefore depends on using this high cost capital infrastructure to its optimal capacity. In addition to the installed base of equipment capacity, operators need to keep pace with growing number of subscribers and features. This growth plan and seamless integration with the existing infrastructure is a key component in Operators infrastructure investment related decision making process.
Core Network Planning & Designing exercise can be broadly distributed over the following stages,
Bidding: This stage of planning is before the equipment is ordered and built. During this stage traffic is assumed and a basic dimensioning exercise is conducted on the basis of assumed subscriber count and their behaviour. This is the design stage and does not involve a detailed capacity planning. Bidding process could be for a number of reasons such as capacity
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 augmentation, new area to be serviced by an established operator or greenfield application. In the first two instances viz., capacity augmentation and new service area the approximation in very close to the real traffic whereas in case of greenfield application this designed network needs to be constantly monitored after deployment (observation stage) to amend the provisioned capacity. Typical flow of different stages of Core network implementation is as shown below,
Core Network Design
Detailed Engineering Site Survey DPQ Dial Plan Routing Equipment Layout System Engineering (Central/Regional) Dial Plan Provisioning (Core Network Design Guideline) LOI Site Interaction with Mot Site Plan Interaction with Supporting customer Supporting for any new Accounts Specification, Testing & Installation & Commissioning Physical Installation Commissioning Database loading Acceptance Testing Customer inputs Post Commercial Cutover Performance Optimisation Growth Planning O &M (Core Network Planning Guideline)
Detailed Engineering: During this stage, the designed system on the basis of assumed traffic parameters, is further refined to a higher level of granularity and the equipment is provisioned and installed.
Observation and Measurement stage: During this stage, the operator collects data from the system to measure actual traffic and load on the system and its key components.
Forecasting: In the forecasting stage, historical data gathered from monitoring the system is projected into the future to determine - when components in the system must be augmented to sustain the growth in traffic. This stage is also called as the Detailed planning stage wherein detailed exercise is conducted to estimate and evaluate the traffic flow patterns within the network. On the basis of this study alteration of provisioned network topology, alteration/augmentation of provisioned network element resources, alteration of engineered link and bearer capacities is taken up.
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 2.3. MOTOROLA NETWORK PLANNING SUPPORT INFRASTRUCTURE 2.3.1. DESIGN & PLANNING SUPPORT
Motorola has a well established guideline to estimate the capacity of new and existing product lines. Important points in this process are identified by M-Gates. Based on inputs from Motorola product support, prediction on the capacity and performance of new design software and/or hardware is made. At M-Gate 11, Network services group works with the regional services and local installation team to collect the data from actual deployments where the equipment is subjected to real customer traffic models. Problems/bugs discovered are fedback in the appropriate stages of Design process for proper redressal and corrective action is invoked before the product is destined for M-Gate 3 release process where it is declared fit for volume introduction into the market as shown in the flow chart below,
Design Engineered Capacity (prediction on the basis of lab tests) Design software/ hardware tools Verification at Customer premises Fix Product Bugs Fix Tool Bugs Mgate 3 Product Release Capacity published, Computing methodology, Results prediction Release of Service Tools
2.3.2. ENGINEERING SERVICES In addition to capacity engineering services, Motorola addresses any other service requirements that may arise from the customer, such as,
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 2.3.2.1. Network load balancing - Balancing the traffic such as data, voice, signalling among all the network elements deployed in the customer network so as to maximise utilisation of all the invested capital equipments such as serving MSC, gateway MSC, IN Platform, SMSC, STP, GGSN, SGSN, PDSN etc 2.3.2.2. Network capacity optimisation / augmentation Analysis of traffic and call flow patterns requiring in depth study of the observed statistics in consonance with the database. This study typically results in exposing problems pertaining to routing and dial plan deficiencies as well as suggesting new network architecture that optimises the flow of traffic so as to augment revenue and minimise utilisation ratio of the deployed network elements. 2.3.2.3. Capacity predictions and validation Predict any adverse capacity impacts due to introduction of new features and/or services that the Operator might contemplate to provide. These predictions help operator in avoiding any misadventures and lose competitive customer base. Complex algorithms are used to internally calculate the resource utilisation due to different call model and suggest preemptive actions. Upon introduction of the new service or software release, the capacity is validated 2.3.2.4. Software tool requirements Based on the deployed network elements Motorola can help the Operator in identifying any specific need for development of new tools that might be useful to the customer including requirement for services to integrate a new network element to extend a specific type of service etc.
The Motorola support structure for supporting Customer on capacity, services and feature issues is as below,
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Customer Field Support Motorola on site Sales/ Marketing Product Development, Testing, Product Services etc Regional Engineering Services Team Product Development FRs, REQs Regional Engineering ServicesTeam Services Development SFRs, REQs Services Pd Mgmt
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3. CAPACITY CONCEPTS 3.1. CAPACITY CONCEPTS While evaluating the capacity criteria for core network elements, some of the fundamental concepts and their relationship with each other are important to understand. Concepts such as, BHCA Call Hold Time Minutes of Use Erlangs Grade Of Service CPU Occupancy Memory capacity Messaging capacity Channel capacity Physical capacity 3.1.1. BHCA BHCA or Busy Hour Call Attempts by definition is a measure of number of call attempts that the network element handles during a busy hour. Typically BHCA relates to the time consistent busy hour ie., call attempts averaged over a period of time. Normally network is engineered to this busy hour call attempt rate of call arrivals.
It has to be noted that from the processor point of view call attempts are not only incoming (or originating calls) call attempts, but calls that are attempted to be terminated are also considered as call attempts. They are called as the O leg and T leg of the call and accordingly registered as 2 separate attempts for a single mobile to mobile call.
Call attempts as an indicator is important due to the fact that control processor of the switch (or any network element) gets actively involved primarily during this phase of the call set up process and hence the internal resource gets utilisated that puts a cap on the engineering limits of that node. 3.1.2. CALL HOLD TIME Call holding time (CHT) by definition is a measure of period during which the conversation phase happens between the calling and the called subscriber. Holding time is therefore a measure of amount of usage of the equipment that gets held due to this conversation. Accordingly the subscriber is charged for this duration.
By definition there is a definite amount of time that lapses during the call setup phase, when the equipment is blocked for handling that call and therefore included in the holding time. As technology improved, the processing speed of handling an instance of call has reduced dramatically causing significance of this delta time to become less important for all practical purposes.
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 3.1.3. MINUTES OF USE Minute of Use is another unit typically used in specifying the call quantity. In simple terms 1 MOU is equal to 60 call seconds. 3.1.4. ERLANGS Named after A. K. Erlang, founder of the theory of telephone traffic, an Erlang is an international dimensionless unit of traffic intensity defined as a product of call arrival rate and call holding time (A = BHCA x CHT). One Erlang is equal to 60 call minutes, which could be a single 60- minute call, 60 one-minute calls, or any combination of calls and minutes adding up to 60. Erlang therefore is a measure of the quantity of call that is handled by a network element.
Alternate definitions of Erlangs that merit consideration are as below, Erlang is an average number of simultaneous occupations for a defined time interval Average occupancy of a trunk over a defined time interval
Expressed mathematically, following applies
A =y . s
A =Traffic in Erlang y =Call intensity (calls/time unit) s =Mean holding time T =Length of the time period t v =Length of occupation no.v N =Total no.of occupations p =No. of simultaneous occupations in the group t p =Total time with exactly p occupations N =Max.no of occupations =no. of trunks A = 1
t v v =1 A = 1
n p. t p p =0
To convert BHCA to Erlangs following equation may be used,
Erlangs = (ACHT) * (BHCA) where ACHT = Average Call Holding Time 3600
For the purposes of calculating erlangs for trunks, standard erlang conversion tables are available. These tables are called Erlang B table and applies to the lost-call-cleared calling scenario. The Erlang B traffic model is the simplest and assumes that all blocked requests for
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 service leave the system forever. The Erlang B table provides the number of trunks (channels and radios) needed versus offered traffic (in Erlangs) for different blocking probabilities.
The Erlang C formula and tables are applicable to a lost-call-delayed calling scenario and determine the blocking probability, average delay time, and average queue length. It assumes that all callers will wait indefinitely to get through.
3.1.5. GRADE OF SERVICE Grade of Service by definition is the number of calls that is allowed to fail due to equipment congestion for every 100 call attempts and therefore expressed as percentage. Hence 1% grade of service would indicate 1 call allowed to fail over 100 call attempts due to equipment congestion. There are many GoS criteria that is applied across different network elements involved in a call. Network GoS is the overall - end to end - grade of service that would involve the entire call chain that is applicable for that call instance. Trunk GoS pertains to the number of calls that can fail over that particular trunk group due to lack of free available trunk circuit to handle a call. Congestion has a direct correlation with the engineered capacity of the system. Trade-off happens between the cost of procuring the equipment and Grade of Service. Very high grade of service (implying very few calls rejected) causes the equipment costs to go up due to excess capacity that needs to be engineered for a particular solution. On the other hand reducing the equipment costs by under engineering capacity causes revenue loss due to call rejection. Over time it has become an industry practice to engineer the network equipment for a 1% or 2% grade of service (as specified by the operator).
3.1.6. CPU USAGE One of the most important parameters in the context of softswitch that needs constant monitoring is the CPU usage. This is due to the architecture that is employed in MSS as well as most of the core network element that is based on server technology such as HLS, INS etc. CPU usage can be defined as the amount of time available in the processor of a system node for execution of tasks, including processing of calls and noncall tasks such as programming, audits, log processing, statistics handling and delta provisioning etc. In case of MSS since the processors run in active backup configuration (such as ccSwitch, Advlr, Adhlr) adequate care has to be taken to ensure that CPU usage does not exceed 50 % - wherein 40 % of Cpocc is attributed to the call processing and the remaining 10 % is used for p2p (peer 2 peer) sync. 3.1.7. MEMORY CAPACITY Memory capacity imposes a limiting factor on number of calls that can be handled by a single CPU card. This is so as enough on board memory should be kept available to ensure enough space exists for new call originations as well as all the existing calls this is in addition to the memory occupied by software load itself. In MSS most of the call processing occurs in CPU card which has a limited memory. Call contexts are maintained for each and every call handled by that CPU card and they are held as compact data structures. Due to the finite limitation imposed by onboard memory, there is a capacity limitation that occurs when the number of calls simultaneously handled by that CPU card increases. As the hardware keeps upgrading, the call handling capacities improves (within the limits imposed by CPU Occupancy) and the latest capacity possible may be referenced from the latest release of MSS Configurator.
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 3.1.8. MESSAGING CAPACITY Messaging capacity refers to the limitation imposed on account of the number of messages that are exchanged between components within a system. The number of messages per call depends on the call scenario and the number of features invoked etc. 3.1.9. CHANNEL CAPACITY Channel capacity refers to the number of network channels available in a switching component. As the softswitch technologies employ Server and Media Gateway combinations, all the network channels are created in the Media Gateway. Typically this is implemented as ATM backplane switching and this capacity refers to the limitations imposed by such an arrangement. 3.1.10. PHYSICAL CAPACITY Physical capacity refers to the number of terminations a component or a group of components supports, such as the total number of ports in a SS7 Message Handler card etc
3.1.11. WIRELESS TRAFFIC Shown below is a typical wireless network involving different components and telephone traffic in a wireless system varies according to the time of day, day of the week, month of the year, and holidays. This traffic variation passes from very low levels to peaks of usage. Operators must study this behavior in order to offer quality services at a reasonable price. Figure below shows traffic capacity units applicable to each system element. For example, some elements real-time capacity is characterized by busy hour call attempts (BHCA) at the recommended engineering limit, calculated from measured values of CP occupancy.
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Mobiles Radios, Voice and Control Channels Cell BSC Media Gateways MSC Erlang per subscriber Call Attempts per subs Erlangs per voice channel BHCA at engineering limit CP occupancy BHCA at engineering limit CP occupancy BHCA at engineering limit CP occupancy Erlangs BHCA at engineering limit CP occupancy
The study of telephone traffic involves an attempt to quantify these traffic variations and help operating company engineers estimate the amount and types of equipment needed for a system. The best way to gather reliable information is to observe the traffic presented by one exchange during preestablished periods of time. After a number of observations, or samples, average data from this traffic is calculated. This establishes rules and trends for the application and forecasting of an optimized telephone system.
3.2. EMERGI NG NEW CONCEPTS With the evolution of telecom networks from the traditional Circuit oriented approach to low cost, highly adaptable packet based solutions such as IP, ATM etc., new concepts and standards are being evolved to cater to this evolving reality. Voice traffic traditionally carried over dedicated circuits due to strict Quality of Service requirements are yielding to packet based carrier primarily due to following reasons Highly adaptable and Low cost packet based equipment Tremendous advances in technologies such as variable and low bit rate compression techniques surrounding packetisation Packet-based networks do not allocate transmission resources statically augmenting benefits accrued due to compression technologies Evolution of standards surrounding IP domain Very wide acceptance of IP technologies with internet providing a catalyst for it ubiquity
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 There are two aspects that emerge as benefits to Operator due to packetisation viz., Packetisation of Signalling and Packetisation of Bearer such as voice, video etc. Both are examined below and implications analysed as under. 3.2.1. CALL CONTROL SI GNALLING OVER PACKET TRANSPORT TECHNOLOGIES Traditionally signalling in Telecom networks has been carried by circuits which were used as transport layer to transport SS7 signalling. SS7 stack has been originally conceptualised and evolved around this transport technology. Most of the call control signalling happening around network elements were evolved using this technology and application layers such as ISUP, MAP, CAP, IS41, WINCAP etc were adopted to be carried by SS7 transport layer viz., the MTP layers. Application layer signalling responsible for call control such as ISUP in its current form therefore are tightly integrated and associated with the underlying bearer transport mechanism through MTP layers. With the emergence of new technologies such as IP/ATM, however, there emerged a need to disengage strict dependence between the bearer control and call control leading to emergence of two solutions which are vying with each other for filling in this gap BICC Bearer Independent Call Control and SIP T Session Initiation Protocol for Telephones. 3.2.1.1. Bearer Independent Call Control This solution is built around upgrading existing call control protocol (ISUP) to its successor, BICC, with the aim of enabling other transport technologies, such as IP or ATM. BICC re-uses most of ISUP, changing only those parts related to the underlying bearer. This scenario is particularly applicable with disaggregated networks of the type of covered under Rel 04 onwards wherein the Call control entity is different and physically seperated from Media handling entity such as a Media gateway. Consider the scenario when a mobile to mobile call traverses through two mediagateways (ie., through Nb interface) and call control happens through Nc interface. Nb interface could be implemented using ATM/IP to derive the advantages of packetisation.
Transcoder is the network function in charge of processing voice traffic and convert it between representation formats, i.e. codecs. Cellular networks have traditionally employed specific codecs that compress speech and utilize efficiently the expensive bandwidth in the radio interface. With the introduction of voice- over-packet transport, compressed voice can also be transmitted across core network, yielding significant bandwidth savings. The main drawback of codecs is the voice quality degradation caused by transcoding. Therefore, it has become critical for cellular networks to incorporate signaling procedures to control which codecs are available and when/where transcoding must be applied.
Thus, the call control protocol must offer signaling procedures to negotiate and select the voice codec, and change or renegotiate it under certain circumstances (inter-system handover, call forward, etc). The main functions that the protocol has to support are Codec Negotiation, Codec Modification, Mid-call Codec Negotiation. When the Codec negotiation procedure finds a codec common to mobiles and network nodes involved, transcoding can be avoided and the call is said to be in Transcoder Free Operation (TrFO). Even if codec negotiation does not result in a TrFO call, it is required when the network supports multiple codecs and it provides
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 other advantages such as inserting the transcoder at the most appropriate location (e.g. Remote Transcoder Operation (RTO) when it is placed at the CN edge).
Gateway MSC Srv BSC MSC Server MG MG Nc Nb Mc Mc BSC BSC A Signaling Interface User Traffic I t f PLMN/ PSTN Nb Nc As shown in the architecture above, the call control protocol (without media or bearer dependency), BICC, is exchanged in Nc interface. The Bearer control (such as bearer identification, codec negotiation etc) signalling is exchanged between Media Gateways but that negotiation does not happen over Nb interface, instead it happens over Mc -> Nc -> Mc interface through a special tunneling mechanism as shown in the conceptual figure below. It is referred to as a tunnel, because the BICC protocol does not inspect, check or understand the information it passes and the advantage it offers is that a separate signaling transport between MGs does not have to be configured; the same BICC signaling transport is used instead.
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 Bearer control is thus employed by the MGs to exchange the bearer instance particulars. For IP transport, ITU reused the IETF Session Description Protocol (SDP), though imposing some constraints on the protocol. The Bearer Control Protocol messages can be tunneled over the Mc and Nc interfaces, (encapsulated inside Megaco and BICC messages respectively) or exchanged directly between MGs via alternative signaling means. 3GPP mandated bearer control tunneling for IP bearers, preventing MGs from sending SDP information directly to each other. The bearer control for IP bearers defined by the ITU as IP Bearer Control Protocol (IPBCP), is just a simplified SDP, where most options have been pruned and a new attribute to signal the IPBCP message type (Request, Accepted, Confused or Rejected) is defined. The main limitation imposed to SDP is that only 1 RTP payload type is allowed in the format list, so that codec negotiation must take place prior to exchange of SDP. The main purpose of the IPBCP messages is to allow MGs to exchange IP addresses and port numbers for both ends of the media stream, as well as media properties of the selected codec. As regards to BICC itself, it is very similar to ISUP except that the bearer part of the signalling has been removed and the CIC field (Circuit Identity Code) has been reused for the Call Instance Code (CIC) in BICC. The function is very similar: this code identifies the signaling relationship between peer BICC entities for a certain call, and all the PDUs for the call should be tagged with the same code. BICC reused the existing ISUP Application Transport Mechanism (APM) to transport bearer-related information between call control instances. This mechanism comprises the Application Transport Parameter (APP), which BICC actually tailored as a container for bearer information, and the Application Transport Message (APM). The APM message is only generated when there is no other BICC message that the APP can piggyback onto. Among the new information elements defined by BICC inside APP, we find the Action Indicator, the Backbone Network Connection (BNC) Characteristics and the Codec list. 3.2.1.2. Session Initiation Protocol for Telephones (SIP-T) This solution is based on the IETF architectural framework Session Initiation Protocol (SIP) for Telephones. The SIP-T recommendation describes how to use SIP to provide telephony over IP networks, and interwork with legacy ISUP networks. In particular, it identifies the basic mechanisms required: encapsulation of ISUP messages into SIP payload, translation of ISUP information into the SIP header and a few new SIP extensions required to adequately emulate ISUP. An important point about SIP-T is that it is focused on enabling plain telephony calls, disregarding additional services. That is the main reason why SIP-T dictates that MSCs must build and encapsulate the whole ISUP message inside the SIP payload. The MSCs need to read the service information in the ISUP message to be able to support supplementary services and other legacy network features. Regarding signaling transport, SIP can run on top of several different protocols, like SCTP or TCP, or TLS (encrypted transport) over those. Obviously, SIP signaling transport, SCTP or TCP over IP, cannot route messages based on telephony number (E.164 numbers). For call delivery MSS has to translate a received E.164 Mobile Station Roaming Number (obtained using MAP Send Routing Info) into the destination MSC IP address.
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 Bearer control uses Session Description Protocol, exchanged as SIP payload. SDP defines a format to formally describe the bearer details for a variety of technologies. It has to be actually carried by another protocol, say Megaco on the Mc interface and SIP on the Nc interface and the only dynamic procedure considered for SDP is bearer negotiation.
On the issue of signalling therefore the implementation of BICC or SIP-T will probably be determined by the preferences of Telecom Operators. BICC is backed by GSM/UMTS operators, while 3GPP2 CDMA standard body seems to support SIP-T. However, there is a technical parameter that favors BICC over SIP-T. BICC has a comprehensive set of detailed specifications, some of them applied to cellular networks. SIP-T specifications are not complete or detailed. Thus, BICC maturity ensures the inter-operability with other vendors. 3.2.2. EVOLUTI ON OF TRANSPORT TECHNOLOGIES FOR SI GNALLING With the evolution of high capacity networks, the existing capacity constraints on signalling bandwidth were imposing severe restraints in growth potential of network elements. In order to enhance the bandwidth capability of signalling carriers, many methods have been tested and are in existence. Some of them are the traditional DS0 or Low Speed Links (LSL), 2M High Speed Links (HSL) and the new age IP based signalling transport or SIGTRAN. As is well established the bandwidth capacity of LSL is DS0 bandwidth which is 64000 bits/sec or 64 Kbps. In case of HSL, the entire 2 M bandwidth capacity (or 1.5 M in case of T1) is considered as a single signalling link so as to exploit a much higher signalling payload capacity with 2 basic types of HSL viz., HDLC HSL and ATM based HSL defined by standards. IP based transport mechanisms are defined by SIGTRAN protocol workgroup and adopted by 3GPP R4. This protocol suite is made up of a new transport layer the Stream Control Transmission Protocol or SCTP and a set of User Adaptation layers which mimic the services of the lower layers of SS7 and ISDN. SCTP addresses many of TCP shortcomings and offers key advantages such as Multi-homing, Multi- streaming facility & SCTP timers are more adapted to SS7 MTP transmission delay timers. In order to carry signalling over SCTP/IP user adaptation layers for different transport layers had to be defined viz., M3UA, M2UA, M2PA, SUA, IUA etc.
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IP SCTP M3UA M2UA M2PA M3UA SUA IUA ISUP SCCP TCAP MTP3 ISDN
3.2.2.1. M2UA or MTP2 User Adaptation The only SS7 MTP 2 user is MTP3. All MTP 3 primitives are backhauled to the MGC for interpretation. One of the key advantages of M2UA is that the SGW does not require a separate PC. Signalling Gateway signifies that the entity performs the gateway function to transform TDM based signalling to IP based signalling. This function is typically implemented in MGW & Remote MGW to take advantage of extracting signalling information communicated from RAN through TDM Messaging timeslot and convert into IP packets without the need for RAN to address Media Gateway by the way of point code. Using M2UA obviates this need and RAN can address CS directly for all signalling information and media gateway routes the same through its M2UA link to Control switch.
TDM Signallin IP Signallin SGW Functio CS RAN MGW Typical Usage of M2UA
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 3.2.2.2. M2PA or MTP2 Peer to Peer Adaptation M2PA is the peer to peer equivalent of M2UA. Rather than provide a link between remotely located MTP2 and MTP3 instances, it replaces an MTP2 link beneath MTP3. The user of M2PA is MTP3 at both ends of the connection (with M2UA, one user is MTP3 and the other is SGWs Inter-working function). Typical M2PA architecture is shown as below,
Signalling Gateway
MTP3 MTP2 MTP1 IP SCTP M2PA Signalling Gateway
MTP3 MTP2 MTP1 M2PA IP SCTP SS7 SS7 M2PA Architecture
This architecture is most applicable for an SGW to SGW connection used to bridge two SS7 network islands. MTP3 is present on each SGW to provide routing and managementof the MTP2/M2PA links. Because of the presence of MTP3, each SGW would require its own pointcode.
The significant difference in function from M2UA is that M2PA actually provides an MTP2 like service itself. M2UA merely provides an interface to a remote MTP2 service. This means that M2PA is responsible for Link activation/deactivation (in response to requests from MTP3), Maintaining link status information, Maintaining sequence numbers and retransmit buffers for retrieval by MTP3, Maintaining local and remote processor outage status etc.
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 3.2.2.3. M3UA or MTP3 User Adaptation M3UA provides adaptation between SCTP and applications that use the services of MTP3 across an IP interface. SS7 User applications such as ISUP, MAP, TUP etc. Typical usage of this adaptation layer is for PSTN connectivity in instances when POI can support IP signalling or Apertio HLR, MSC to SGW/STP etc. The architecture of M3UA is shown as below,
Signalling Gateway
MTP2 MTP1 IP SCTP M3U MGC
ISUP M3U IP SCTP SS7 MTP3 NIF M3UA Architecture
The MTP3 in the SGW is unaware that the ISUP user is located remotely. Similarly ISUP layer at the MTC will not be aware of the transport layer used to carry the signalling interactions. This architecture is most appropriate in instances wherein there is a high enough density of SS7 links that normal SS7 link capacity is insufficient to cater to the entire signalling requirement, SS7 links are physically accessible at a single point
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 3.2.2.4. SUA or SCCP User Adaptation SUA provides a means by which an application part (such as TCAP) is sent over to an IP SCP through a SGW. The network architectures associated with SUA allows for multiple IP SCPs to be reached via a single SGW. The IP SCP do not have local MTP3 instances and so do not require their own SS7 point codes. The functionality of SUA could be provided by MTP2 or MTP2 UA. However SUA provides the mapping between SCCP addresses and IP addresses (at the SGW). Without such a function, SCCP would have to be present at each IP SCP and the external SS7 network would require knowledge of each such SCCP instance. SUA can abstract the presence of each IP SCP, providing one SCCP address to cover all nodes. The services of individual databases are addressed via Subsystem Number (SSN). SUA provides a service similar to SCCP GTT to map
Signalling Gateway
MTP2 MTP1 IP SCTP SUA IP SCP
TCAP SUA IP SCTP SS7 MTP3 NIF SCC SUA Architecture
these SSNs into SCCP Connection IDs (which are used to route the Application part messages to the appropriate IP SCP). SUA is also flexible enough to support
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 Application parts running between two network nodes wholly within the IP network. This is particularly relevant to emerging networks where there may be no need for an underlying traditional SS7 network. In this case the IP SCP stack would be the same on both (IP based) nodes. SUA will further allow Service Databases in the SS7 network to be accessed from the IP network. In this case the architecture would be the same as shown in the figure above. 3.2.2.5. IUA and V5UA Similar to the previous user adaptations, ISDN user adaptations work to adapt the ISDN layers to the IP transport 3.2.3. BEARER OVER PACKET TRANSPORT TECHNOLOGIES As in case of signalling carried over packet technologies, bearer support over packet technology is emerging as the new trend in Core Network domain due to incorporation of QoS criteria meeting the stringent needs of Voice and Real time services requirement being built into protocols. Various options such as G.711 PCM over IP, G726 over IP, EVRC/FR/EFR Bypass, Accelerated A1p/A2p, TFO/TrFO are available with varying degree of advantages and development effort to incorporate them in the product portfolio. G.711 PCM over IP just packetises the PCM samples and sends it over the IP does not involve voice payload compression and result in higher bandwidth requirement due to header used in instances when there is no dearth of IP bandwidth. G726 over IP same as before except the payload is compressed voice using dominant format used in enterprise networks today - R-MGW would use standard TDM-based RAN interfaces to set up calls and route G.711 bearer, after which MGW compresses the voice internally from G.711 to G.726-32, and places on IP backbone network. EVRC/ FR/ EFR bypass are applicable in CDMA(EVRC) and existing GSM (FR/EFR) networks wherein the encoded speech is sent as it is to MSC and MSC-RAN signaling messages communicate to setup this vocoder bypass (Motorola CDMA solutions call it Supercell TRAU). As transcoding is minimised, this results in best quality of speech. The RANs at each end (mobile-to- mobile call) look for inband, robbed-bit signaling to communicate call setup capabilities and mid- call changes used in instances when IP network bandwidth gain combined with low bit rate provides significant economical benefits. Accelerated A1p/ A2p is the same as previous option except the RAN has pushed up the use of packet interfaces into the 2005 timeframe. Thus the LBR voice and PCM for a given call are sent in independent RTP streams to/from the MGW. A mechanism is provided to toggle between using the PCM and the LBR streams. TRAU is eliminated from use in the RAN-TRAU interface. The inband, robbed-bit signaling to communicate call setup capabilities and mid-call changes is replaced with proprietary A and A1 messages and the MSS control takes over more control of call setup and mid-call modifications. This is desirable when the IP network used for VoIP is so incredibly expensive that LBR produces significant economical benefits. Compared to others, the voice quality is best because the number of transcoders in minimized. This option is NOT currently available on the MSS; the ability would have to be added for AudioCodes, Sonus and Motorola MGWs. TFO/ TrFO solution would use full negotiation methods to select transcoding type and location for inter-system calls. This is desirable when the IP network used for VoIP is so incredibly expensive that LBR produces significant economical benefits. The R-MGW would use IP RAN interfaces to set up calls and route LBR for M-M, PCM (transcoded in the CORE) bearer for M-L and L-M on RTP/UDP/IP streams. The MGW is instructed to use the appropriate voice format after end-to-end negotiations determine where and how vocoding occurs.
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 Header compression is not prevalent today in Media Gateways suppliers, but will more than likely be deployed in a later phase. Since voice payload compression is farther along, they are more readily available. It is interesting to compare the fixed bandwidth (64,000 bps) per voice call today in the TDM network to the bandwidth required to carry VoIP in an IP network. The key thing to consider is that the IP bandwidth per call depends on the packetization rate of the IP endpoints, with a typical current practice today of setting this to 5-20 milliseconds.
Figure below depicts the relationship between bandwidth, packetization rate (PR) and Voice Activity Detection (VAD). The bandwidth for a TDM system is the benchmark and is shown as straight black line at 64 Kbps (TDM systems do not vary due to VAD or PR. Generally, for a VoIP system, the higher the PR, the lower the VoIP bandwidth. The more time an IP endpoint takes to gather multiple PCM-TDM voice samples before putting into an IP message, the lower the percentage overhead impact due to RTP/UDP/IP addressing, and the lower the over all bandwidth. At the far end of typical practice (5-20ms) the effect reaches asymptotic limits sine there is always a fixed amount of IP addressing that requires its own contribution to bandwidth.
VoIP for 64Kbps PCM Input DSP 20 40 60 80 100 120 140 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 Packetization Rate (ms) V o I P
B a n d w i d t h
P e r
C a l l
( K b p s ) TDM (baseline) w/o VAD/CNI 20% VAD/CNI 40% VAD/CNI 60% VAD/CNI 80% VAD/CNI Bandwidth per Call when Using G.711 voice formatting over IP
A few observations from above fig G.711 over IP: Without VAD, putting 64Kbps payload with IP addressing overhead can never beat traditional TDM, Even if experiencing 20% VAD (means speech averages idleness 20% of time and the ability to NOT send packets during those moment of quiet reduces the bandwidth proportionally), the VoIP solution cannot beat TDMs 64,000 fixed bandwidth requirement.
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 There would need to be VAD at 40-80% (statistically unlikely) to offer a hope that PCM over IP beats TDM in lower bandwidth, and even then, it may require a favorable PR of 8 ms or higher. Figure below shows a similar comparison, only replacing PCM voice format in the payload with a compressed voice. Although the figure shows G.729 in the title, it generally represents a compressed voice payload of 8Kbps, such as EVRC or GSM-FR, etc. That is, the results of this comparison can be applied to various LBR voice formats.
VoIP for 8Kbps Compressed Voice Input DSP 20 30 40 50 60 70 80 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 Packetization Rate (ms) V o I P
B a n d w i d t h
P e r
C a l l
( K b p s ) TDM (baseline) w/o VAD/CNI 20% VAD/CNI 40% VAD/CNI 60% VAD/CNI 80% VAD/CNI Bandwidth per Call when Using G.729 voice formatting over IP
A few observations from Figure above showing compressed voice over IP: VAD offers little extra savings due to the relative efficiency of the compressed voice (G.729) versus PCM (G.711); being able to clip voice idleness is not as important if payload is compressed than if at full rate, like PCM. The savings using compressed VoIP is almost immediate within the PR range. The degree of savings is much more pronounced than PCM over IP, even when VAD is none or little (20%).
Another way to think of the bandwidth savings of LBR VoIP to PCM TDM is to measure the number of calls per unit of bandwidth. This is particularly useful when the physical facilitlies would be common, such as channelized (circuit) vs non-channelized (packet) E1/T1. A useful common unit of bandwidth is the benchmark TDM DS0 slot of 64Kbps.
Figure below shows the extra gain in calls per DS0. Notice the gains in number of calls per DS0 can range from 1.0 to 3.0 depending mostly on PR.
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Total Calls per Fixed Transmission Varies by Packetization and VAD Rates Using Compressed 8Kbps as input to VoIP 0.500 1.000 1.500 2.000 2.500 3.000 3.500 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 Packetization Rate (ms) C a l l s
p e r
E v e r y
6 4
K b p s
o f
B a n d w i d t h TDM (baseline) w/o VAD/CNI 20% VAD/CNI 40% VAD/CNI 60% VAD/CNI 80% VAD/CNI Number of Calls per DS0 for LBR VoIP by Packetization Rate and by VAD
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3.3. ELEMENTS OF NETWORK TOPOLOGY As below, various network elements are involved in a typical MSS-G and MSS-C network topology. Network elements that are typically associated with a wireless architecture are BTS, BSC together they form the part of Radio Access Network or RAN. The Core network comprises of Media Gateway where the bearer trunks from the BSC is terminated and signalling trunk which is terminated in MSS chasis itself.
The Motorola End-to-End Architecture CDMA-One/CDMA- IP Backbone TDM Backbone IP SS7 IOS 4.x RA Core SIP, MAP+/IP, IOS/A on IP ANSI-41 HLR, AuC, SCP ANSI- 41,IS771, GSM MAP, CAMEL TDM / ATM IP ISUP MSC/ PSTN Switch TDM TDM/ATM/IP Control Switch Media Gateway
ALL-IP IP RAN IP RAN Motorola GSM/UMTS BSC MSS-C MSS-G MSS-U Future Control Switch Media Gateway
Typically PSTN, Other operators are connected to the operator network through Points of Interconnect or POI in Gateway MSC which is configured to perform the IN/HLR/AuC interactions for Mobile termination scenarios. In case of large networks multiple serving MSCs are interconnected through bearer trunks if inter MSC roaming scenario is applicable, otherwise they are directly connected to the GMSC (signalling and bearer trunks) to cater to Land to Mobile call scenario and vice versa. Inter MSC trunks may be provided in the above case if there is substantial mobile to mobile traffic between the regions served by these MSCs, else the calls may be allowed to transit through GMSC. In smaller markets the serving MSC itself functions as the gateway MSC where POIs are terminated in which case, VLR has to be provisioned.
In case of a typical GSM network HLR is an external network entity and signalling link is extended from MSS directly to HLR or through an STP in case of a large network. HLR is an SCP type of
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 SS7 network element that typically supports authentication. In instances when Mobile number portability, has to be supported, many markets incorporate the function of Signalling Relay Function in HLR itself. Value added services such as Short Message Service, Multimedia messaging service, Voice Mail service etc are typically provided by standalone servers that are connected to the MSS. In instances when SMS, MMS servers are provisioned as a standalone network elements, only signalling links, based on SMS/MMS BHCA, need to be extended from the MSS. On the other hand in case of Voice Mails systems, both signalling(based on busy hour Voice mail retrieval rate) and bearer trunks (based on erlang) have to be provisioned.
Prepaid solution and other intelligent network solutions are provisioned through an external IN platform which is WINCAP capable in case of CDMA and CAMEL compliant in case of GSM. All the IN solutions need only signalling interaction with the serving MSS and signalling links are calculated on the basis of number of services the operator wishes to host. 3.4. TRAFFIC FLOW CHARACTERISTICS To characterise the calls handled by MSC, typically H diagram is used to graphically represent the combination of incoming and outgoing traffic where Mobile traffic is shown on the left and Land traffic on the right. The Hdiagram shows the possible terminations of a call, whether a call originates from a mobile or from a land trunk. Intra-switch traffic occurs when the call follows the mobile-to-mobile flow. The flow of traffic originated by mobiles runs mainly from the outgoing trunks to the land trunks, and has a small amount of inefficiency when a block in the exchange or
M M-L M M INTRA (mobile to mobile) TANDEM (Call forward, Voice mail etc) Mobile Origination Mobile Termination Incoming traffic (From PSTN, Gateway etc) Outgoing traffic (To PSTN, Gateway, Intra MSC etc) Failed mobile originations (Setup failures, Invalid attempts etc) Failed mobile terminations (Page timeout, Invalid mobiles etc) M o b i l e
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 on the outgoing route occurs. Out of this a portion of the flow terminating in mobiles represents the traffic between mobile units (ie., mobile to mobile traffic). Similarly, the incoming traffic generated from the land trunks terminate largely in the mobiles; a portion of the traffic is tandem and a portion is ineffective. Another additional component of the tandem calls is represented by the traffic in the tandem trunks in MSS, for example, calls transferred back to the network itself. Other tandem trunks are added to permit an adaptation of the signaling supported by the exchange and the signaling characteristics of the operators network.
3.5. BASIS OF BUILDING CALL MODEL
Core Network Design involves arriving at the traffic flow pattern within various network elements that it is comprised of. Network designing activity involves the determination of the traffic volumes between the various core sites of a network and involves all elements where traffic can be originated or terminated. From core networks point of view, typically, those elements are the MSC itself, the voice mail centre VMSC, Points of Interconnection towards PSTN or PLMN. Between those elements a logical fully meshed traffic relation exists which can be expressed in end-to-end matrices. Those matrices exist for each call type, i.e. mobile originating traffic, mobile terminating traffic, mobile to mobile traffic and forwarded traffic etc. Typical flow chart involving such a design process is shown as below,
Core Network Design Flowchart
Inputs Call Model #of Subs Determine traffic distribution pattern for all MSCs Network wide traffic planning for all types of calls Traffic routing and Network topology Signalling network planning Link dimensioning
Those matrices need to be calculated separately and superimposed later to do dimensioning. Transit MSCs or Gateway MSCs are not considered here as they do traffic routing only. Routing and interfaces dimensioning are not considered at this stage and are treated in subsequent steps. Normally traffic planning is related to payload traffic as this represents the major traffic in a
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 network. Signalling planning will be done in later steps, which is based on the output of traffic planning.
3.5.1. BUILDING BASIC TRAFFIC MODEL The basic input to traffic planning are the subscriber figures and the traffic call model. It may so happen that there is no homogeneous traffic model across different MSC in the network. This does not only apply to the traffic per subscriber but also to the proportional traffic share for the various types of call such as mobile to mobile, land to mobile, mobile to land etc., thus traffic planning needs to be done for each individual MSC.
3.5.2. TRAFFIC DISTRIBUTION PATTERN FOR EACH MSC At this point it is necessary to obtain information about the traffic distribution for each MSC, such as what percentage of traffic is routed locally, type of traffic routed at different nodes, how many POIs are provisioned for National and International calls etc. Traffic dissemination in each node is computed at this stage and a separate traffic distribution for the MSC under consideration might appear as shown in the figure below. If such information pertaining to traffic discriminatioin is not available (e.g. in rough offer cases), it will be necessary to make assumptions about the traffic flows in the network and document them. It is important to know that any parameters changed after assumption have a direct impact on the designing exercise output, hence it is important to record the assumption to establish the reference frame. Based on extrapolation of different call scenarios (m2m, m2p, p2m, p2p etc) applicable for the MSC under consideration, the following detail may be arrived at,
To From MSC 1 MSC 2 POI 1 POI 2 VMS 1 BSC 1 BSC 2 MSC 1 75 12 31 42 10 30 40 MSC 2 POI 1 POI 2 VMS 1 BSC 1 BSC 2
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POI 1 POI 2 MSS 2 VMS 1 MSS 1 BSC 1 BSC 2
3.5.3. NETWORK WIDE TRAFFIC MATRIX Based on above traffic planning results for the individual MSCs it will be possible to extrapolate all the traffic matrices that describes the end-to-end dependencies between all involved nodes. Thus we obtain the entire traffic distribution picture between all the nodes involved in the core designing process. All network elements that act as traffic sink or source are part of the matrix. Typically those elements are MSCs, PSTN, PLMN, VMSC etc. Other elements as pure transit MSCs TMSC may not to be considered here as they fulfil routing functions only. For a small network consisting of 2 MSCs and 2 POI elements, the logical traffic flow may be calculated as follows. Note that some arrows are bi-directional, implying traffic summed up both ways etc.
To From MSC 1 MSC 2 POI 1 POI 2 VMS 1 BSC 1 BSC 2 MSC 1 75 12 31 42 10 30 40 MSC 2 6 10 25 41 8 POI 1 32 42 5 POI 2 76 34 7 VMS 1 BSC 1 32 BSC 2 45 Please note that the above shown values are just dummy values and are not backed up with detailed calculation
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 The above matrix is the result of this planning step and is the base for the subsequent traffic routing exercise. The individual traffic shares for MOC, MTC etc can be recognised in this table: traffic between MSCs must be MMC traffic, traffic from PSTN to MSC must be MTC etc.
End to End Traffic Flow
POI 1 POI 2 MSS 2 VMS 1 MSS 1 BSC 1 BSC 2
3.5.4. TRAFFIC ROUTING AND NETWORK TOPOLOGY Traffic Matrix once arrived, the volume of traffic between different nodes becomes evident. Next is the routing process to decide on the route to be taken for these traffic relationships. During this step decision on network element structure as well as transmission bandwidth needed between them is evaluated. The logical fully meshed end-to-end traffic matrix is to be mapped on the physical network structure and decision on which specific paths are available for each specific end-to-end traffic relationship are considered. Traffic might either be routed directly or using other nodes as relay points. Aspects like load sharing and redundancy are to be regarded here as they directly influence the traffic on individual trunks. Different traffic portions belonging to different end-to-end connections will share the same trunks and will compete there for resources. Therefore they need to be treated together for dimensioning. This is also very important as different traffic portions sharing the same lines will lead to a higher traffic multiplexing gain on the trunks. These aggregated values can then be used to determine the
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 needed bandwidths between the different network elements. The traffic in the end-to-end matrix needs to be considered on each link along which the specific traffic is routed. This leads to a revised table which then represents the aggregated traffic on the links and reflects the network structure. If further elements such as Gateway MSCs or Transit/tandem MSCs are to be used in the network, they have to be added to the matrix and have to appear as well. Retaining the example from the traffic planning, the possible routing results could look as follows.
Network Topology
POI 1 POI 2 GMSS 2 GMSS 1 MSS 1 MSS 2 VMS 1 BSS 1 BSS 2
The assumed topology above is adopted to highlight concepts of load sharing and redundancy. MSS 1 can for instance send the outgoing traffic via the Gateway MSS 1 or the Gateway MSS 2. In this case it is necessary to define a load split between all possible paths. In this example, a 50% load split for both Gateways shall be assumed. This does not apply for the MMC traffic which shall be routed via the direct route between MSS 1 and MSS 2. This leads to the following table. Note that traffic between two nodes are transferred via the same link no matter the fact which node sends or receives the traffic. Thus the traffic between two nodes can be summed up as follows,
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To From MSC 1 MSC 2 POI 1 POI 2 VMS 1 BSC 1 BSC 2 GMSS 1 GMSS 2 MSC 1 18 62 85 95.5 95.5 MSC 2 75 75 POI 1 67.5 67.5 POI 2 100 100 VMS 1 15 15 BSC 1 BSC 2 GMSS 1 GMSS 2
MSC1 sends the entire traffic towards the Gateways, except the m2m traffic. Thus the traffic per Gateway must be equal to 50% of the rest, which is 32 Erlang to POI1, 42 Erlang to POI2, 10 Erlang to VMS, 32 Erlang from POI1, 76 Erlang from POI2. Thus the traffic is 50%*(32+42+10+32+76) = 95.5 Erlang. In this matrix it is no longer possible to identify the individual traffic shares. This matrix shows now the traffic per connection and is the input for the link dimensioning. If there are several services in the network, this matrix needs to be individually determined for each service. 3.5.5. SIGNALLING NETWORK PLANNING After the arriving at the network wide traffic matrix, it is now necessary to consider signalling traffic. This is necessary as this traffic shares resources with the payload and hence needed to be accorded high priority. Also signalling planning involves planning the SS7 network elements such as STP, SCP, HLR, IN, SMSC etc. Due consideration has to be accorded to all the different types of signalling traffic that flows in a signalling link and typically it is a very detailed exercise involving processing of a numerous inputs. However there are rules of thumb available that is fairly accurate which may be used as not all the inputs are not available at the design stage. These traffic types are also described by their own traffic matrices following the same pattern as above. Signalling bandwidth demand needs to be added to the previously determined payload bandwidth if the same media is to be used for carrying signalling as well as payload traffic. Normally the signalling demand is derived from indicators such as traffic per link or number of trunks per signalling link etc. 3.5.6. LINK DIMENSIONING Link dimensioning pertains to provisioning enough bandwidth to carry the traffic per route as determined in the preceding step. The bandwidth dimensioning can be done using Erlang's theories. Standard Erlang B tables are used for this purpose when it involves lost-call-scenario which is normal in PSTN/PLMN networks. A further input for the dimensioning is the maximum allowable blocking probability P, which normally should be a very low value (0,1% or 0,01%) as a high quality network operation is desired as shown below,
76 74 73 72 70 69 67 65 63 60 Dimensioning needs to be done path-wise. As an output one obtains either the number of traffic channel with a bandwidth of 64kbps (for E1 TDM systems) or 56 kbps (as in T1 TDM systems). After all traffic components have been calculated and their bandwidths determined, it is finally possible to determine the number of E1/T1 links between the sites. Therefore one has to compare the bandwidth demand between the sites with the required capacity of a specific node connection. 4. NETWORK TOPOLOGIES 4.1. TYPICAL TOPOLOGIES Network topology broadly combines two distinct topologies viz., the bearer topology and the signalling topology. Network Topology mainly refers to the way bearer traffic flows in the network as it determines the scale of economies underlying the overall network architecture. Nevertheless signalling topology is also important in view of fault alleviation, disaster management, redundancy requirements that is demanded by newly emerging networks.
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 4.1.1. BEARER TOPOLOGIES In large networks or in cities where more than one MSC will be needed to cater to the traffic requirement of the market, the issue of networking the MSCs, pros and cons of the same need to be considered. Typical topologies that are used in the Telecom network domains are Star topology, Mesh topology, Pyramid, Double Pyramid topology and Hybrid structures. Represented figuratively these topologies would appear as below,
MSC MSC MSC MSC Tandem Star Topology MSC MSC MSC MSC Mesh Topology
MSC MSC MSC MSC GMSC Pyramid Topology MSC MSC MSC MS C GMSC Tandem Double Pyramid Topology
Star topology is rarely used in telecom domains due to its inability to be fault resilient which is critical decision altering factor to determine the topology. Any failure in any link will isolate the node and therefore is not a good solution. Mesh topology on the other hand evolves through a ring architecture where all the nodes are interconnected in a ring format without any cross links. The issue with the ring architecture is that the traffic in any node increases due to transit traffic. One of the basic thumb rule that need to be observed in Core Network Design is to congregate as much traffic as possible from all possible call originating sources, but this traffic need to be quickly disseminated out of the core network as soon as the routing design permits. This is so as to reduce the wastage of valuable call processing capability of various nodes in the network as well as to preserve the bandwidth of the intereconnecting transmission network, resulting in optimal utilisation of the network element resources. In order for this to be realised, theoretically, if we can realise a very large capacity of switch, then that will provide the most optimal solution. Mesh topology is fault resilient and any failure of transmission link does not isolate a particular node as alternative routes are available to reach the destination. It may not be possible to realise a Mesh architecture in all market deployments, but most of the deployments almost always tend to migrate towards mesh architecture as the network grows. One of the practical evolutions from mesh architecture is the Pyramid and Double Pyramid architectures. In Pyramid architecture, the Gateway MSC is at the top of the pyramid and is
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 responsible for interconnecting this PLMN with other PLMN as well as PSTN networks. There are a lot of advantages of this configuration that accrues due to optimal routing and least wastage of trunk / switching resources. Double Pyramid architecture improvises by addition of Tandem MSC. Tandem MSC does not have gateway functionality and hence the processing capacity of Tandem MSC is very high and can cater to higher capacity networks. All the outgoing traffic from the PLMN is routed through Tandem whereas all Incoming traffic from external networks is routed through Gateway MSC. The disadvantage of this design is that it becomes prohibitive in cost and is justifiable only in high capacity networks. Many an instance these perfect architectures are not realised due to limitations of trunking resource availability/connectivity and a hybrid of these topologies is implemented with a stub connection to one or more nodes etc.
Providing network solutions catering to a number of cities within a region is a common remit of operators. Networking issues and transmission design to cater to this multi city environment requires adequate planning and preparation. City A Link l, Traffic = 2yeab/ Traffic = 2yeac/ o p q m n City B City C City D
Shown above is a typical instance of four cities interlinked with each other. Based on degree of affinity, mobiles from any of the city above viz., A, B, C, D can call among themselves constituting Inter City Mobile to Mobile traffic scenario that flows in link l, m, n, o, p, q. Traffic between any two nodes is calculated and as an example in link l ie., between City A and City B it is 2yeab/ where y is intercity m2m traffic, e is Erl/subs, a is City A mobile population and b is City B mobile population, is sum of all subscribers. This traffic is shown figuratively as directly flowing between A and B, but it may not be so due to Transmission facility availability and may have to be routed via available trunks through other cities. Typically this rerouted traffic is not recommended to flow through switching nodes in the rerouted path as it serves no purpose to unnecessarily load the intermediate switch. Depending on alternate route availability, there may be more than one path to reach the destination (in case of no direct trunking facility available) and during those instances a route matrix is constructed with the cost of alternate routes. Based on the cost structure, a weighted average determines the percentage distribution of traffic through these alternate paths.
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 4.1.2. SIGNALLING NETWORK One of the important components of any Telecom Network is the Signalling component. Almost all of modern day telecom networks are built around common channel signalling wherein the signalling path is kept distinctly different from the bearer path to capitalise on bandwidth advantages obtained by delinking signalling from the bearer path. This requires therefore to construct an overlay network capable of handling signalling and the fundamental network element in this layer is Signalling Transfer Point this plane of Network is also termed as the Control Plane (vis a vis the User Plane which signifies the Bearer path). In addition to routing signalling between MSCs and PLMNs, STP is used to provide connectivity to databases such as HLR, EIR, AUC, IN etc. From a network perspective following are the reasons for implementing STP, Helps in load balancing signalling traffic load among distributed databases Ability to provide resilient and failsafe signalling route to destination Allows minor application NE to connect to all available databases in the network as well as itself being made available to all NEs within that domain (eg., Missed Call Alert system), thereby easing introduction of new services STPs act as the signalling gateways to the interface to the other domains be it TDM to IP or PLMN to PSTN or PLMN to Other PLMNs etc. STP in its functional capacity as border gateway network element implements Global Title Translation functionality obviating the need to have extensive routing database to be managed in internal NEs. Minimises the need to have Public Point Codes within a PLMN network STP can also add value by accounting for SCCP transactions which are used by some operators to settle inter network accounts eg., SMS accounting for home subscribers roaming in other PLMNs STP functionality has enhanced due to advances in transport layer technologies as detailed earlier and can work as Signalling Gateway. Signalling Gateway works as the Border Network Element between TDM based SS7 signalling network and IP based signalling network. An STP/SGW is always duplicated in a network due to its critical functionality, and failure of any one STP does not affect the healthy functioning of the network. Topologically therefore for the signalling network, Star Topology is used due to the availability of redundant nodes. Typical STP configuration is shown as below,
MSC MSC MSC HLR HLR in IN SMSC LBS MCA OTA EIR BSC BSC BSC
IP/MPLS Star Topology of typical Signalling Network
4.2. NETWORK EVOLUTION With the onset of Next Generation Networks or NGN, standards body have become active in defining the network architecture as well as links interconnecting different NEs within this new architecture. Notably 2 standards organisation viz., 3GPP and 3GPP2 have been defining this for the UMTS market and CDMA market respectively. As shown in the figure below, network architecture evolved with the release of new specifications from R96 to R99 with the inclusion of Location based service offerings (R98) and IN services, UTRAN (in R99) etc.
IuPS i/f Network Entities and Interfaces evolution in 3GPP d d There has been more dramatic changes in the architecture subsequent to R99 with the onset of NGN that started with Rel 04. Rel 04 captured the concept of MSC legacy platform bifurcated in functionality - physically as well as logically into the Call Control part which is the MSC server (or MGCF) and the Bearer Handling part which is the Media Gateways (or MGW). Server based software oriented telecom systems came into existence and traditional legacy HLRs, IN platforms have evolved into Server based HLR and Server based IN platforms. Onward from Rel 04, Rel 05 brought in another radically new concept to the Telecom domain viz., IMS or the IP Multimedia Core Network subsystem. IMS is expected to herald the onset of feature rich service offerings by Telecom operators as warranted by evolution from 2G to 3G networks. Elementary IMS functionality was further refined and clear roles defined for different elements constituting this domain as the release progressed from Rel 05 to Rel 06.
Shown below, an extract from the Network architecture specification 23.002 480 of Rel04 standards that highlights the Core Network elements and their interfaces. As can be seen new interfaces Mc, Nc, Nb pertaining to MSC - MGW, Inter MSC interface and Inter MGW bearer interfaces respectively have been defined. MSC MGW interface or Mc interface typically uses MGCP or MEGACO protocol for MSC server to control the functioning of MGW. MSC MSC interface earlier used standard ISUP to carry MAP traffic, had to be revised with Nc interface to carry BICC signalling. Similarly MGW MGW interface, Nb, though shown as a direct interaction between the two entities, in reality is implemented through Mc Nc interface as discussed earlier.
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BSS BSC RNS RNC CN Node B Node B IuPS Iur Iub USIM ME MS Cu Uu MSC server SGSN Gs GGSN GMSC server Gn HLR Gr Gc C D Nc H EIR F Gf Gi PSTN IuCS VLR B Gp VLR G BTS BTS Um RNC Abis SIM SIM-ME i/f or MSC server B PSTN cell CS-MGW CS-MGW CS- MGW AuC Nb Mc Mc Nb PSTN PSTN Nc Mc A Gb E
NGN Architecture Rel 04 heralding split between Call control and Media handling functions
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 4.3. MSS ARCHITECTURES 4.3.1. MSS CARD AND CHASIS MODULARITY MSS has a flexible architecture that allows innovative configurations which provide cost-effective solutions to the operators. Personnel involved in core network architecture will understand that one of the primary reasons for escalating costs of core network is the inability to scale the size of MSC based on demand. Fundamental design issues crop up when all the traffic cannot be handled by one MSC and needs multiple MSCs. The resources consumed for handling inter-MSC calls becomes prohibitively high when the number of MSCs increases. MSS has one of the best scalable architecture available in the market today. MSS capacity can be augmented by simple addition of CPU cards running the desired process till all the slots available in a chasis are exhausted. If the MSS capacity has to be augmented beyond a chasis, additional control chasis shelf which can accommodate additional cards can be engineered - modularly expanding the capacity of the system. Currently MSS supports upto 2 control chasis stacked together. 4.3.2. MSS WITH MULTIPLE MEDIAGATEWAYS MSS handles the bearer traffic through media gateways and any media gateway that supports MGCP can interwork with MSS seamlessly. Motorolas prefered mediagateways are Motorola Mediagateway, Audiocode Mediagateway, Sonus Mediagateway. These mediagateways may be
located locally as shown above or remotely as below,
When multiple media gateways are involved, provision has to be made to address traffic flow between the media gateways. There are 2 methods viz., Local trunks configured between these two MGW, or using GigE RTP voice streams to transport bearer from one MGW to MSS followed by MSS to the other MGW. Providing local trunks reduces the bandwidth availability on the MGW and calls for additional resources such as E1/T1 interface cards, transmission availability etc.
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 Also if the number of MGW exceeds 2, then a matrix of local trunks need to be provisioned for, all of which results in sub-optimal utilisation of available switching/MGW/transmission resources as highlighted below. Mesh Configuration of Local Trunks
MSS 1 MGW 3 MGW 1 MGW 2 Mesh Configuration of Local Trunks MGCP Control & Management only TDM for Bearer
On the other hand GigE solution is more acceptable when the number of MGW exceeds 2, as these MGWs can interwork in a star configuration as against mesh configuration in the previous case. Also in this case any future additions of MGW can happen seamlessly without affecting the existing configuration. Decision regarding provisioning one of the two methods would depend on if these MGWs are located remotely or locally as also the proximity of these MGWs and availability of IP bandwidth etc.
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Star Configuration of Local trunks realised through GigE RTP bearer
MSS 1 MGW 3 MGW 1 MGW 2 Star Configuration of Local Trunks realised through GigE MGCP Control & Management only GigE for Bearer
4.3.3. MSS WITH REMOTE MEDIAGATEWAY In case of MSS connected to MGW remotely, there are significant bandwidth gain vis a vis the Legacy solution. In Legacy solution, there are remote concentration units or remote line units which performs similar functions, but in all those solutions, the bearer traffic has to be brought to the centrally located MSC, as call control, switching crosspoint and the associated CDR generation, is performed in this location. Softswitch solution specifically adds value in this scenario with the use of Remote MGW. In this instance, as against the legacy solution, the bearer path need not be extended all the way to the MSS since the control and the bearer have been clearly demarcated and seperated right from the conceptualisation stage. Hence only the control part or the MGCP signalling bandwidth need to be provisioned for while designing for Remote MGW. Only in instances when media mixing has to be performed in control chasis, the bearer needs to be brought to the control chasis and the engineering detail of the IP bandwidth requirement etc needed for this purpose is documented in MSS System Resource Guide maintained at http://compass.mot.com/go/136985679 .
While provisioning for Remote MGW, as a rule of thumb, the total size of MGCP or MEGACO signaling messages in each direction for a typical call scenario is 16KB. Using this information, the bandwidth requirements for the signaling path between the Control Switch and the MGW are determined by multiplying the bandwidth requirements per call (16KB) times the call rate. Because MGWs are typically sized in terms of Erlangs of traffic, the call rate can be specified as:
Call Rate = Erlangs / Average Call Hold Time
Using this value, the bandwidth needed is then: Signaling BW (kbps) = 16KB per call * (Erlangs / Average CHT)
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 5. ENGINEERING NETWORK ELEMENTS In real life situations such detail of traffic discrimination as highlighted previously is hardly available and basic call models are provided as inputs. Based on the call model, transmission or link dimensioning exercise is taken up.
Traffic Call Model Interface Dimensioning Hardware Dimensioning Software Requirements Spare Requirements Bill of Material Typical Core Network Design Process 5.1. TRAFFIC CALL MODEL Traffic call models provided by different operators are different from each other and no standard model is followed universally. Accordingly, there may be more inputs than what might be required (rarely) or less inputs than what is needed (most often). Considered decisions need to be adopted in instances when insufficient information is available and standard call models as adopted by Motorola may be used as reference for the purposes of computation and customer signoff on the assumptions may be secured.
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Traffic Ratio M 2 M Intra 10% Inter 10% M 2 L 40% L 2 M 40%
GOS Um/Uu 1.00% A/IuCS 0.10% PSTN 1.00% Other interfaces 1.00% PLMN 1.00% Data 1.00%
Traffic Defaults Avg. Traffic per subs 25 mE Total Traffic 7500 VLR Capacity 300,000 Avg. HT per subs 90 s Avg. BHCA per Subs 1 SMS Subs 100% Msg/day/user 4 BHSM 240000 IN Subs 70% Avg. Traffic per subs 25 mE IN Subs Total 210000
Default Utilisation factors E1 Traffic factor 0.8 ATM Traffic factor 0.8 IP Traffic factor 0.3 Traffic Load per SS7 link 0.2
5.2. INTERFACE DIMENSIONING Transmission or link dimensioning involves two aspects viz., the bearer path dimensioning and signalling link dimensioning. For the bearer path dimensioning there are basically two methods available and the first one is based on the use of Erlang B tables (needs Grade of Service and Total traffic in Erlangs as input) while the second one is based on the utilisation ratio of the trunks provisioned (such as 80% etc). Based on the above default values and the reference network diagram the dimensioning of all the transmission links follows (replace 30 with 24 at all instances where E1 basis is to be substituted with T1 calculations).
In case of signalling links as the load distribution of signalling traffic is on the basis of SLS bit which distributes traffic among all the provisioned links. Hence the signalling links have to be provisioned as per the formula Number of signalling links = 2 n Where n can have value from 1 to 4 (indicating the 4 bits in SLS field). This implies the number of signalling links have to be either 2 or 4 or 8 or 16 in order to ensure even distribution of traffic among the links as well as to ensure proper traffic reallocation in case of any link failures etc. All the indicative formulae furnished hereunder follows MS Excel formats.
Signalling links are of different types viz., LSL or Low Speed Links (64 kbps or 56 kbps as the case may be), HSL or High Speed Links (2 M or 1.5 M depending on the market) or Sigtran links (IP FE type links). While engineering signalling link requirement for a particular application, it is typically estimated as x number of LSLs which is later extrapolated into HSLs or Sigtran based on the field conditions as required. Conversion methodology employed to calculate the
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 equivalence of LSL, HSL and Sigtran varies based on the vendor. In case of Huawei, following applies, HSL interface card capability - 12.6 Erl With 1:1 redundancy, capability - 6.3 Erl LSL Erlangs with redundancy - 0.4 Erl (varies on market reqmnt) 16 LSL/ HSL ~ HSL Equivalence in LSL - 6.2/0.4
Similarly in case of Sigtran links,
Sigtran interface card capability - 1280 MSUs/sec (varies with Vendor spec.) Assumed average MSU size - 100 Octets (varies with signalling application) LSL capability @ 0.4 Erl/LSL - 64000 *0.4/8 - 3200 Octets/sec Above expressed in MSUs - 3200/100 - 32 MSUs/LSL Sigtran equivalence in LSL - 1280/32 - 40 LSL/Sigtran 20 LSL/ Sigtran Considering 1:1 redundancy - 40/2 - 5.2.1.1. MSC to BSS Link This link dimensioning pertains to A interface and transmission needs for this interface involves consideration to both the bearer and signalling traffic requirements. As a thumb rule since just one messaging link is enough to cater to about 900 trunks or 30 E1s, the bandwidth requirement for signalling is insignificant in comparison to bearer transmission requirement (ie., one time slot for 30 E1s) hence separate computation is not shown and is assumed incorporated in the overall transmission need as calculated below,
IuCS Dimensioning involves providing ATM interface ie., 155 Mbps if BSS also supports UTRAN ATM interface.
Iu ATM = Roundup((Total_UTRAN1_CS_traffic in Mbps)/155 * 0.8)
A Interface typically involves provisioning at least 2 E1s to carry bearer and signalling traffic with 2 messaging links distributed across each E1 / T1 etc. In case of Erlang B table method of computation for deriving transmission need,
A E1 = Roundup(ErlangB(Total_CS_traffic in Erlangs, Blocking probability)/30)
In cases involving utilisation factor, such as 0.8,
A E1 = Roundup(Total_CS_traffic in Erlangs/0.8/30)
Octet Method:
Depending on the type of BSC such as 2/2.5G BSC which carries BSSAP messages or 3G UTRAN which carries RANAP messages, dimensioning of interfaces follows as below,
On the User plane, traffic is carried as 12.2 K encoded voice which is transcoded in MGW or passed without transcoding as in case of TFO or TrFO scenarios. This 12.2 K voice requires a bandwidth of 22 kbps due to ATM overheads.
5.2.1.2. MSS to HLR Link C interface computation involves evaluation of just the signalling bandwidth requirement. Exact computation method will involve computing the transactions per second that will be exchanged with the HLR for the market of deployment which would in turn need a number of inputs such as how frequently location updates occur, whether authentication is supported, how many SRIs can be expected from SMSC, how often IMEI check is conducted etc. These details may not be
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 available at the design stage and hence typical rule of thumb is used in this case. Rule of thumb for HLR queries is that for every 20K subscribers, 0.4 Erl of traffic is generated (Important : please note that this rule of thumb does not include MNP-SRF, mobile number portability signalling routing function, when performed by the HLR itself). On the HLR side there is also a limitation to restrict the number of signalling links to 8 per E1 with each signalling link engineered to carry 0.4 Erl of traffic this is applicable to Apertio HLS, whereas other manufacturers may have other specifics. There are other HLR manufacturers who recommend only 0.2 Erl of traffic per signalling link resulting from 10K subscriber count and a limitation of just 4 links per E1. Once these limitations are understood, depending on the HLR being considered for the proposal, the following empirical formula may be used to adapt to the situation:
C 64K = Max(Roundup(Number of subs block/Number of links per E1),2)
Number of subs block = Susbcriber count expressed in 10K (in case of 0.2 Erl/link) or 20K (in case of 0.4 Erl/link) Number of links per E1 = 8 (In case of Apertio) or 4 (Depending on the manufacturer specification)
Again it is pertinent to design at least 2 signalling links distributed over 2 E1s in case HLR is located in a distant location to secure redundancy protection over media failures. Depending on the network architecture, HLR interface may be extended through GMSC or directly from the Serving MSC itself and this redundancy needs to be kept in mind from the point where HLR is connected to the current PLMN.
In case of high capacity systems, instead of using 64 kbps signalling link, there are manufacturers who support 2 Mbps signalling link (represented as 2M signalling link), but the ground rules remain the same as only the bandwidth for one signalling link now goes up. Care is again taken to ensure that at least 2 x 2M link is provided for redundacy purposes.
C 2M = Max(Roundup(Number of subs block/32),2)
Octet Method:
Bulk of traffic to HLR primarily results on account of call attempts from subscribers or BHCA attempts. BHCA Attempts/subs = x attempts/hr Total subscriber count = n subscribers Average Octet size per MAP transaction = 300 Octets Octet messages handled per sec = x * n * 300 / 3600 LSL capacity at 40 % loading (octet/sec) = 8000 * 0.4 = 3200 Number of LSLs required = x*n* 300 / 3600 / 3200
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 5.2.1.3. Inter MSC E Interface exists between MSC to MSC within the same PLMN. Traffic that flows in this link includes Inter MSC handover, M2m calls for call termination (here it is assumed that there is an underlying mesh network to interconnect serving MSCs within PLMN). There could be other type of traffic that flows in this interface depending on the network architecture and a proper evaluation of the traffic in Erlangs need to be arrived at (as highlighted in the Section 3.4) before the transmission media requirement is estimated as below. In addition to the bearer traffic that flows between the MSCs there is also signalling relation that exists between these MSCs that handle E interface messages as well as ISUP call termination messages.
Bearer capacity using Erlang B Formula,
E E1 = Roundup(ErlangB(Inter_MSC_Traffic, GOS)/30)
Bearer capacity using Trunk utilisation factor (say 0.8),
E E1 = Roundup(Inter_MSC_Traffic/30/0.8)
64K Signalling link requirement for the E link calculation depends on the fact that at least 2 links are needed and one signalling link catering to every 1000 trunks generates 0.4 erl traffic (thumb rule). Incorporating those cases that might need different link loading (ie., other than 0.4 erl/link) the formula as below,
E 64K = Max(Roundup((E E1 *30/1000)*(0.4/Load_per_SS7link)),2)
Similarly for 2M signalling link (very rare !!)
E 2M = Max(Roundup((E E1 *30/1000/32)*(0.4/Load_per_SS7link)),2)
In addition to the above there are instances when we may have to route the traffic through IP cloud using Fast Ethernet. In these instances please note that there is an IP Transmission Efficiency factor which is much lower, say 0.3, as compared to TDM trunk efficiency factor which is typically maintained at 0.8. On the other hand there is a compression efficiency obtained in bearer compression called Efficiency factor which is missing in traditional TDM method of transmission. Computationally to calculate the FE requirement, E FE = Roundup(Inter_MSC_Traffic_in_Mb*EfficientFactor/100/IPEfficiencyfactor)
If ATM is used to carry this traffic then
E ATM = Roundup(Inter_MSC_Traffic_in_Mb*EfficientFactor/155/ATMEfficiencyfact or)
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In case of Rel 04 with MSC Server and MGW architecture employing IP transmission facility control plane for inter MSC traffic employs M3UA (including MSC <-> GMSC interface) signalling ie., the Nc interface and the dimensioning shall follow as below,
BHCA attempts on inter MSC link = N cBHCA Average MSU size at application layer (BICC) = 150 Octets M3UA Overhead = 35 Octets SCTP Overhead = 48 Octets IP Overhead = 20 Octets Number of messages on average per call = 7
Bearer capacity required to cater to signalling traffic with 0.3 link load,
N c = Roundup((N cBHCA * 253 * 8 * 7)/(3600 * 0.3 * 1024 * 1024),0) Mbps Similarly user plane traffic signified by N b interface when implemented on IP transport has its advantages and disadvantages due to voice compression as discussed earlier. When implemented without voice compression the following is applicable, Bearer traffic = 64 kbps IP Overhead on packetisation = 31.2 kbps Voice Activity Detection factor = 50 % Bandwidth required per call = (64 + 31.2) * 50% = 47.6 kbps Inter MSC MGW traffic on IP = x % MGW Erlangs = MGW erl
N b = Roundup((MGW erl * x % * 47.6)/1024) Mbps 5.2.1.4. MSC EIR Link MSS to EIR link is called the F link that carry just the signalling information to perform the equipment identity check and ensure that the mobile does not feature in the black list or grey list etc. Apertio HLS has capability to function additioinally as EIR and capacity calculation for HLR may be used for this purpose as well.
F 64K = Max(Roundup(Number of subs block/Number of links per E1,0),2) 5.2.1.5. MSC SGSN Link The interface that exists between MSS and SGSN is the Gs interface and carries only signalling information needed by SGSN on the current location details of the mobile as well as page request message from MSS to SGSN. Assumes accomodating 4 signalling links to SGSN on one E1 (to be confirmed with the manufacturer of SGSN). Also minimum of 2 E1 links are needed to provide for failsafe redundancy,
and subs_ block is 10K subscribers. 5.2.1.6. MSC SMSC Dimensioning - Value added services such as SMS and MSS are interconnected using the H interface and only signalling information is exchanged between the entities that carries the SMS messages. Traffic contributed by the SMS messages depends on the average length of the messages which may not be immediately available and hence thumb rules are followed. The SMS traffic is typically measured as BHSM or Busy Hour Short Messages and based on the market, an average of about 40K to 48K BHSMs generate about 0.4 Erl of traffic, accordingly the SS7 links are calculated as below, subject to a minimum condition of 2 E1 links. Also the number of signalling links per E1 varies from manufacturer to manufacturer. Also, designer needs to keep in mind if the SMS server is remotely connected and based on that make a considered decision on the number of signalling links to be accomodated in an E1 to safegaurd against media failures. Following assumes 4 signalling links may be accomodated in 1 E1,
H 64K = Max(Roundup(NumberOfSignalLinks/4,0),2)
Where the number of signal links is calculated on the 48K BHSM basis as below,
NumberOfSignalLinks = Roundup(BHSM/48,0)
Where BHSM is expressed in 1000s of SMS
Octet Method:
SMS traffic calculation is typically split into SMS-MO and SMS-MT traffic because the path followed by SMS on Origination leg is different from path taken by SMS on Termination leg. Whereas SMS-MO is forwarded by the received MSC to the target SMSC (may be through STP), SMS MT results in MAP query to HLR for locating the terminating mobile before SMS is forwarded to the destined mobile. This results in varying traffic on various legs of the network as shown in the figure below.
SMS-MO Calculation:
MSC <->SGW/STP
BHSM per subscriber = x BHSM/Sub/hr Number of subscribers = n Subscribers Total BHSM of subs/sec = x * n / 3600 BHSM/sec Av. size of SMS msg = 200 Octets (varies with market)
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 SMS traffic/sec = 200 * x * n / 3600 Octets/sec Expressed in LSL (@0.4) = 200*x*n / 3600 / 8000 / 0.4 LSL
* AvSMSsize/3600/8000/0.4,0),2) H LSL = Max(Roundup(Tot BHSM
Above traffic calculation holds good for forward and reverse direction on this leg
SMS-MT Calculation:
SGW <->SMSC
In addition to the traffic calculated above in the forward direction, there is additional traffic in this leg of the traffic flow due to routing query to HLR. Part of this traffic is diverted towards HLR by SGW and the remaining part proceeds to the terminating MSC as shown in the diagram below,
5.2.1.7. MSC SCP Dimensioning SCP or Service Control Platform is connected to MSC through an L interface. SCP is typically a database where a number of service logics are performed and implemented to cater to intelligent network services. Services such as Prepaid, Closed User Group, Virtual Private Number, Follow on service, Home zone billing, USSD services, Voucher redemption and a number of other services could be implemented and provided by the operator. GSM uses Camel Service Environment or CSE which functions as SCP and uses CAP (Camel Application Protocol) whereas CDMA uses WINCAP protocol to interact with IN Server. IN traffic calculation is very sensitive to the services hosted and this needs to be kept in mind while actually provisioning the number of links. As an example for a sample system with 300K subs with 70 % IN subscribers ie., 210000 IN subscribers at 1 BHCA would generate 3.89 Erl of traffic (just for pure MO call scenario). This traffic will increase as the number of services supported increases as well as subscription to those services change. INS palettes are available that compartmentalise different services and the transaction per second(TPS) each of them entails. Empirical formulas that may be used as below,
L E1 = Max(Roundup(NumberOfSignalLinks/4),2)
where,
NumberOfSignalLinks = Roundup((TrafficOfCAPs)/(LoadPerSS7*8000) TrafficOfCAPs = NumberOfCAPs*NumberOfTCAPs*LengthOfTCAPs (in bytes)
Elaborating Above:
In case of IN in GSM/UMTS markets, proper estimation of BHCA needs to be arrived at and usually is higher than BHCA for a voice call. This is due to the fact that there are additional attempts due to SMS, Data calls and
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 other services hosted by the IN platform. The traffic estimated here is routed between MSC and IN platform either directly or through STP/SGW.
BHCA for Voice Call = x SMS-MO + SMS MT = y Other Data services etc = z Total BHCA = (x+y+z) - (Typically if x=3, x+y+z = 4.2) Number of Subs = n Total BHCA = (x+y+z) * n = Tot BHCA Av. No. of Txn per call = 4 Av. Transaction size = 124 octets * 4 * 124 /3600/8000/0.4 LSLs Traffic expressed in LSL = Tot BHCA
L LSL = Max(Roundup(Tot BHCA * 4 * 124 /3600/8000/0.4,0),2) 5.2.1.8. MSC PSTN Dimensioning PSTN interface is the Points of Interconnect with the system. Traffic that flows in this interface is both the bearer traffic as well as the signalling traffic. Based on the traffic matrix arrived at (as highlighted in Section 3.4), in the earlier stage, we will have the number of Erlangs for which the transmission medium needs to be provided for.
Bearer capacity using Erlang B Formula,
E E1 = Roundup(ErlangB(PSTN_Traffic, GOS)/30)
Bearer capacity using Trunk utilisation factor (say 0.8),
E E1 = Roundup(PSTN_Traffic/30/0.8)
64K Signalling link requirement for this interface depends on the fact that at least 2 links are needed and one signalling link can cater to every 1000 trunks generating 0.4 erl traffic (thumb rule). Incorporating those cases that might need different link loading (ie., other than 0.4 erl/link) the formula as below,
E 64K = Max(Roundup((E E1 *30/1000)*(0.4/Load_per_SS7link)),2)
Octet Method:
Calculation on the basis of ISUP octet size, as below:
Total BHCA = x * n = Tot BHCA Inter MSC call percent = p % Av. ISUP txn. size = 150 octets Number of Txns per call = 5
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 Traffic expressed in LSL = Tot BHCA * p % * 150 *5/3600/8000/0.4 LSLs
E LSL = Max(Roundup(Tot BHCA * p * 150 * 5 / 3600 / 8000 / 0.4,0),2) 5.2.1.9. MSC PLMN Dimensioning PLMN interface dimensioning follows the same principles as PSTN above. Once the traffic in Erlangs is estimated in this interface, it is possible to arrive at the trunk capacity as well as the signalling bandwidth requirements as above. 5.2.1.10.MSC MGW Dimensioning Mc interface is an IP interface that is responsible for handling H.248 - MGCP/MEGACO messages that are used to setup end points, add and remove connections. In addition to the MGCP messages, if implemented to take care of signalling requirement of Nb interface (information such as IP address, port number, RTP compression etc is exchanged between the MGWs) which typically is the case. Typically, the following messages are considered for the purpose of calculation Notify Request and Reply AuditCapabilities and Reply Add Request and Reply Modify Request and Reply Notify Request and Reply Subtract Request and Reply Average of 9 messages considered per call. Average message size = 150 octets Overheads UDP / IP = 103 Octets (Breakup details of UDP/IP overhead SCTP Overhead = 48 Octets, IP Overhead = 20 Octets, M3UA Overhead = 35 Octets) Total Octets = 253 Link load = 0.3 BHCA for the calls hosted by a particular MGW = MG BHCA which can be calculated as below,
= MG /Av. CHT or MG MG BHCA ERL ERL * BHCA per sub/Erl per Sub
In addition to the above there are other signalling that happens between MGW and MSC server viz., the M2UA links to carry BSS signalling (for BSSAP and RANAP) and M2UA signalling to carry any ISUP traffic. Details as below,
5.2.1.11.SMSC-HLR Dimensioning This interface caters primarily to route query performed by SMSC to locate the terminating mobile location. Details of this calculation is furnished earlier under MSC-SMSC dimensioning.
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 5.2.1.12.IN HLR Dimensioning This interface comes into existence if the network has to comply with CAMEL Phase II and above. Functions like USSD etc if supported by the operator, will require interface provisioning between these two NEs.
5.2.1.13.SGSN HLR Dimensioning This interface Gr is used by SGSN to obtain the subscriber details, authenticate, attach, purge etc. Majority of the messages are Location Update message which are responded to by the HLR with ISD data. Based on the subscription profile, this size could vary but on an average 150 octets may be considered for every transaction, accordingly, Average Location Update message size = 150 octets Number of Transactions per LU procedure = 4 Subscriber base of SGSN = N SGSN Frequency of LUs/hr = P SGSN
5.2.1.14.MSC SGSN Dimensioning - Gs interface is used to exchange location related information between SGSN and MSC/VLR. This interface may be used to page the mobile through Packet interface enabling mobile to monitor a single paging channel and conserving the battery life. This interface is often not implemented.
5.2.1.15.SMSC SGSN Dimensioning Gd interface is established between these two entities and it enables operators to deliver SMS messages through Packet interface. MO SMS are sent as per mobile settings which can be controlled through OTAPA (by the operator) if it will use the packet channel to send SMS or circuit channel to route them. When a mobile station is on GRPS mode all SMS are sent through the Gd interface. Interface dimensioning for this interface will follow the same guideline as in 5.2.1.6 MSC SMSC Dimensioning. Accordingly, SGSN STP/SGW - SMSC, G d LSL = Max(Roundup(Tot * AvSMSsize/3600/8000/0.4,0),2) BHSM
5.2.1.16.MMSC HLR Dimensioning MMSC server interfaces with HLR for route query as well as to find the SMS service centers of MT mobile (used to send MMS notification through SMS message in instances when MMS is not supported by the recipient mobile such as legacy mobiles). Route query message size and dimensioning follows those detailed in Sec 5.2.1.6. MMSC STP/SGW HLR dimensioning
MM = Max(Roundup (Tot * 150/3600/8000/0.4,0),2) 5 LSL BHMM
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5.2.1.17.MSC - MCA Dimensioning Missed Call Alert Server functions to alert the subscribers of the missed call events due to their absence from RF coverage area or power down condition. Typically the call is forwarded to a voice mail system in the event of inability to reach the subscriber and if the subscriber is not subscribed to voice mail services, then MCA server is alerted about the call instance by sending IAM message, upon receipt of which MCA server sends a REL message. This IAM message is used to decode the originators identity and MCA server SMSs the Called party about the missed call event. SMS server delivers the SMS as soon as the called party registers on to the network again.
Average message size of IAM & REL = 100 octets Total Busy Hr Missed Call Events = Tot BHMCA
In addition to the above, for those subscribers who are subscribed to Voice mail, the VMServer can outdial the called party and indicate through CLI the missed call event. Dimensioning for provisioning, as below,
Total Busy Hr Missed Call Events = Tot BHMCA Min Holding time for call to mature = HT min secs
Outdial Erlangs = Tot BHMCA * HT / 3600 Erl min
5.2.1.18.Welcome Server Dimensioning Welcome server serves to welcome In roamers and typically thank out roamers through SMS messages. In addition, promotional campaign services are hosted for roamers through this server. Typically probes are put up at DDF locations where signalling messages are tapped and filtered to detect roamers and appropriate messages are triggered by the server to be sent through SMS to reach the roamers. Messages such as LU, CL, ISD, SRI SM , SRI FORWARDSM , SAI are tapped and its dimesioning as below,
Average size of (LU +CL +ISD +SAI) * Tot +(SRI BHLU SM + SRI FORWARDSM )* Tot BHSM = Tot Avmsg Total Roamers = Tot Roam
Welcome Server LSL reqmnt = Roundup(Tot Avmsg * Tot /3600 / 8000 / 0.3 , 0) Roam
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 5.2.1.19.HLR USSD Dimensioning USSD server is typically a part of IN system which offers a number of Personalised services as well as the popular Call back services. USSD messages when sent by prepaid subscriber are received by Home HLR which in turn initiates dialogue with USSD server to SMS in case of call denial (lack of account balance etc) Initiate Call back through Home MSC or Initiate procedures for Personalised services
Dimensioning rules applicable for both the cases above presented as below,
Average Message size of USSD message = 60 Octets Total Busy Hour USSD requests = Tot BHUSSD (Callback +Personalised services)
To initiate Call Back, USSD server initiates call origination to both A party (USSD Subscriber) and B party, resulting in 2 call legs for every USSD call attempt.
Total Busy Hour USSD Call Back requests = Tot BHUSSDCB Above call back request results in call origination = 2 * Tot BHUSSDCB Average message size for call originations = 400 Octets
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 5.3. STP/SGW DIMENSIONING STP typically combines the functionality of Signalling Transfer point with that of Signalling Gateway function required to interwork with IP transmission path of Control Plane. Based on the signalling relation between various core network entities, STP is dimensioned. Guidelines for using/not using STP for control plane signalling need to be taken based on the factors enlisted earlier and the number of links to be routed between any two signalling network entities is filled in as below for evaluating the interface needs of STP.
M S C
1 M S C
2 P S T N G M S C H L R S M S C I N S G S N E I R V M S / U M S M M S C O T A P A L I G G S N I M S P o C S D P G M L C W A P I M P C S S M S _ C B C MSC 1 MSC 2 PSTN/PLMN GMSC HLR SMSC IN SGSN EIR VMS/UMS MMSC OTAPA LI GGSN IMS PoC SDP GMLC WAP IMPCS SMS_CBC
Many of the fields in the matrix above may have to be blanked out (in addition to those already blanked out) depending on the signalling relation between the two NE concerned. Please note there are various factors such as - vendor specific, solution offered, protocol version dependent etc., that determine if a signalling relation exists or not. Once this table has been arrived at, then all the entries involving a particular NE (such as green highlighted IN platform) is added up to extend so many links from STP. This summed up quantity may be downward revised (by say 10%) due to consolidation efficiency.
Having arrived at the number of LSL equivalents that need to be extended to the STP for all the NEs, following decisions have to be made and implemented, Determination of NEs connection to STP
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 o Distribution of links to be routed through STP vis a vis direct connection. For example IP signalling points IPSPs may be load shared by the MSC itself instead of routing through STP Distribution of LSL equivalents o LSL equivalent of a signalling relation may be distributed into LSL, HSL, Sigtran links in the order of 25%, 25%, 50% depending on the quantity of links o As shown earlier, 1 HSL = 16 LSL 1 Sigtran = 20 LSL Make provision for signalling relation with existing STPs P2P B/D links, say, 2 HSL from each STP Based on the above, quantity of LSL, HSL and Sigtran interfaces that need to be hosted by the STP is calculated and STP is engineered from the vendor. 5.4. HARDWARE DIMENSIONING Once the interfaces are all dimensioned, then the equipment hardware needs to be dimensioned. This section will discuss with the hardware planning that highlights principle behind dimensioning hardware requirements of MSS. MSS is capable of handling GSM, CDMA or UMTS traffic and the capacity of the switch varies depending on the call model. 5.4.1.1. MSS Configuration MSS is a large capacity softswitch capable of handling from 800,000 BHCA (90 s CHT) as per standard call model. MSS can scale to higher capacities based on changes in call model upto about 1,700,000 BHCA (25s CHT) with a corresponding change in hardware that can support these ratings. MSS chasis, also called force chasis is the control switch and comprises of the following cards that needs to be configured.
In addition to the cards that need configuration, there are other elements that need to be considered for design purposes. Motorola Soft Switch Equipment Planning Guide gives a complete description of all the part numbers that needs ordering and the kit details that constitute the same. MSS chasis is shown below as an example extract from the above guide for ready reference. Once the functionality of all the components comprising of MSS chasis is understood, the need for these components becomes self evident. For example EMS Server #1 and #2 are always
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 needed as they are configured in redundant mode for any application. RAID array is needed as a part of the EMS Server. 2 x Cisco ethernet switch is advisable for any commercial application to interconnect various elements of the MSS chasis through 10/100BaseT FE ports. Gigabit Ethernet switches are typically needed if RTP bearer path is to be brought from the media gateway to the control switch or if the configuration entails 2 control switches to work together. Terminal server is a part of the Base kit etc.
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 MSS Chasis
Power Dist Unit iTouch Terminal Server Cisco Ethernet Switches EMS Server #1, #2 RAID Array DAT Backup Server Control Chasis (Force Chasis) Fan & Filter Unit Media Gateway
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06
Author : Page 68 MSS Configurator, maintained at http://compass.mot.com/go/136985679 is a tool available to configure the hardware needs of a particular call scenario. Shown below is an extract from the configurator that gives an idea of capacity of MSS components pertaining to different software releases as well as hardware releases for both GSM and CDMA technologies.
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 Based on the above, depending on the software release intended to be deployed, capacities of various cards can be ascertained. Please note that the capacities highlighted here are basically Lab condition capacities and do not pertain to the real live call scenario conditions which has to take into account Network Factor (NwF). Network Factor denote this critical modifier used in the network call model to account for network events other than RF call-based activities. Network factor components, such as the following, accumulate to form a network factor that enables calculation of total call processing capacity of the MSS, which is referred to as Network BHCA.
Call Redirection Call Delivery Short Message Service (SMS) Message Delivery Packet and Circuit Data Delivery Registration / Location Update Factors Paging Impact Factors Handoffs Prepaid Traffic
Typically these factors are factored in depending on the customer call model. CCSW calculation Based on the MSS release that is planned for ccswitch capacity can be calculated as below,
No of Ccsw cards = Even(Roundup((1+NwF)*Calc_BHCA/Rated_BHCA,0))
Where Calc_BHCA is the calculated BHCA that the MSS is expected to carry and Rated_BHCA is the BHCA capability of the CPU card depending on the hardware and the software release intended to be deployed with.
ADVLR Calculation Again based on the MSS release as well as the hardware intended to be used, the VLR can cater to a capacity of 300K (1.2 GHz) upto 500K (1.6 GHz). Please note at the current release, there can be only one instance of ADVLR (active + backup) that can be run in a switch. Expressed as formula the following may be used,
No of ADVLR cards 2
ADHLR Calculation Similar to VLR, the HLR capacities can range anywhere from 150K (1.2 GHz, 1 GB RAM) to 300K (1.6 GHz, 2 GB RAM) subs per pair of card. Unlike ADVLR process, ADHLR process is scalable and the number of cards provisioned to run HLR process can be augmented in pairs to cater to the market needs. Please also note that HLR process has to be run in pair ie., an active process should always be backed up by another standalone process running in a separate CPU card.
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 The backup card cannot run one more instance of active HLR process as in the case of ccswitch. Expressed as formula the following,
No of ADHLR cards = 2*Roundup(No_of_Subs/CPU_card_cap,0)
Where No_of_subs is the number of subscribers homed in on that MSS and CPU_card_cap is the HLR capacity of the card.
MRS Calculation Media Resource Server is a process run in the CPU card to take care of media mixing need as in case of 3 way calling, legal intercept, announcements (such as for CAMEL/IN working as an Intelligent Peripheral) etc. A single card with MRS process running in it can handle 100 call leg instances. Please note that while MRP server is run in a CPU card, RTP packets of bearer information is mixed and hence it is not backed up with another instance of MRS process etc. Hence based on the engineering need only so many cards are configured, subject to a minimum of 2 cards so as not to loose the functionality of conferencing or LI of MSS. Expressed as formula,
No of MRS Cards = Max(Roundup(No_of_3wc/35,0),2)
Where No_of_3wc is the number of 3 way conference calls handled by the card. As the limitation stems from the total number of call legs that can be handled, in case of 6wc, the number of sessions is limited to 100/6 ~ 16 etc.
SS7MH Calculation SS7 Message handler cards are capable of hosting 4 E1 ports resulting in 128 time slots to bring in the signalling traffic. Each SS7 card consists of a mother card and two daughter cards. Each daughter card supports physical connectivity for two T1 or E1 connections. However, the software on the daughter cards limits each card to 32 links per daughter card. Each SS7 card (mother card + daughter cards) support up to 750 message signaling units (MSUs) per second at 40% utilization (assuming each MSU is of maximum size of 272 octets). Due to this, out of total flexibility of 128 time slots, only 64 time slots could be equipped at any point of time to carry the signalling traffic. Typically all the messaging links are distributed evenly among 2 message handlers to ensure fault tolerance.
No of SS7MH cards = Max(Roundup(No_of_signalling_lnks/64,0),2)
IPMH Calculation IP Message Handlers need to be scaled if there are SIP call instances and currently they just perform the function of load balancing IP messages between the entities within MSS. Since we do not have SIP at the moment, a minimum of 2 process instances are provisioned - in active backup configuration.
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 PCMGR Calculation Typically we provide 2 point code manager process that runs in active standby configuration. This is not a scalable process and is limited to one instance per MSS.
In most instances we do not need to provision separate cards for IPMH, PCMGR processes as they can be multiplexed to run on the same card that hosts MRS or ADVLR process etc.
5.4.1.2. Media Gateway Configuration Media Gateways are primarily responsible for handling bearer traffic and providing switching crosspoints for circuit switched voice. In addition Media gateways also handle Packet traffic as well as conversion between circuit switched voice call and packetised voice and vice versa. MSS being the control server, controls the functions performed by media gateway through the control protocols such as MGCP or MEGACO. MSS provides support for a number of media gateways including Audio Code media gateway and Sonus Media Gateway.
Audio Code Mediagateway Audio code mediagateway currently in use is TP1610 which is a card that can be housed in the MSS control chasis itself. Audio code is very useful for small market applications which can handle a number of signalling applications including R2 protocol. The AudioCodes MGW supports circuit interfaces including DS1 and E1 and is designed specifically to meet the needs of CDMA and GSM networks. The MSS supports multiple AudioCodes MGW cards in the control chassis for scalability. A single AudioCodes TP-1610 card supports 16 T1 or E1 ports for a maximum capacity of 384 T1 or 496 E1 timeslots.
No of Audiocode MGW = Roundup(No_of_required_E1/16,0)
Sonus Mediagateway - The Sonus GSX9000 MGW is a compact, high- performance system that significantly reduces space requirements over legacy circuit switches, lowering operational costs for cooling, power and other facilities. The Sonus MGW supports high density circuit interfaces including T1, DS3, E1 and OC3/STM-1/ STM-0 as well as IP, ATM and POS packet interfaces. The Sonus MGW chassis provides 16 slots, two of which are reserved for redundant Management Network Servers (MNS). The 14 remaining slots can be populated with any combination of Packet Network Server (PNS) or Circuit Network Server (CNS) interface modules. Typically 2 PNS and 2 MNS cards are populated with a variable number of CNS cards. There are a number of varieties of CNS cards available to suit the requirement and following is a summary,
CNS10 - Maximum of 12 T1 spans (288 chls) can be terminated CNS20 - Maximum of 8 E1 spans (240 chls) can be terminated CNS25 - Maximum of 12 E1 spans (372 chls) can be terminated CNS30 - Maximum of 1 T3 span (672 chls) can be terminated
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 CNS60 - Maximum of 3 T3 spans (2016 chls) can be terminated CNS71 - Maximum of 1 OC3 span (2016 chls) can be terminated
5.5. SPARE REQUIREMENTS
For every order of MSS equipment, a list of spares is to be maintained as highlighted in System Equipment Planning Guide which also provides spare requirement calculations for Sonus Mediagateway. An extract of the same for ready information as below,
New Part Number Description Supplier STLN4807A On CS / AD Chassis: STLN6197A SS7 Card w/o RTM Force STLN6196A SS7 RTM Card Force SGDN4628A CPU 700MHz 1GB SPARE Force STLN6186A Ethernet w/o RTM Card Force STLN6185A Ethernet RTM Card Force STLN6188A Alarm w/o RTM Card Force STLN6187A Alarm RTM Card Force STLN6189A Power Supply, CS/AD Force STLN6190A Fan Tray, CS/AD Force STLN6191A Air Filter, CS/AD Force Spare for EMS STLN6166A Linus Server Arrow Spares for RAID (100977) SANnet II SCSI, RAID controller w/ 512 MB cache STLN6172A Dothill STLN6173A SANnet II SCSI, event managenment unit Dothill STLN6174A SANnet II SCSI, RAID IO module Dothill STLN6175A SANnet II SCSI, termination card Dothill STLN6176A SANnet II, SCSI cable, bus config Dothill STLN6177A SANnet II, drive bay air mgmt module Dothill STLN6178A SANnet II, DD, Ultra 160 SCSI, 36GB10k RPM Dothill STLN6179A SANnet II, DC power and cooling mod Dothill STLN6180A SANnet II, DC power cord Dothill STLN6181A Cable, Ultra160, VHDCI-VHDCI, 3 foot Dothill
5.6. SOFTWARE DIMENSIONING
Software modules and the ordering numbers are also highlighted in the System Planning Guide.
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 6. CASE STUDIES 6.1. CASE STUDY # 1 Customer Input:
Customer Inputs
Basic details Total Subscribers 1000000 BHCA 1000000 Total Erlangs 104000 Erl
Call Mix Mobile to Mobile 20% Mobile to Land 40% Land to Mobile 40%
Value Added Services IN Subscribers 60% SMS 100% SMS BHSM 0.5 VMS 100% VMS BHCA 0.4 VMS Mean Holding Time 50 s % of active subs using VMS 20%
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 As we can accommodate maximum of 300000 subscribers per MSS, number of MSS needed for the total subscriber base = Roundup(1000,000/300,000,0) = 4. With 4 MSS configuration, to handle the Mobile to Mobile traffice, we need Inter-MSC trunks.
Inter-MSC call matrix to handle M2M calls :
Erlangs per MSC = 20800/4 = 5200 Incoming into MSC = 5200/2 = 2600 Outgoing out of MSC = 5200/2 = 2600
Outgoing to other MSC = 2600/4 = 650 Incoming from other MSC = 2600/4 = 650
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 Accordingly the network diagram at this stage..
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 MSS BSC Dimensioning sing internetworking trunking efficiency of 0.8, nks to BSC - A E1 = Roundup(Total_CS_traffic in Erlangs/0.8/30) ignalling A 64k = Roundup(1084/30,0) ased on BSC capability, number of BSCs and the signalling links to these BSCs need to SS HLR Dimensioning
U
Li = Roundup((5200+20800)/0.8/30) = 1084 E1s
S = 37
B be decided.
M LR is distributed among 4 MSSs as 250,000 subs/HLR. ssuming 20K subs contribute 0.4 Erl traffic query to HLR, number of links o. of 0.4 Erl Signalling links = Roundup(250000/20000,0) ut of t above links fr S ndup(13*40/60,0) 3*15/60,0) ter MSC Dimensioning
H HLR queries will originate from Other MSSs (20%*3/4) GMSC (40 %)
A
N = 13 O he om GM C = Rou = 9 Links from other MSS = Roundup(1 = 4
In sing internetworking trunking efficiency of 0.8, SS1 <-> MSS2 <-> MSS3 <-> MSS4 = 1300 erl ter MSS link E E1 = Roundup(Inter_MSC_Traffic/30/0.8) ignalling traffic pertaining to voice (assuming 30 E1s traffic can be handled by 1 ax(Roundup(E E1 /30,0),2) herefore the total signalling links between any 2 adjacent MSS is (incl. HLR queries)
U
M
In = Roundup(1300/30/0.8) = 55 E1s
S signalling link of 0.4 Erl), E 64K = M = Max(Roundup(55/30,0),2) = 2
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06
C SMSC Dimensioning = 5 MS MS BHSM Traffic = (No_of_Subs_per_MSS*BHSM) ssuming 0.4 Erl per signalling link for carrying SMS traffic to SMSC, nks to SMSC - NumberOfSignalLinks = Roundup(BHSM/48) umber of E1s - H E1 = Max(Roundup(NumberOfSignalLinks/4),2)
SC IN Dimensioning
S = (250,000*0.5) = 125,000 = 125 K
A
Li = Roundup(125/48) = 3 SS7 links
N = Max(Roundup(3/4),2) = 2 E1s
M otal IN subscribers/MSS = 60%*250000 template) 4 Erl/ k SC GMSC Dimensioning
M earer Traffic = 10400 Erl 0400/0.8/30,0) 0) otal signalling links including GMS HLR = 15 + 9 = 24. ince the total number of signalling links exceed 16, we may have to employ MSSs SC Tandem Dimensioning
B No of E1s = Roundup(1 = 434 E1s Signalling Traffic = Roundup(434/30, = 15 Links
T
S secondary OPC configuration to accommodate this condition.
M earer traffic same as above = 434 E1s MS Dimensioning
B Signalling traffic = 15 links
V MS may be connected to Tandem MSC and the link engineering as below, V
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 Number of CCSW CPU cards = 6 Number of ADVLR CPU cards = 2 Number of ADHLR CPU cards = 2 (Max 300 K subs per card reqd 250 K) Number of MRS Cards = 2 Number of SS7MH cards = 2 (as below)
Signalling links total = 5*3+37+24+15+3+10 = 104 @64 links per card reqd SS7MH = Roundup(104/64,0) = 2
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 6.2. CASE STUDY # 2
Input Parameters
Voice BHCA/sub = 3 Total Subscribers = 45.5 Million Erlang Per Subscriber = 50 mErl IN Subs = 100 % BHSM/Sub = 0.5 Signalling link loading = 30 % M2M % = 60 % M2P % = 10 % P2M % = 30 % Inter MSC BICC traffic = 50 % HLR signalling reqmnt = 2.5 LSL/ 10,000 subs subject to a minimum of 128 links
Derived Parameters
MHT Mean Holding Time = Erl/BHCA = 0.05 x 3600/3 = 60 s
Above capacity is to be distributed over 21 PLMNs which are contained among 3 Regions. Out of the required network elements, some are to be engineered at Region level, some at PLMN level, and some more at Central level and the list furnished as below,
MSC, MGW, GMSC, GMGW PLMN RAN, UTRAN PLMN HLR/HSS PLMN SMSC / SMSC MO gateway Regional IN Platform/VoMS/USSD Regional UMS Regional / PLMN MMS Regional OTAPA Regional LI PLMN SGW/STP PLMN IP/SRP PLMN SGSN PLMN GGSN Regional WAP Regional SDP Regional Central IN Central EIR Central GRX Central GMLC Regional SMLC PLMN
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 User Plane traffic shall be on IP (within PLMN) Interfaces to be engineered for STP/SGW for other elements, standard configuration to be applied. Control Plane traffic assumed to be distributed as 25 % LSL, 25 % HSL, 50 % Sigtran
Capacities of standard configurations categorised under A, B, C for MSC and GMSC servers whereas for Mediagateways, categorised under A, B, C, D & E Capacity required shall be supplied in 3 phases with Phase 1 , Phase 2, Phase 3 having subscriber ratios in the following proportion, GSM : UMTS 75 : 25 50 : 50 25 : 75 Various interfaces from each of these MSC servers identified and provisioned in the Bid requirement, except for some which needs Bidder to calculate. Listed below the interface requirement,
MSC Server Dimensioning
MSC Server Category Remark A B C Subscriber Capacity 400K 200K 100K Expn VLR Cap 600K 300K 150K
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 BHCA 1.2 Mil 0.6 Mil 0.3 Mil CCS7 Links - 64 kbps (min) 512 256 128 CCS7 Links - 2 Mbps - to STP/other nodes 24 12 6 Number of Simultaneous calls that could be handled 40000 25000 15000 50% Inter MSC S traffic Nc Interfaces, BICC 6 Mbps (min) 3 Mbps (min) 2 Mbps (min) Mc Interfaces, H.248 10 Mbps (min) 8 Mbps (min) 6 Mbps (min) On IP 100 Mbps IP Signalling ports Optical 5+5 3+3 2+2 Redundancy ATM Signalling ports for HSL 2+2 2+2 2+2 Redundancy 16+2 port IP Site Router for MPLS VPN 2 2 2
GMSC Server Dimensioning
GMSC Server Category Remark A B C BHCA 1.2 Mil 0.9 Mil 0.6 Mil Erlang 30000 25000 15000 E1 1500 1300 800 STM - 1 25 20 15 STM - 4 5 4 3
Similarly the MGW interfaces to be provisioned as below,
MGW Dimensioning
Traffic Profile Values Remarks BHCA/Sub 3 Traffic/Sub 0.05 IN Calls 100% BHSM/Sub 0.5 GSM Sub 75% WCDMA Sub 25% Link Loading 30% Mobile To Mobile 60% Mobile to PSTN 10% PSTN to Mobile 30% Media Gateway on IP (Nb, Mc) 100% MGW Interfaces
Interface Interface Type Unit Cat A Cat B Cat Cat C D Cat E Remark Traffic Capacity Erlang 16000 12000 8000 4000 2000 (i) GSM No. 14000 10000 7000 3500 2000 Simultaneous call contexts (ii) UTRAN- Voice No. 4500 3500 2300 1200 600
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06
Author : Page 85 (iii) UTRAN- Video No. 180 150 100 50 25 E1 No. 660 495 330 165 83 A Interface STM-1 No. 2 2 2 2 1 lu-CS Interface STM-1 No. 4 3 3 2 2 E1 No. 640 480 320 160 80 MGW- PSTN,USAL,CMSP STM-1 No. 10 8 6 3 2 E1 No. 64 64 48 32 20 E Interface STM-1 No. 2 1 1 1 1 MGW-MSC Mbps 102 77 51 26 24 STM-1 4 4 3 3 2 Nb Interface GE 1 1 1 1 1 Mc Interface No. To be provided by the bidder as per actual calculations AVF No. 20000 15000 10000 5000 2500 Transcoders lu-CS No. 5000 3750 2500 1250 625 Transcoders are to be in a single pool Echo Cancellers No. 20000 15000 10000 5000 3000 LSL No. To be provided by the bidder as per actual calculations SS7 links to BSC, PSTN, GMSC etc HSL No. To be provided by the bidder as per actual calculations Number of Annoucement Devices No. 600 400 300 200 100 No. To be provided by the bidder as per actual calculations SIGTRAN Ports Mbps To be provided by the bidder as per actual calculations IWF Devices No. 100 100 100 100 100
Network Elements supplied as per Bill of Quantity - adhering to geographical and capacity redundancy as well as disaster relief provisioning as per requirements Network Elements such as IN, SMSC, HLR need swapping the existing one and replacing with new one catering to augmented capacity or augmenting the existing one to meet the existing and new requirement so that multiple makes and models of a particular network element is avoided for the Region needs or PLMN needs as the case may be.
Based on the above customer requirement, the following interfaces have to be engineered,
MSC - MGW Interfaces A, Iu Interface Nc, Nb Interfaces MSC PSTN Interface SGW
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06
MSC - MGW Interface Dimensioning Mc (H.248 Component)
This interface between MSC and MGW has to cater to various configuration scenarios based on the models requested. Following the calculations : Required IP bandwidth (Ref 5.2.1.10 for details)
In addition to H.248, M2UA links will be established to handle BSC traffic from 2/2.5G BSSAP subscribers. Bandwidth requirement for the same is shown as below,
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 Proposed BICC traffic, flow in this interface. As highlighted in Sec 5.2.1.3, dimensioning will follow the formula as below,
N c = Roundup((N cBHCA * 253 * 8 * 7)/(3600 * 0.3 * 1024 * 1024),0) Mbps
On application of the above formula, Nc bandwidth requirement in Mbps tabulated as below,
MSC Cat A Cat B Cat C BHCA 1200000 600000 300000 Inter MSC BICC % 50% 50% 50% N C 7.51 3.75 1.88
MSC MSC Interface Dimensioning Nb Interface
In case of User plane traffic carried via Nb interface, bandwidth dimensioning depends on BICC negotiation of an appropriate protocol. Since that is indeterminate, maximum bandwidth is dimensioned, wherein, we assume G.711 compression with a VAD of 50% is applicable. When expressed as formula, following is the bandwidth applicable (for details ref. sec. 5.2.1.3)
N b = Roundup((MGW erl * x % * 47.6)/1024) Mbps
Above when applied on different media gateways, results in the following bandwidth requirement for each of the required configurations (Mbps),
MGW Cat A Cat B Cat C Cat D Cat E Erlang 16000 12000 8000 4000 2000 Inter MGW Traffic % 50% 50% 50% 50% 50% Nb (Mbps) 371.88 278.91 185.94 92.97 46.48
SGW/STP Dimensioning
Signalling Gateways are required to be provided in most of the PLMN based on subscriber population. Since the subscriber population varies widely, the Signalling Gateway dimensioning had to be split into 6 different categories with the following guideline for SGW implementation.
Subscriber Population Model Number <1 Million 1 >1 Million and <2 Million 2 >2 Million and <2.5 Million 3 >2.5 Million and <3 Million 4 >3 Million and <4 Million 5 >4 Million 6
Most of the network elements listed earlier would use the services of SGW/STP to interconnect with the other Network elements. Bulk of this traffic would come from HLR Queries MAP Routing Queries and Location Updates
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 SMSC traffic Both MO and MT traffic IN transactions For all the various services proposed to be hosted ISUP traffic As against the above there would be a lot of variation in traffic that may flow via SGW which would be implementation specific details of which are not available at this stage. Hence a margin factor to the above traffic may be used to make a learned assumption of the traffic to be handled by the SGW. Details of traffic on account of above items calculated as shown below,
MAP Calculations -
MSC <-> SGW (Refer Section 5.2.1.2)
MAP BHCA m/sgs : 3 call attempts per sec per subs Avg Octet size : 300 octets per transaction BH msgs/mil subs : 3000000 Msgs/mil/sec : =3000000/3600 = 833 msgs/sec/million subs MAP traffic in octets : = 833 * 300 = 250,000 octets/sec/million subs MSC map lsl/mil. Subs : = 250000/8000 * 30 % = 104 LSLs from MSC Since this traffic would be routed by SGW towards HLR, traffic towards HLR would be the same as above henc, HLR map lsl/mil. Subs : = 104 LSLs towards HLR
SMSC Calculations -
MSC <-> SGW (Refer Section 5.2.1.6)
MO BHSM m/sgs : 0.5 msgs per sec per subs Avg Octet size : 200 octets per transaction BH msgs/mil subs : 500000 Msgs/mil/sec : =500000/3600 = 139 msgs/sec/million subs SMS traffic in octets : = 139 * 200 = 27,778 octets/sec/million subs MSC sms lsl/mil. Subs : = 27778/8000 * 30 % = 12 LSLs from MSC
Voice BHCA m/sgs : 3 call attempts per sec per subs Other Svcs using IN : 0.5 (SMS) + 0.7 (assume) = 1.2 (SMS, Bal. query etc) CAP BHCA m/sgs : 4.2 msgs per sec per subs
Using the formula as refered above, L LSL = Max(Roundup(Tot BHCA * 4 * 124 /3600/8000/0.4,0),2)
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06
Additionally a 20% margin is added to the above to cater to other services such as location based services, OTAPA etc and above interface capacity is then split into LSL (25%), HSL (25%) & Sigtran (50%). This is further proportioned on the basis of subscriber population and growth expected in Phase 1, 2, 3. On the basis of PLMN categorisation to belong to a particular model, interfaces are provisioned for the highest subscriber population within that model bracket.
HLS/HSS Dimensioning
Required to supply HLR of following configurations, 1 million, 2 million, 3 million, 4 million, 5 million and 6 million subs capacity. Geographical redundancy with Disaster recovery required, hence 2+1 configuration offered. Per the RFP requirement, following are the number of signalling links to be provisioned,
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 Sun Quad Ethernet Cards For DIAMETER (HSS) 3 3 3 3 3 3 Sun Quad Ethernet Cards For DIAMETER 2 2 2 2 2 2 Phase 2 Adax PCI Cards (HLR) 3 4 6 8 9 13 Sun Quad Ethernet Cards For SIGTRAN (HLR) 3 4 6 7 9 13 Sun Quad Ethernet Cards For DIAMETER (HSS) 3 3 3 3 3 3
Phase 3 Adax PCI Cards (HLR) 3 4 6 8 9 13 Sun Quad Ethernet Cards For SIGTRAN (HLR) 3 4 6 7 9 13 Sun Quad Ethernet Cards For DIAMETER (HSS) 3 3 3 3 6 6
As above, each FE is provided with 1 Adax card. Each Adax card can terminate 4 E1 links. Each link has 31 signalling time slots hence each card can handle 31 x 4 = 124 signalling TSs. This results in a total signalling interface of 124 x 3 = 372 LSL. In addition 3 Sigtran interface cards are provided and each Sigtran card can handle 124 LSL equivalents resulting in a total capacity of 2 x 124 = 372 LSL. Combined total therefore is 372 + 372 = 744 LSLs which is more than 424 LSLs as requested in the RFP for 1 million subs configuration.
SMSC Dimensioning
SMSC interfaces with MSC, HLR, IN (for CAMEL PH III & above) through SS7 interfaces. SMSC interactions with other network element such as VMS, UMS, OTAPA, WAP etc happens through the SMPP protocol. As for MSC and HLR interactions, RFP states the requirements to be at least 512 LSL equivalents (to cater to MO SMS traffic as well as other interfaces such as Gs, IN platform etc)and for expansion phases, the calculation is performed as highlighted in Section 5.2.1.6. Considering an average SMS of 240 octets/message, and using the formula for Phase 2 and 3 as below
* AvSMSsize/3600/8000/0.3,0),2) H LSL = Max(Roundup(Tot BHSM
wherein,
Tot BHSM /zone as per RFP (Ph 2,3) = 1500000 AvSMSsize = 240
No of LSL for SMS = 1500000 * 240 / 3600 / 8000 / 0.3 = 42 lsls
Above LSLs if needed to be provisioned as Sigtran interface, then the bandwidth needed for the same would have to accommodate instead of MTP 2/ MTP3 overheads, SCTP/IP overheads. SCTP/UDP overheads approximate to 100 octets/message (M3UA ~ 25 + SCTP ~ 48 + IP ~ 20 Octets)
___________________________________________________________________________________________ Issue :One Date : 20 Nov. 06 Depending on the actual planning need - part or whole of traffic may be transported through Sigtran and/or LSLs.
MMS Dimensioning
UMS is unified messaging system that takes care of Voice mail as well as Welcome service functionalities. This interfaces with HLR for locating MT subscribers SMSC on the basis of which SMS is sent to the mobile in case of legacy mobile and MMS forwards the MMS message to the mobile itself. Route query dimensionining as in Section 5.2.1.16
= Max(Roundup (Tot * 150/3600/8000/0.4,0),2) MM 5 LSL BHMM
As per RFP Tot BHMM = 600000 Therefore MM = 600000*150/3600/8000/0.3 = 11 LSL 5lsl
MCA Dimensioning
Missed Call Alert is detected by UMS server which outdials to Voicemail subscribers a missed call to the intended recipient of call to notify of the missed call event. As explained bandwidth required for outdial in Sect 5.2.1.17,
Outdial Erlangs = Tot BHMCA * HT / 3600 Erl min
Per RFP MCA subscribers for one zone = 2184000 Percentage of this subscriber using outdial = 1 % => 21840 Total incoming calls per day = 8 Percent of calls happening during BH = 10 % Outdial calls in BH = 21840 * 8 * 10 % = 17472 HT min assumed = 15 secs Erlangs of traffic generated due to outdial = 17472 * 15 /3600 = 72.8 Erl Number of trunks needed for 5 % blocking = 80 trunks = 3 E1 s
Welcome Server Dimensioning
Welcome server is employed in welcoming roamers into the current network. As explained in Sec 5.2.1.18, the LSL requirement is,
Welcome Server LSL reqmnt = Roundup(Tot Avmsg * Tot /3600 / 8000 / 0.3 , 0) Roam
Where Tot Avmsg = (LU +CL +ISD +SAI) * Tot +(SRI +SRI )* Tot BHLU SM FORWARDSM BHSM
As an eg., plugging values for above events may be approximated as below for one sub/day, (120 +120 +810 +110)*4 +(270 +110)* 0.5 =4830 Octets /sub/day
Per RFP calls occuring during BH = 10% Per RFP number of Roamers (5% of population) = 108333 Total traffic to be handled = 108333 * 4830 * 10 % = 53234839 Octets /hour
USSD provided in the network had to cater to a number of services such as personalised services, Call back services etc. Dimensioning of the same as illustrated under Sec 5.2.1.19, reproduced below, = Max(Roundup(Tot * 60 / 3600 / 8000 / 0.3, 0),2) HLR USSD LSL BHUSSD
USSD is a regional element and the capacities of subscribers it needs to cater to for a representative region as per RFP is 15,480,000.
Number of USSD req/hr/million subs (given) = 12000 * 20 % Personalised USSD req for subs base/sec = 15480000 * 12000 * 20% / 1000000 /3600 = 11 Number of Callback req/hr/million subs (given) = 5000 Callback USSD req for subs base/sec = 15480000 * 5000 / 1000000 / 3600 = 22 = (11 + 22) * 3600 = 33 * 3600 Total USSD requests per BH - Tot BHUSSD