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Prerequisites 1. Basics of information and communication technologies (ICT): networks architecture, networks components, networking skills; 2. Basics of OSI/ISO protocol stack; 3. Basic knowledge in the area of Next Generation Network concept (Module 1); Learning goals Main objective of this module is to acquire basic knowledge in the area of Next Generation Network architectures, its principles network components and evolutions steps. Key emphasis is put on architectures supporting transfer of the voice over IP networks. Trainee will also get basis of NGN architectures based on a Softswitch configuration as well as the architecture of IMS including characteristics of their individual components.

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Although the hybrid networks (PSTN/IN networks interworking with IP networks, such as Internet) offered a source of new services, only the development of new packet network architectures, the so-called VoP (Voice over Packet network) and especially VoIP (Voice over IP), made it possible to provide the integrated services of voice and data transfer over a single network. There are several networks enabling voice or multimedia transfer by packets. This module deals with the issue of voice transfer over IP networks (networks using IP protocol). Other examples of VoP networks are ATM (VoATM) and Frame Relay (VoFR). Even if the majority of VoP technology suppliers have their own network architecture, the basis of VoP is mostly a managed IP network. Some new components of this platform (e.g. gateways) allow provision of voice services over the packet network and its interworking with the existing telecommunication network (classical PSTN).

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1.1 VoIP architecture


Integrated VoIP architecture consists of three main network types. In sequence, the first one is IP Access network which provides access to the service for user. This is based on packet data networks using IP protocol mainly. The second one is IP backbone network; it ensures transport channels between all of points connected to the access network with providers elements of VoIP service. IP backbone network uses IP based architecture as well. But a lot of providers use ATM of FR based networks as a backbone. Usage of these networks is decreasing nowadays. Important part of VoIP architecture is connection between users which doesnt use VoIP services and use only POTS services with VoIP users. The third one network, known as Public Switched Telephone Network is an area of all existing POTS users. Basically each VoIP architecture must have these three parts, whether is it SIP based architecture or H.323 architecture or another not standardized architecture. On Fig. 1 base SIP architecture is depicted. VoIP architecture based on SIP protocol includes some specific elements. Namely SIP User Agent, SIP server and SIP to PSTN gateway. Detailed information about these elements see Module 3 Unit 2 SIP protocol.

Fig. 1: SIP based VoIP architecture

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1.2 Voice transfer over IP networks


Voice transfer over IP networks is usually connected with thinking about transfer by protocols of the TCP/IP architecture. The model of TCP/IP architecture has 4 layers (network interface, network, transport and application layers) in contrast to 7 layers of the OSI reference model (physical, switching, transport, relation, presentation and application layers). Voice sample is, therefore, input into RTP/UDP/IP packets and subsequently for example into the Ethernet frame, which is transferred over the LAN or WAN network. The application or terminal equipment then inputs data, including voice, into the RTP application protocol (description of the RTP protocol can be found in Module 3 Unit 4) and that one is input into the UDP datagram, which is transferred by IP packet to the target destination, where it is transferred to sound by the reverse process. The network layer (IP layer) ensures the transfer of packets over the network, and their routing in different network groupings respectively. The transport layer ensures the flow of data between two terminal nodes of the network two communicating entities. In this layer, two protocols - TCP and UDP - are used. TCP enables reliable data transfer, provides services with connection and receipt confirmation. The UDP protocol provides services without connection and without receipt confirmation (Fig. 2).

Fig. 2: Voice sample and signalling transfer over the packet network

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NGN architecture based on software switching technology (softswitch) can be supposed as first and unique evolution step in NGN, although there are more modern architecture available nowadays (for example see Unit 3 - IMS based architecture). However, it has build up philosophy of building of new NGN networks and validated principles and features of NGN architecture and its components towards its next evolution. This architecture was the first which drive was significantly motivated by telco vendors, naturally reflecting on massive development of VoIP protocol family and by telco providers demand to implement services more modern and more efficient way. Due to this fact it is not strictly standardized and we can see there several different attitudes of telco vendors how to provide some features, how to distribute components across network including distribution of functional entities inside control element plane. One of the major advantages of the softswitch architecture is separation of call set-up from the physical switching that is handled by the media gateways. Separation here means that the same software platform can telco operator use for various physical transport media, he can employ the use of standards and finally to use commercially available standard server platforms. Flexibility and scalability of this architecture allows for smaller and more innovative vendors to be specialized in just one function and another vendor in different ones. Knowledge of this architectural approach is important for understanding of next evolved architectures and principles. Components of softswitch based architecture you can see in Figure 3.

Fig. 3: Components of softswitch based architecture Media Gateway Controller/call agent/softswitch: generally serves as components for controlling of

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communication relations of users and other network components; provides call routing, network signaling, billing, and other logical functions. Media Gateway: operate within transport plane, perform all function related to media physical transport between different networks, media processing functions (transcoding, echo cancellation, jitter managing), tones processing and management of information transport Trunking gateways: interface between the PSTN/PLMN and VoIP network Residential gateways: provide traditional analog (RJ11) interface to VoIP network Access gateways: provide traditional analog or PBX interface to VoIP network Signaling network: provide change of signalization systems between PSTN or PLMN network to VoIP network Application Server: it is obviously implemented to perform functionalities specific to certain service, perform specialized service logic call control, also includes more functionalities in terms of user web interface, end-points management, etc. For example it can provide specific videoconferencing service, Call Centre service or IP Centrex service. Media Server: provides functionalities allow interaction between calling party and application using end-point device. It provides Media Resource Functions (tones detection, speech synthesis and recognition, compressions, media mixing, etc.) and Media Control Functions control of media functions (voce message play management, conference bridge, fax messages management, etc.).

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The initiative of organization institutions 3GPP within the specifications of UMTS architecture (3GPP within the UMTS architecture 5/6 [2]) has defined two domains: Circuit switching domain, Packet switching domain. The packet switching domain extends the existing GSM network and other mobile 2nd generation (2G) networks by the CDMA-based access, while the packet switching domain extends the abilities of the GPRS and other systems of 2.5 generation. The subsystem for supporting multimedia services, telephony and IP-based message sending, designed in the framework of the packet switching domain is called IMS (IP Multimedia Subsystem). IMS is based on the IP architecture for multimedia and it was placed as a supporting network element to provide standardised and universal services for mobile users. As it was one of the first concepts on which all the standardisation institutions agreed and which conformed to the NGN principles, it is becoming one of the reference concepts for the fixed networks as well.

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3.1 3GPP
The 3GPP adopted SIP [1] (described in Module 3 Unit 2), which was originally standardized by the IETF. In time, the 3GPP discovered that there were gaps between the SIP, as initially defined by the IETF, and the features that were required to provide full support for IMS networks. Because SIP did not address all the requirements of IMS networks, the 3GPP subsequently defined dozens of new SIP extensions that are specific to IMS networks. Collectively, these extensions comprise the IMS SIP protocol, which is defined in the 3GPP TS.24.229 standard [3]. IMS SIP extensions, such as extended call control, presence and instant messaging, extend the functionality of SIP on IMS networks. By definition, SIP is not a protocol designed for a specific network or application. To use SIP, you can define a usage profile can be defined. Usage profiles work much like templates, and provide a varied, flexible environment for application development in which you can easily develop an application suited to your particular requirements can be easily developed. In effect, this is what IMS SIP did. The IMS SIP usage profile is the most important in the telecommunications industry, as it affects the entire telecom industry and not only mobile networks. The usage profile used by IMS SIP is actually the most appropriate for NGN networks. There are numerous IMS SIP extensions, each of which are described in the Module 6 that follow. The figure below illustrates a typical IMS network [4]. Note that all SIP interfaces are shown in orange and specify the name of the interface between two adjoining entities. For example, the AS uses the ISC interface.

Fig. 4: SIP in IMS


HSS

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Home Subscriber Server (HSS) is a combination of the currently existing UMTS/GSM HLR and the needed register functions for IP Multimedia Subsystem. HSS will provide the following functions: user identification, numbering and addressing information. user security information: Network access control information for authentication and authorisation user location information at inter-system level; HSS handles the user registration, and stores intersystem location information, etc. the user profile (services, service specific information) [4]
P-CSCF

Proxy Call State Control Function (P-CSCF) performs the following functions: is the first contact point for UE within IM CN subsystem, forwards the registration to the I-CSCF to find the S-CSCF and after that forwards the SIP messages between UE and I-CSCF/S-CSCF behaves like a proxy in RFC 2543 [5], i.e. accepts requests and services the internally or forwards them possibly after translation may behave also like a RFC 2543 [5] User agent i.e. in abnormal conditions it may terminate and independently generate SIP transactions is discovered using DHCP during registration or the address is sent with PDP context activation may modify the URI of outgoing requests according to the local operator rules (e.g. perform number analysis, detect local service numbers) detects and forwards emergency calls to local S-CSCF generation of charging information maintains security association between itself and UE, also provides security towards S-CSCF provides the policy control function (PCF) authorisation of bearer resources, QoS management and Security issues are currently open in standardisation [4].
I-CSCF

Interrogating Call State Control Function (I-CSCF) performs the following functions: is the contact point within an operators network for all connections destined to a subscriber of that network operator, or a roaming subscriber currently located within that network operators service area. It can be regarded as a kind of firewall between the external IMSS and the operators internal IMSS network. There may be multiple I-CSCFs within an operators network assigns a S-CSCF to a user performing SIP registration routes a SIP request received from another network towards the S-CSCF obtains from HSS the Address of the S-CSCF charging and resource utilisation in performing the above functions the operator may use I-CSCF to hide the configuration, capacity, and topology of the its network from the outside additional functions related to inter-operator security are for further study
S-CSCF

Serving Call State Control Function) (S-CSCF) performs the following functions: performs the session control services for the terminal. Within an operators network, different SCSCFs may have different functionality it maintains session state and has the session control for the registered endpoint's sessions acts like a Registrar defined in the RFC2543 [5], i.e. it accepts Register requests and makes its information available through the location server (e.g. HSS) may also behave as a proxy or as a user agent as defined by RFC 2543 [5] interacts with Services Platforms for the support of Services

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obtain the address of the destination I-CSCF based on the dialled number or SIP URL on behalf of a UE forward the SIP requests or responses to a P-CSCF or an I-CSCF if an I-CSCF is used in the path in the roaming case generates charging information security issues are currently open in standardisation [4]
MGCF

Media Gateway Control Function (MGCF) Provides the following functions: protocol conversion between ISUP and SIP routes incoming calls to appropriate CSCF controls MGW resources [4]
MGW

Media Gateway (MGW) provides the following functions: Transcoding between PSTN and 3G voice codecs Termination of SCN bearer channels Termination of RTP streams [4]
T-SGW

Transport Signalling Gateway provides the following functions: maps call related signalling from/to PSTN/PLMN on an IP bearer provides PSTN/PLMN <-> IP transport level address mapping [4]
MRF

Multimedia Resource Function provides the following functions: Performs multiparty call and multimedia conferencing functions [4]
BGCF

The S-CSCF, possibly in conjunction with an application server, shall determine that the session should be forwarded to the PSTN. The S-CSCF will forward the Invite information flow to the Breakout Gateway control function (BGCF) in the same network. The BGCF selects the network in which the interworking should occur based on local policy. If the BGCF determines that the interworking should occur in the same network, then the BGCF selects the MGCF which will perform the interworking, otherwise the BGCF forward the invite information flow to the BGCF in the selected network. The MGCF will perform the interworking to the PSTN and control the MGW for the media conversions. 3.1.1 Self-SIP in IMS SIP and SDP as a protocol has been selected to some and IPv6 as the only solution to all of the IP Multimedia Subsystem interfaces. As shown by the Figure 5 the basic SIP [1] has been selected as the main protocol on the following interfaces: Gm: P-CSCF UE Mw: P-CSCF S-CSCF and P-CSCF I-CSCF Mm: S/I-CSCF - external IP networks & other IMS networks

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Mg: S-CSCF BCGF Mk: BCGF external IP networks & other IMS networks

Fig. 5: SIP protocol in IMS [4] Eventually there may be differences in the SIP procedures of Gm and Mw reference points. This implies that there is a difference in UNI and NNI interfaces [4]. The following procedures have been defined for the 3GPP IM subsystem in [4]: Local P-CSCF discovery: Either using DHCP or carrying address in the PDP context S-CSCF assignment and cancel S-CSCF registration S-CSCF re-registration S-CSCF de-registration (UE or network initiated) Call establishment procedures separated for Mobile origination; roaming, home and PSTN Mobile termination; roaming, home and PSTN S-CSCF/MGCF S-CSCF/MGCF; between and within operators, PSTN in the same and different network Routing information interrogation Session release Session hold and resume Anonymous session establishment Codec and media flow negotiation (Initial and changes) Called ID procedures Session redirect Session Transfer 3.1.2 SIP in Service SS The service subsystem and its connections to IM subsystem is shown in the Figure 6. The S-CSCF

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interfaces the application development servers with SIP+ protocols. The SIP application server can reside either outside or within operators network [4]. The OSA capability server and Camel refer to already standardised 3G and GSM based service generation elements.

Fig. 6: Service Subsystem connections with IMS SIP is used to interface the Application servers on the following interfaces: S-CSCF- SIP Application server S-CSCF- Camel Server S-CSCF-OSA Service Server

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3.2 TISPAN
The TISPAN network architecture is based on 3GPP IMS, which is a basis for control and provision of the real-time conversation services (based on SIP protocol) [6]. 3GPP IMS architecture is extended in TISPAN NGN to support various types of access networks, such as xDSL, WLAN, etc. TISPAN architecture is extended mainly by: Access networks control (QoS, access control and authentication), Co-ordination of various control subsystems via one transport network to control resources, Interworking and interoperability with public networks (legacy networks), Separation of the application layer from the connection control layer and the transport layer, Independence of access technologies from the call control layer and the application layer. For services on other than SIP basis, the TISPAN NGN architecture can include other subsystems defined in TISPAN [6]. Figure 7 illustrates the NGN components and functionalities.

Fig. 7: Architecture of a TISPAN NGN [6]

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3.3 PacketCable
The PacketCable architecture is based on the IMS architecture, but PacketCable initiative leaded by cable operators to enhance the IMS to support the unique technology requirements of the cable industry, and also addresses cable operator business and operating requirements. Extensions include use of additional or alternate functional components compared with the IMS architecture, as well as enhancements to capabilities provided by the IMS functional components. Some of the PacketCable enhancements to the IMS include [7]: Support for Quality of Service (QoS) for IMS-based applications on DOCSIS access networks, leveraging the PacketCable Multimedia architecture; Support for additional access signaling security and UE authentication mechanisms; Support for provisioning, activation, configuration, and management of UEs; Support for regulatory requirements such as number portability, preferred carrier and PacketCable lawful interception. An overview of the PacketCable architecture elements and functional groupings is illustrated in Figure 8.

Fig. 8: Architecture of a PacketCable 2.0 [7]

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3G 3GPP AS ATM BGCF BICC BSF CCS7 CDF CDMA CGF CMS CMTS DHCP DNS DSL EMS ENUM FR GPRS GSM HLR HSS IBCF ICSCF ID IMS IN IP ISUP LAN MG MGC MGCF MGCP MGW MRF NAT NMS NNI OSA OSI PAC

Third Generation 3 Generation Partnership Project Application Server Asynchronous Transfer Mode Breakout Gateway Control Function Bearer Independent Call Control Bootstrapping Server Functionality Common Channel Signaling 7 Charging Data Function Code Division Multiple Access Charging Gateway Function Call Management Server Cable Modem Termination System Dynamic Host Configuration Protocol Domain Name Server Digital Subscriber Line Element Management System Telephone E.164 Number Mapping Frame Relay General Packet Radio Service Global System for Mobility Home Location Register Home Subscriber Server Interconnection Border Control Function Interrogating Call State Control Function Identifier IP Multimedia Subsystem Intelligent Network Internet Protocol ISDN User Part Local Area Network Media Gateway Media Gateway Controller Media Gateway Control Function Media Gateway Control Protocol Media Gateway Multimedia Resource Function Network Address Translation Network Management System Network Network Interface Open Services Architecture Open System Interconnection Provisioning, Activation and Configuration

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PCF PCSCF PDP PLMN POTS PSTN QoS RFC RTCP RTP SCN SCS SCSCF SIP SLF SQ SS STUN TCP TDM TISPAN TrGW TSGW UDP UE UMTS UNI URI VoATM VoFR VoIP VoP WAN WLAN

Policy Control Function Proxy Call State Control Function Packet Data Protocol Public Land Mobile Network Plain Old Telephony Service Public Switched Telephone Network Quality of Service Request For Comments Real Time Control Protocol Realtime Transport Protocol Switched Circuit Network Service Capability Server Serving Call State Control Function Session Initiation Protocol Subscription Locator Function Signaling Gateway Service Subsystem Session Traversal Utilities for NAT Transport Control Protocol Time Division Multiplexing Telecommunications and Internet Converged Services and Protocols for Advanced Networks Transition Gateway Transport Signalling Gateway User Datagram Protocol User Equipment Universal Mobile Telecommunications System User Network Interface Uniform Resource Identifier Voice over ATM Voice over Frame Relay Voice over IP Voice over Packet Network Wide Area Network Wireless LAN

SIGTRAN Signaling Transport Group

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[1]

M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg: SIP: Session Initiation Protocol, IETF RFC 2543, Mar. 1999 3GPP: TS 24.229 Internet Protocol (IP) multimedia call control protocol based on Session Initiation Protocol (SIP) and Session Description Protocol (SDP); Stage 3 Radvision: "IMS SIP and Signalling, The Radvision Perspective - Technology overview", 2006 http://www.radvision.com/Resources/WhitePapers/ims_sip.htm

[2] 3GPP TS 23.002 V5.12.0 (2003-09), Network architecture, (Release 5) [3]

[4] 3GPP: TS 23.228 IP Multimedia Subsystem (IMS); Stage 2 [5]

[6] ETSI ES 282 001, (TISPAN); NGN Functional Architecture Release 1 [7] PacketCable 2.0 Architecture Framework Technical Report, PKT-TR-ARCH-FRM-V03-070925, 2007

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1. For relation management in VoIP networks are used following signalization protocols a. b. c. d. H.323, SIP, RTP RTP, ATM, H.323, CCS7 SIP, RTP, ATM SIP, H.323

2. Basic physical elements in distributed NGN architecture are: a. b. c. d. media gateway control (MGC) and application server (AS) media gateway (MG) and application server (AS) media gateway control (MGC) and media gateway (MG) media gateway (MG) and media server

3. Media gateway (MG) is responsible among others for a. b. c. d. Adaptation of PCM and RTP/IP media transport Provide application logic for specific relation Control function for signalization between end-point users attended on relation Interaction between calling party and application using end-point device.

4. Trunking Gateway a. b. c. d. manages digital virtual circuits and main circuits of TDM works like an interface for wireless network provides data access to the IP network allows interaction between calling party and application using end-point device.

5. In the context of NGN architecture the IMS stands for a. b. c. d. Instant Messaging Subsystem Internet Media System IP Multimedia Subsystem IP Management System

6. Concept of IMS-3GPP system is based on layers: a. b. c. d. application, transport, media and access application, relation control, media and access application, transport and access application, transport and network

7. Reference point ISC uses following protocol:

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a. b. c. d.

RTP H.248 Diameter SIP

8. Reference point between CSCF and HSS: a. b. c. d. Sh Cx Si Mr

9. Conceptual model of NGN defined in ETSI (TISPAN) a. b. c. is used only as a substitution for PSTN network and services is able to provide and control conversational services in real time defines storing of services and customer profiles for every application server separately d. doesnt support various types of access networks 10. What doesnt Proxy Call State Control Function (P-CSCF) perform? a. b. c. d. It realizes protocol conversion between ISUP and SIP. It detects and forwards emergency calls to local S-CSCF. It generates the charging information. It provides the policy control function (PCF).

11. Multimedia Resource Function a. b. c. d. routes a SIP request received from another network towards the S-CSCF; provides transcoding between PSTN and 3G voice codecs; maps call related signalling from/to PSTN/PLMN on an IP bearer; performs multiparty call and multimedia conferencing functions;

12. TISPAN architecture is extended mainly by integration of the application layer with the connection control layer and the transport layer; b. dependence of access technologies on the call control layer and the application layer; c. interworking and interoperability with public networks (legacy networks); d. transport networks control; a.

Correct:

Wrong:

Blank:

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