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International Journal of Electronics Communication and Computer EngineeringVolume 3, Issue 2, ISSN 2249 –071X
Technical Specification Group Services and SystemAspects, IP Multimedia Subsystem (IMS)
Ashwini Patil Department ofInformation Technology, Bharati
Vidyapeeth Deemed University CollegeOf Engineering, Pune-46
H. K. SawantDepartment of Information Technology,Bharati Vidyapeeth Deemed University
College Of Engineering, Pune-46
Abstract —The paper is aimed at studying and analyzingthe network performance parameters of SIP protocol. SIP iscontent based protocol, in which various message arerequired to be transacted so that a session could be created,terminated or modified. Therefore, the objective is to analyzevarious SIP activities and the delay incurred in session start-up under various network conditions. The overall project issplit into key areas like Networking, Traffic modeling,Simulation and Reporting. The objectives behind the workare as follows: Getting acquainted with the concept of IMS,SIP and their functionalities, To define a Hybrid NetworkTopology consisting of the IMS entities and the fixed andmobile SIP User Agents, To set the proper configuration ofthe nodes such that they are able to communicate with eachother properly, To identify the parameters based on whichthe performance evaluation will be done, To find the optionsof specifying different types of traffic flow in the network, Tovary the parameters for different scenarios and simulatethem, To study the variation in the results and analyze theperformance of SIP protocol in the network.
Key Words — SIP protocol, SIP activities and the delayincurred in session, IMS platform, DIAMETER- improvisedversion of RADIUS protocol.
I. INTRODUCTION
In the past few years, the evolution of cellular networkshas reflected immense success and growth which wasexperienced by Internet in the last decade. This leads tonetworks where Internet Protocol connectivity is providedto mobile nodes. The result is third generation (3G)networks where IP services such as voice over IP (VoIP)and instant messaging (IM) are provided to mobile nodes(MN) in addition to connectivity. With import of Wi-Fi,Wi-MAX, digital video broadcasting, satellite, internet,etc. the current architecture of telecommunication networkhas been facing a challenge of interoperability.
Hence, the solution was to devise an interoperableplatform which can provide the services irrespective of theaccess technology. The basic approaches to convergenceare:
UMA: Unlicensed Mobile Access (UMA) is a newtechnology that provides access to GSM services overWireless LAN or Bluetooth. It also challenges theassumption of closed platform, since it is relatively easy toimplement a UMA phone purely in software running onstandard PC hardware and operating systems. In the UMAsolution, exiting cellular network remains unmodified, anda new network element, the UMA Network Controller(UNC), is introduced. UNC acts as a gateway between themobile operator core network and Internet or a broadbandIP access network such as ADSL or cable. The phoneconnects to the IP network using a standard WLAN or
Bluetooth access point. Since GSM/GPRS core securitymechanisms; new mechanisms are defined only forprotecting the communication between the phone andUNC.
SIP: The Session Initiation Protocol (SIP) is a signalingprotocol used for establishing sessions in an IP network. Asession could be a simple two-way telephone call or itcould be a collaborative multi-media conference session.The ability to establish these sessions means that a host ofinnovative services become possible, such as voice-enriched e-commerce, web page click-to-dial, InstantMessaging with buddy lists, and IP Centrex services. SIPis a request-response protocol that closely resembles twoother internet protocols, HTTP and SMTP; consequently,SIP sits comfortably alongside Internet application. UsingSIP, telephony becomes another web application andintegrates easily into other Internet services. SIP is asimple toolkit that service providers can use to buildconverged voice and multimedia services. SIP is an“application-layer control (signaling) protocol for creating,modifying, and terminating sessions with one or moreparticipants. These sessions include Internet telephonecalls, multimedia distribution, and multimediaconferences.”
IMS: IP Multimedia Subsystem (IMS) is referred as theheart of NGN. The 3GPP has published a number ofspecifications that define the IMS as the part of the 3Gwireless environment which will “enable the convergenceof, and access to, voice, video, messaging, data and web-based technologies for the wireless user.” The 3GPPdefined the IMS specifications in support of the UniversalMobile Telecommunication System (UMTS). The UMTSis the 3G evolution of the GSM.
IMS is the envisioned solution that will provide newmultimedia rich communication services by mixingtelecom and data on an access independent IP basedarchitecture, defined in 3rd Generation Partnership Project(3GPP), 3rd Generation Partnership Project 2 (3GPP2) andInternet Engineering Task Force (IETF) standards.
The aim of IMS is to provide all the services, currentand future, that the Internet provides with roamingfacilities. To achieve these goals, IMS supports peer-to-peer IP communications between existing technologystandards while providing a framework for inter-operability of voice and data services for both fixed(POTS, ISDN) and mobile users (802.11, GSM, CDMA,UMTS). It provides session control, connection controland an application services framework with bothsubscriber and services data, while allowinginteroperability of these converged services betweensubscribers. IMS truly merges the Internet with the cellularworld; it uses cellular technologies to provide ubiquitous
International Journal of Electronics Communication and Computer EngineeringVolume 3, Issue 2, ISSN 2249 –071X
access and Internet technologies to provide appealingservices.
Figure 1: IMS Overview NGN Convergence
II. LAYERED ARCHITECTURE OF IMS
The definition of IMS by 3GPP is an all packet corenetwork that is able to accept all types of access networksi.e. WiMAX, CDMA, GSM in order to deliver a widerange a multimedia services which can be offered to theuser by any device connected to the network. The use ofSIP in IMS enable the support of IP-to-IP sessions overany wire-line connection system like DSL as well aswireless networks like Wi-Fi, GSM and CDMA. The IMSallows the inter-working between the traditional TDMnetworks and the IP networks.A. Access Layer
The IMS architecture is independent of any accessbearer. In the mobile networks the access layer could beany or a combination of the following: General PacketRadio Service (GPRS), Enhance Data Rate for GSMEvolution (EDGE), Code Division Multiple Access(CDMA), Wireless Interoperability for Microwave Access(WiMAX), Universal Mobile Telecommunication System(UMTS) and Wireless Local Area Networks (W-LAN orWi-Fi). The fixed line networks makes use of AsymmetricDigital Subscriber Lines (ADSL) and cable networkaccesses.B. Transport Layer
This is an all IP network which consist of IP Routers.These routers are Label Edge Routers and Core SwitchingNetworks. The IP/MPLS (Multi-protocol cable Switching)is the transport layer technology for IMS platform. MPLSdefines a mechanism for the forwarding of packets in arouter network. Due to its flexibility, the MPLS hasbecome the default IP transport network for the NextGeneration Networks making use of the IMS core in orderto reliability and Quality of Service.C. Session Control Layer:
This layer consists of network control servers which areused for the management of calls, establishment andmodifications of sessions in the IMS platform. Two mainelements in this layer are the Call Session ControlFunction (CSCF) and the Home Subscriber Server (HSS).These two elements form the core of the ISM architectureand are sometimes referred to as SIP Servers. The CSCFprovide end-point registration and routing of SIP signaling
messages and provide interworking with the transportlayer for guaranteed QoS of all services. The HSS is thedatabase that is used to store the subscriber service profilesand service triggers. The other information stored by theHSS includes the dynamic data of the subscribes likelocation information.D. Application layer
This layer makes use of application and content serversto provide value-added services. The Application Server(AS), the Multimedia Resource Function Controller(MRFC) and Multimedia Function Resource processor(MFRP) form the core of this layer. The AS is responsiblefor the execution of service-specific logic i.e. userinteraction with subscribers and call flows. The MRFP isalso known as IP media server is used to provide mediaprocessing for the application layer. The media server isused to enable to delivery of some non-telephony serviceslike Push-to-Talk (PTT), speech enabled services, videoservices and other services like conferencing, prepaid andpersonalized call-back tones.
The control and application layers are access andtransport independent and this is very useful in order toensure that the user is able access the ISM servicesrequired from any access network and from anyconnection device.
Figure 2: Layered Architecture of IMS
III. IMS ARCHITECTURE DESIGN
In the General architecture of IMS, 3GPP standardizefunctions but not nodes. So we can say that the IMSarchitecture is a collection of functions linked bystandardized interfaces. If any implementer want they canmerge two functions into single node as well as they candivide single function into two or more nodes. Figure 3shows an overview of the IMS architecture as standardizedby 3GPP.[1,2] Here we include only most importantnodes. The common nodes included in the IMS are asfollows:
A. CSCF (Call/Session Control Function)
International Journal of Electronics Communication and Computer EngineeringVolume 3, Issue 2, ISSN 2249 –071X
CSCF is a SIP (Session Initiation Protocol) server whichprocesses SIP signaling in the IMS. CSCFs aredynamically associated, service-independent andstandardized access points. It distributes incoming calls tothe application services and handles initial subscriberauthentication. There are three types of CSCFs dependingon the functionality they provide.
P-CSCF (Proxy-CSCF): The P-CSCF is the first pointof contact between the IMS terminal and the IMS network.All the requests initiated by the IMS terminal or destinedto the IMS terminal traverse the P-CSCF. This nodeprovides several functions related to security. The P-CSCFalso generates charging information toward a chargingcollection node. An IMS usually includes a number of P-CSCFs for the sake of scalability and redundancy. Each P-CSCF serves a number of IMS terminals, depending onthe capacity of the node.
I-CSCF (Interrogating-CSCF): The I-CSCF providesthe functionality of a SIP proxy server. It also has aninterface to the SLF (Subscriber Location Function) andHSS (Home Subscriber Server). This interface is based onthe Diameter protocol (RFC 3588). I-CSCF retrieves userlocation information and routes the SIP request to theappropriate destination, typically an S-CSCF.
S-CSCF (Serving-CSCF): The S-CSCF is a SIP serverthat performs session control. It maintains a bindingbetween the user location and the user’s SIP address ofrecord (also known as Public User Identity). Like the I-CSCF, the S-CSCF also implements a Diameter interfaceto the HSS.B. SIP AS (Application Server)
The AS is a SIP entity that hosts and executes IPMultimedia Services based on SIP.C. MGCF (Media Gateway Control Function)
MGCF implements a state machine that does protocolconversion and maps SIP to either ISUP (ISDN User part)over IP or BICC (Bearer Independent Call Control) overIP. The protocol used between the MGCF and the MGW isH.248 (ITU-T Recommendation H.248).D. MGW (Media Gateway)
The MGW interfaces the media plane of the PSTN(Public Switched Telephone Network) or CS (CircuitSwitched) network. On one side the MGW is able to sendand receive IMS media over the Real-Time Protocol(RTP). On the other side the MGW uses one or more PCM(Pulse Code Modulation) time slots to connect to the CSnetwork. Additionally, the MGW performs trans-codingwhen the IMS terminal does not support the codec used bythe CS side.E. HSS (Home Subscriber Server)
It contains all the user related subscription data requiredto handle multimedia sessions. These data include, amongother items, location information, security information(including both authentication and authorizationinformation), user profile information and the S-CSCFallocated to the user. The SLF (Subscription LocationFunction) is a simple database that maps users’ addressesto HSSs. Both the HSS and the SLF implement theDiameter protocol.
Figure 3: IMS Architecture
IV. SIGNALING
A main question in integration of circuit switchednetwork and packet switched network is of signaling. Boththe networks use different sort of signaling, which requiresto be mapped into the form, which used by other networkwhen necessary to navigate the links of that network.[4,6]Following table shows the signals used by both type ofnetworks and their equivalent at either sides:
The protocol stack shows layer by layer usage ofdifferent protocols. At Physical Layer SONET, ATM,V.34 etc are used. Likewise at Network Layer IPv4 &IPv6(not currently) are being used. Transport Layer usesTCP and UDP. Finally at Application Layer for signalingSIP, SDP, MGCP, H.323, RTSP are used. Finally forQuality of Service RSVP and RTCP are used and forvideo transmission RTP is used.
Figure 3: Protocol Stack
In practice, there are two main protocols used forinitiating multimedia sessions viz. H.323 and SIP.
A. H.323
International Journal of Electronics Communication and Computer EngineeringVolume 3, Issue 2, ISSN 2249 –071X
The first one is H.323, specified by ITU-T for theimplementation of multimedia services. It is not a singlestandard but is an umbrella of standards. This protocol wasoriginally created to provide a mechanism for transportingmultimedia applications over LANs but it has rapidlyevolved to address the growing needs of IP networks. Onestrength of H.323 was the relatively early availability of aset of standards, not only defining the basic call model, butin addition the supplementary services, needed to addressbusiness communication expectations. H.323 was the firstVoIP standard to adopt the IETF standard RTP to transportaudio and video over IP networks.B. SIP
Session Initiation Protocol (SIP) [14, 16] is anapplication-layer signaling protocol that can establish,modify, and terminate multimedia sessions (conferences)such as Internet telephony calls. SIP can also inviteparticipants to already existing sessions, such as multi-castconferences. Media can be added to (and removed from)an existing session. SIP transparently supports namemapping and redirection services, which supports personalmobility - users can maintain a single externally visibleidentifier regardless of their network location. SIPsupports five facets of establishing and terminatingmultimedia communications:
• User location: Determination of the end systemto be used for communication.
• User availability: determination of thewillingness of the called party to engage incommunications.
• User capabilities: Determination of the mediaand media parameters to be used.
• Session setup: “Ringing”, establishment ofsession parameters at both called and callingparty.
• Session management: Including transfer andtermination of sessions, modifying sessionparameters, and invoking services.
V. SYSTEM DESCRIPTION
Several works analyzed the subject of SIP signalingfrom different angles. All these works did not associateany constraints to the access network and assumed aninfinite bandwidth over the link between the User Agents,IMS entities and intermediate routers. This assumption isnot true and the impact of SIP signaling on the networkand its consequences on the SIP nodes must be evaluated.Apart from this, most of the studies were carried out withtopology containing either fixed nodes or mobile nodesonly; therefore the performance of a hybrid topology canbe studied too.
So keeping in mind the drawback of the previous works,we propose a topology consisting of both mobile and fixednodes with proper specification of suitable links betweenthem. Statistic to be obtained1. Tunneled Traffic Sent/Received: Amount of traffictunneled by Agents and de-tunneled by agents and hosts. Itis given in packets/sec.
2. Jitter: If two consecutive packets leave the source nodewith time stamps t1 & t2 and are played back at thedestination node at time t3 & t4, then:Jitter = (t4 - t3) - (t2 - t1)Negative jitter indicates that the time difference betweenthe packets at the destination node was less than that at thesource node. It is given in sec.3. Packet End to End Delay: The total voice packet delay,called "mouth-to-ear" delay = network delay + encodingdelay + decoding delay + compression delay +decompression delay. Specified in seconds.4. Wireless LAN Delay: Represents the end to end delayof all the packets received by the wireless LAN MACs ofall WLAN nodes in the network and forwarded to thehigher layer. Specified in seconds.5. Retransmission Attempts: Total number ofretransmission attempts by all WLAN MACs in thenetwork until either packet is successfully transmitted or itis discarded as a result of reaching short or long retrylimit. Specified in packets.6. Registration Traffic Sent/Received: Amount of MobileIP registration packets sent/received. Mobile IP nodes androuter register their addresses with agent nodes(home/foreign) to receive mobile ip service.7. Throughput: This statistic represents the averagenumber of packets successfully received or transmitted bythe receiver or transmitter channel per second.8. Utilization: This statistic represents the percentage ofoccupancy of an available channel bandwidth with respectto time period, where a value of 100.0 would indicate fullusage.9. SIP Active Calls: Number of active calls at any giventime.10. Call Setup Time: Time to setup a call in seconds.11. Calls Connected: It includes calls initiated by this nodeand also the incoming call requests.12. Calls Initiated: Number of calls initiated at a particularSIP UAC node.13. Call Duration: Duration of each call defined as thetime at which the node got call connect confirmation to thetime at which it got call disconnect confirmation.
VI. CONCLUSION
Next Generation Networks (NGNs) aims at providing awide range of services to end-users over an accessindependent platform while allowing for better Quality ofservice (QoS), charging mechanism and integration ofservices as compared to conventional fixed or mobilenetworks. The NGN core network is known as IPMultimedia Subsystem (IMS) which is the ThirdGeneration Partnership Project (3GPP) standardized corenetwork for all IP-convergence of fixed and mobilenetworks. The core functionality of the IMS is built on theSession Initiation Protocol (SIP), the Internet EngineeringTask Force (IETF) standardized protocol for the creation,management and termination of multimedia sessions onthe internet. Hence to study the signalling of SIP traffic isthe subject of major interest.
International Journal of Electronics Communication and Computer EngineeringVolume 3, Issue 2, ISSN 2249 –071X
In this work we have to analyze the impact of differenttypes of traffic load with varying pattern of the SIPsignalling carried over the network. We need to configurea hybrid network topology consisting of IMS entities I-CSCF, P-CSCF, S-CSCF, intermediate routers and SIPenabled fixed and mobile nodes. Then we have to createdifferent scenarios and varied their respective parameters.Other signalling protocol like H.323, etc, do not supportmobility but SIP handles this problem because of theMobile IP support. The IMS is based on Session InitiationProtocol (SIP) which is a text based protocol. The IMSwill generally create additional signaling traffic in the IPbased networks, so there is a need to take necessaryprecautions to minimize the signaling overload.
[2] 3GPP, Technical Specification Group Services and SystemAspects, IP Multimedia Subsystem (IMS) - Stage 2 (Release, TS23.228 v5.6.0, 2002-09.
[3] 3GPP, Technical Specification Group Services and SystemAspects: QoS Concept and Architecture (Release 5), TS 23.207v5.6.0, 2002-09.
[4] 3GPP, Technical Specification Group Core Network; Signallingflows for the IP multimedia call control based on SIP and SDP;Stage 3 (Release 5), TS 24.228 v5.2.0, 2002-09.
[5] C. Perkins, "IP mobility support for IPv4," RFC 3344, InternetEngineering Task Force,2002.
[6] V. Planat, N. Kara; “SIP Signaling retransmission analysis over3G network”, Proceedings of MoMM 2006
[7] Jie Xiao, Changcheng Huang and James Yan; “A Flow- basedTraffic Model for SIP Messages in IMS”, 978-1-4244-4148-8/09IEEE "GLOBECOM" 2009 proceedings.
[8] Arslan Munir, Gordon Ross, “SIP-Based IMS SignalingAnalysis for WiMax-3G Interworking Architectures”, 1536-1233/10 IEEE Transactions on mobile computing, Vol. 9, No. 5,May 2010.
[9] 3GPP. TS 24 229 “IP Multimedia Call Control Protocol based onSession Initiation Protocol (SIP) and Session DescriptionProtocol (SDP)” V5.16.0, March 2006.
[10] Session Initiation Protocol (SIP), RFC 3261, H. Schulzrinne, J.Rosenberg. June 2002
[11] GPP. TS 24 228 Signalling flows for the IP multimedia callcontrol based on Session Initiation Protocol (SIP) and SessionDescription Protocol (SDP) V5.14, December 2005.
AUTHOR’S PROFILE
Ashwini Patilpursuing M.Tech from Information Technology Department at BharatiVidyapeeth Deemed University College of Engineering, Dhankawadi,Pune India. Her areas of interest are Software Engineering and networks.
H. K. Sawantworking as an Professor in Information Technology Department atBharati Vidyapeeth Deemed University College of Engineering,Dhankawadi, Pune India. He was awarded his Master of TechnologyDegree from IIT Mumbai. He is pursuing his PhD from JJTU. His areasof interest are Computer Network, Software Engineering and MultimediaSystem. He has nineteen years experience in teaching and research. Hehas published more than twenty research papers in journals andconferences. He has also guided ten postgraduate students.