Performance evaluation of SIP-based multimedia services in UMTS

doi:10.1016/j.comnet.2005.05.013

Computer Networks

Volume 49, Issue 3, 19 October 2005, Pages 385-403

https://doi.org/10.1016/j.comnet.2005.05.013 Get rights and content

Abstract

With an ever increasing penetration of IP technologies and the tremendous growth in wireless data traffic, the wireless industry is evolving the mobile core networks towards IP technology. The third Generation Partnership Project (3GPP) has specified an IP multimedia sub-system (IMS) in UMTS Release 5/6, which is adjunct to the UMTS packet-switched (PS) GPRS core network. This IP-based network provides full packet call control capabilities by using the text-based Session Initiation Protocol (SIP). Initial indications, as to the signalling delay associated with SIP messages, have concerned mobile operators about the viability of SIP services over the UMTS air interface. This article provides an insight into the UMTS system performance, focusing on selected UMTS SIP-based services. Typical services with real-time requirements such as voice as well as delay-sensitive and non-sensitive applications, such as real-time chat and instant messaging services are investigated. Furthermore, the paper discusses and analyses the requirements and possible solutions for improving efficiency of SIP usage in a wireless environment through signalling protocol message compression. Results of a performance evaluation of SIP signalling scenarios are presented in terms of time delay and message overload in the system. Results show that message compression can considerably reduce SIP message transmission time on the radio access network while core network delay contributions are found to be still high.

Introduction

Second-generation wireless systems, such as the global system for mobile communications (GSM), were primary designed to provide mobile voice services to the end user with additional short messages and low-rate (9.6 kbps) data services in a second phase. However, with a growing demand for mobile data access and the explosive growth of Internet data services, wireless data applications are seen as a major new revenue stream, which has spawned much development initially towards 2.5G services such as GPRS and now focused on third generation mobile networks, i.e., the universal mobile telecommunication system (UMTS).

UMTS evolves the mobile core network towards an all IP technology with a new radio network that provides higher capacity and data rates required for the support of advanced multimedia services. UMTS introduces a new radio access network based on WCDMA and evolves the core network towards a novel packet-switched IP-based transport and service platform, the IP multimedia core network sub-system (IMS), which is specified in UMTS Release 5/6 [1]. Through this all the associated features of a circuit-switched call become available to the packet-switched user in the IP domain.

The Release 5/6 standards employ the IETF defined text-based Session Initiation Protocol (SIP) [2] as their call control protocol. Text-based protocols have the advantage that they are generally easier to develop than bit-wise presentations, but their messages are long. SIP message overheads are very large and therefore potentially inefficient for a limited bandwidth wireless system such as UMTS. This has motivated the investigation of SIP signalling performance and possibilities to improve SIP-based signalling [3]. The aim of the investigation presented here is the development of a UMTS IMS sub-system model designed and implemented in a computer simulation environment, compatible with the proposed 3GPP standards. The developed model allows the evaluation of different SIP-based services and their capabilities within UMTS, provides for an understanding of the implication of introducing SIP-based services in a 3G wireless networks, and yields an insight into the UMTS system performance. Previous contributions [4], [5], [6] have presented SIP signalling, described SIP services in 3G networks and have pointed at possible performance difficulties due to the large SIP message sizes. The analysis presented here goes beyond prior work in that it contributes a new insight into the expected performance of SIP services in UMTS beyond previous articles and shows how SIP protocol message compression has the potential to significantly improve delay issues for SIP signalling in UMTS.

In the following, we briefly discuss the UMTS network architecture, then present a brief overview of the session initiation protocol and describe the SIP services we modelled: Voice over IP and instant messaging with presence capability. Then we present a physical reference implementation of the IMS UMTS architecture and present the results of a performance evaluation of selected SIP multimedia services based on the UMTS model.

Section snippets

UMTS Release 5/6 IMS

The Universal Mobile Telecommunications System (UMTS) is a third generation mobile communication system that provides a range of broadband services. UMTS Release 99 includes two distinct core networks for support of packet-switched (PS) and circuit-switched (CS) services as depicted in Fig. 1. Call control for CS services is similar to call control in GSM. The UMTS Terrestrial Radio Access Network (UTRAN) consist of the NodeB and the RNC, where the NodeB operates as the base station and the RNC

SIP-based applications

The Session Initiation Protocol (SIP) is an application-layer control protocol developed by the IETF and defined in [2]. This text-encoded protocol was initially designed to establish, modify and terminate multimedia sessions with one or more participants in an IP environment. However, SIP’s mobility support and extensibility make it ideal to support other services and applications, such as Presence and Instant Messaging. SIP messages use the Session Description Protocol [10] to request certain

SIP in UMTS

The Session Initiation Protocol has been standardised in UMTS Release 5/6 as the application layer signalling protocol for packet-switched (PS) sessions between the UE and the IMS. The new network elements introduced in the IMS act as SIP proxies/servers routing and handling SIP messages through the network. This allows the wireless network to provide a simple voice telephone call to a PS user as well as a wide range of IP-based services such as instant messaging with presence.

SIP messages in

Simulation model and environment

In order to evaluate the performance of SIP-based multimedia services in UMTS Release 5/6 we have developed an event driven simulation environment that allows the implementation and simulation of SIP message flows within a model of a UMTS network. In order to facilitate the efficient simulation of a full UMTS network including a large number of radio access network and core network elements as well as IMS network elements, the simulator implements network elements as consisting of processors

Performance characteristics

In order to obtain performance characteristics for SIP-based multimedia sessions in UMTS Release 5/6 we have simulated packet-voice call setup delays for mobile originated and terminated call initiation and for the two instant messaging schemes using the models presented above. First we present typical SIP message sizes on which the signalling flows for multimedia services are based and then we present results for session setup delays for voice and instant messaging sessions.

Conclusions

We have presented an overview of SIP-based multimedia services proposed for implementation in UMTS Release 5/6 using the new IP-based Multimedia Core Network Subsystem. Part of the overview was to show SIP messages flows associated with the new service propositions. In order to efficiently implement these services, an SIP-based packet voice service (voice over IP) and a presence and instant messaging service, we also proposed physical reference implementations of the IMS in order to limit

References (22)

H. Kaaranen et al.
UMTS Networks
(2001)
J. Rosenberg et al., SIP: Session Initiation Protocol, IETF RFC 3261, June...
M.I. Pous, G. Foster, A. Sesmun, V. Kenneally, D. Pesch, Performance evaluation of UMTS packet voice call control, in:...
I.D.D. Curcio, M. Lundan, SIP call setup delay in 3G, in: Proceedings of the IEEE ISCC 2002, Taormina, Italy, July...
C. Urrutia-Valdés et al.
SIP in 3G wireless networks: Service models, architecture, and network design
Bell Labs Technical Journal
(2004)
M. Cortes et al.
On SIP performance
Bell Labs Technical Journal
(2004)
3GPP TS 23.002, Network Architecture,...
3GPP TR 23.922, Architecture for an All IP network,...
3GPP TS 23.228, IP Multimedia Subsystem (IMS); Stage 2,...
M. Handley, V. Jacobson, SDP: Session Description Protocol, IETF RFC 2327, April...

3GPP TS 24.228, Signalling flows for the IP multimedia call control based on SIP and SDP; Stage 3,...

Cited by (31)

Control mechanisms of presence updates: A tradeoff between traffic optimization and information consistency
2012, Computer Communications
Citation Excerpt :
The IMS, which is the foundation of NGNs, evolves mobile operators towards an all-IP technology for the support of advanced multimedia services. The performance analysis in [5] shows that SIP signaling traffic introduces long transmission delays on the UMTS network. Much of this delay is due to excessive traffic on the network core, and hence optimizing this traffic is necessary to provide multimedia services in real time.
Presence information, which consists of a wide variety of user context, is seen as a key enabler of user personalization in multimedia applications. Presence protocols tend to generate much traffic due to timely dissemination of presence changes and presence subscription maintenance. On one hand, controlling the rate of presence publications saves presence traffic on the access network link. However, low publication rates may introduce excessive delays in presence applications, which usually have real-time constraints. We therefore propose a strategy for adjusting publication rates to the frequency of presence changes based on the Markov theory, which are referred to as sojourn-based rates. On the other hand, presence overload is specially challenging in large-scale presence federation scenarios, in which there are millions of subscriptions between the federated domains. We propose a queuing system that controls the rate of both presence notifications and presence publications in these scenarios. We also propose an adaptive control mechanism that dynamically changes the queuing system’s parameters in order to satisfy the federated domains’ Quality of Service (QoS) needs on traffic optimization and information consistency. This mechanism can be combined with sojourn-based rates in order to improve its performance. We provide a mathematical model of both proposals and analytically estimate their performance.
A token-bucket based notification traffic control mechanism for IMS presence service
2011, Computer Communications
Citation Excerpt :
This does not decrease the number of NOTIFY messages, but significantly reduces the volume of NOTIFY messages. Compressing SIP messages is another technique to reduce the volume of messages sent in the network [18,19]. Usually these compression algorithms substitute words with letters in SIP messages.
Presence is a service that allows a user to be informed about the specified state of another user. Presence service has become a key enabler for next-generation applications such as instant messaging, push-to-talk and web2.0. However, recent studies show that the notification traffic of presence service causes heavy signaling load on IP multimedia subsystem (IMS) network. This paper introduced a token-bucket based notification traffic control (TNTC) mechanism, which is an application layer solution deployed at the presence server. The TNTC aims at upgrading valid access probability while controlling the notification traffic. A mathematical model of a queuing system is proposed to describe TNTC. We analyzed its main probability features and investigated the effects of different parameters on the performance of TNTC. Extensive simulations verified that TNTC can effectively control notification traffic and perform better than the existing schemes in terms of valid access probability and update arrival rate.
Evaluating extensions to IMS session setup for multicast-based many-to-many services
2011, Computer Networks
Citation Excerpt :
In this section, the delays related to the session and user plane setup will be evaluated. Session setup delay in IMS networks is a relevant parameter that has been studied in several scenarios (see [24–26]). The evaluation has been structured in two parts.
Telecommunication networks are converging towards an all-IP paradigm that integrates a broad set of value-added services. In this context, the IP Multimedia Subsystem (IMS) is being developed by the 3GPP as a key element to achieve the convergence. Additionally, multiparty services are nowadays acquiring an increasing interest from the industry. In this respect, network multicast provides a cost-effective solution to deliver these services to the user. Nevertheless, although network multicast is being considered as an enabler for one-to-many services (e.g. IPTV) in the IMS, the specifications for many-to-many services still follow a unicast approach (e.g. push-to-talk and conference). This paper describes extensions to the session control procedures in the IMS, to support multicast based multi-user services. The idea was first described in a prior work, but this paper presents enhancements to provide a comprehensive solution and to improve the grade of service (GOS) perceived by the users. In addition, the GOS achieved by the proposal is evaluated. First, the bandwidth utilization for the multicast-based multi-user services is analyzed and compared against the unicast scenario. Next, the GOS is evaluated using an analytical approach, by obtaining the mathematical expressions for the session and user plane setup delays. Finally, the GOS is also evaluated using an experimental approach, and the results are compared with values recommended by the ITU-T.
Supporting mobility in an IMS-based P2P IPTV service: A proactive context transfer mechanism
2010, Computer Communications
Citation Excerpt :
The final results are presented in two ways: a general equation and an approximate value, in order to compare all mobility mechanisms. Some numerical input values for our analysis were extracted from [31], and are summarized in Table 1. These values are defined as follows:
In recent years, IPTV has received an increasing amount of interest from the industry, commercial providers and the research community, alike. In this context, standardization bodies, such as ETSI and ITU-T, are specifying the architecture of IPTV systems based on IP multicast. An interesting alternative to support the IPTV service delivery relies on the Peer-to-Peer (P2P) paradigm to distribute and push the streaming effort towards the network edge. However, while P2P IPTV was studied in fixed access technologies, there has been little attention paid to the implications arising in mobile environments. One of these involves the service handover when the user moves to a different network. By analyzing previous work from the perspective of an IPTV service, we concluded that a proactive approach is necessary for the handling of inter-network handovers. In this paper, we propose a new general handover mechanism for the IP Multimedia Subsystem (IMS), while studying its applicability to a P2P IPTV service. Our solution, called proactive context transfer service, incorporates the existing IEEE 802.21 technology in order to minimize the handover delay. The proposal is validated by comparing it against solutions derived from previous work.
An analytical evaluation of mobility management in integrated WLAN-UMTS networks
2010, Computers and Electrical Engineering
Ensuring uninterrupted service continuity for handoff calls in an all-IP inter-networked heterogeneous environment requires successful session management among participating access networks. As such, a mobility-aware novel interworking architecture is presented in this article that facilitates session management including session establishment and seamless session handoff across different networks. This framework conveniently enables any 3G cellular technology such as the Universal Mobile Telecommunications System (UMTS) to interwork with a given Broadband Wireless Access (BWA) technology such as the Worldwide Interoperability for Microwave Access (WiMAX) network or the Wireless Local Area Network (WLAN) under a common signaling platform. This framework exploits the IP Multimedia Subsystem (IMS) as a universal coupling mediator for real-time session negotiation and management. Next, a Queuing Theory based analytical model for evaluating the performance of vertical handoff management between these interworked 3rd Generation (3G) cellular networks and the WLANs is presented. The analysis includes vertical handoff performance measures such as delay, transient packet loss, jitter, and signaling overhead/cost. The latter part of this paper presents some results from OPNET based simulations for the verification of the analytical model and results.
Quality of protection analysis and performance modeling in IP multimedia subsystem
2009, Computer Communications
Citation Excerpt :
SIP signaling delay includes server response time and network transmission delay directly which impacts on User QoS during session control. Dirk Pescha studied performance of IMS core network in [20], they designed signal simulator SigSim to predict end-to-end signal workload and delay which shows message compression could reduce transmission time during RAN while delay in core network still very high. Fathi evaluated SIP session setup delay in IMS with various underlying protocols [21].
IP Multimedia Subsystem (IMS) is widely considered as the main solution for the next generation multimedia rich communication. In order to provide multi-level security service in IMS to mobile users for multimedia applications, it is insufficient to take the security benefits into consideration, but adequately analyzing the impact of security policies in IMS on the performance cost quantitatively is also necessary and significant. In this paper, we first propose a novel study of IMS performance by a QPN model to much more precisely describe SIP signaling in IMS, and made performance evaluation. Then we defined totally seven levels of IMS security policies according to IMS specifications in 3GPP and proposed QoP partition model which quantitatively reflects strength of protection of SIP signaling and users’ security needs. Further more, we did our original contributions to use the QPN model to evaluate the impacts of security mechanism on system performance cost quantitatively. With the multi-view security partition introduced, different security policies could be adjusted according to the application and users’ security requirements, so that multi-level security service can be provided to diverse users and applications for a better tradeoff between security requirements and system performance in IMS.

View all citing articles on Scopus

Dirk Pesch was born in Krefeld, Germany in 1966. He received a Dipl. Ing degree from Aachen University of Technology in 1993 and a Ph.D. degree from the University of Strathclyde, Glasgow, UK, in 1999, both in Electrical and Electronic Engineering. He was with Nokia Mobile Phones from 1993 to 1995. From 1996 to 1998 he was a research fellow with the Mobile Communications Group in the Department of Electrical and Electronic Engineering at University of Strathclyde. Since 1999 he has been a lecturer in the Department of Electronic Engineering at Cork Institute of Technology. He is co-founder and research director of the Centre for Adaptive Wireless Systems at Cork Institute of Technology. He is a regular reviewer for a number of international journals and has been on the programme committee of a number of international conference including IEEE Globecom, IEEE ISWCS, IFIP MWCN, European Wireless, and others. He has published over 45 papers in peer-reviewed journals and conferences. His research interests are in the area of design and evaluation of algorithms and protocols for mobile and wireless networks, sensor and ad-hoc networks, and performance evaluation of multimedia services in mobile networks.

Maria Isabel Pous received an engineering degree from the University of Valencia, Spain in 2001 and a Master of Engineering degree from Cork Institute of Technology in 2003, both in Electronic Engineering. Since late 2003 she has been with a consultant with Accenture in Madrid. Her research interests are in the area of performance evaluation of multimedia protocols in mobile networks.

Gerry Foster graduated in communications engineering at Plymouth Polytechnic in 1986. He worked at British Aerospace until 1989 designing components for microwave and millimetre wave communications systems. He then moved to STC for two years, where he developed microwave components for the Squarial satellite flat-plate antenna and worked on the communications link for a reconnaissance aircraft. Next, he moved into datacom for British Gas, where he worked on PMR modem design, scanning telemetry, underwater, and pipe-borne communications systems. In 1997 he moved to Lucent to work on GPRS GSNs, and for the last five years he has been at Motorola working on UMTS systems analysis.

View full text