Conference Paper

SIP call setup delay in 3G networks

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Abstract

In this paper, call setup time in Session Initiation Protocol (SIP)-based videotelephony is analyzed. We used a 3G network emulator to measure post-dialing delay, answer-signal delay and call-release delay. The results are compared to local, national, international and overseas intranet LAN calls. Furthermore, we have also studied the effect SIP calls over lossy channels with restricted bandwidth, typical of mobile network signaling bearers.

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... Therefore, delays could fluctuate and it might not be possible to identify the origin of such discrepancies. However, research in [4][5] provide simulations for similar scenarios in Internet Telephony systems. ...
... Since all of the network elements were located in a private network, the environment could be though of as providing local calls. Our results also show that an embedded mobile VoIP client experiences an increased delay compared to a PC client, such as the one measured in [4] with a WCDMA network. E.721 [11] recommends values call setup delays for circuit switched calls. ...
... reflects the time it takes from the moment the receiving end accepts the call until the call is actually established. E.721 recommendation is 0.75s for local, 1.5s for toll, and 2.0s for international connections, with 1.5s, 3.0s, and 5.0s as 95% values. Finally, " call end " (call-release delay) is the time it takes for the call to be terminated. [4][5] VoIP calls setup delays in a cellular system might experience additional delays in cases were there is no active PDP context, and also due to a required registration to the IMS. The PDP context activation delay was ~3 seconds in our tests. Simulations in [18] propose 2.24 seconds. Based on these values, calls with always-on enabled c ...
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Wireless broadband enables the use of VoIP with handheld mobile deices. Likewise, multi radio terminals provide the user with ubiquitous access. Even though broadband access might be available in different locations, performance still depends on the wireless technology used. We study the main signaling delays that take place in a voice call (registration and voice call setup delays) and compare them for different wireless accesses: HSDPA, WCDMA and WiFi. Likewise, possible optimizations with always-on mode and their drawbacks are presented. Signaling delays and total battery lifetime affect the perceived end user experience directly, and thus are important items to consider in addition to the overall voice quality.
... SIP-based videotelephony is one of the most challenging Multimedia over IP applications. A test in [2] has been carried out in a 3G network emulator, to measure post-dialing delay, answersignal delay and call-release delay. These delays happen during the lifetime of a SIP call. ...
... This is the time elapsed between when the caller clicks the button of the terminal to call another caller and the time the caller hears his terminal ringing. Tests incase of bandwidth limitations has been done [2]. The call success rate was always 100%, thus bandwidth would not stand in the way for a successful SIP call. ...
... In realistic 3G network configurations, the bearer allocated for signaling could be a few kbps. [2] ASD values were constantly 45ms for channels of at least 5kpbs, but ASD increased for very narrow channels (2kpbs to 166 ms). We think this due to the fact that loss is much higher for narrow bandwidth, whereby a 200/OK must be retransmitted. ...
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Developers of SIP mostly concentrated on making SIP to do what no other protocol can do. In a way they failed to consider the weaknesses and dangers. From this realization we did a study on the weaknesses of SIP and SIP signaling in Wireless Networks and Services. From our study we identified Delay and Security as the most severe problems, which we discuss in this paper.
... E.721 recommendation is 0.75s for local, 1.5s for toll, and 2.0s for international connections, with 1.5s, 3.0s, and 5.0s as 95% values. Finally, " call end " (call-release delay) is the time it takes for the call to be terminated [18,19]. The signaling [16] and delay measurements for voice call setups when a PDP context is active and the terminal is registered to the IMS system are depicted inFigure 14. ...
... Since all of the network elements were located in a private network, the environment could be though of as providing local calls. Our results also show that an embedded mobile VoIP client experiences an increased delay compared to a PC client, such as the one measured by Curcio and Lunden [18] with a WCDMA network. The setup delays for VoIP calls might be impacted with additional delays in a cellular system in cases were there is no active PDP context, and also due to a required registration to the IMS. ...
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Third generation (3G) packet switched WCDMA networks with high-speed downlink packet access (HSPDA) are currently being deployed worldwide to provide wireless broadband connectivity. When introducing HSDPA in 3G networks the end user experience and system capacity with voice over IP applications improve considerably. When later on adding also high-speed packet uplink access (HSUPA), the system capacity and end user experience will improve even further. This paper analyzes with measurements the VoIP quality over current Release 5 HSDPA networks. VoIP is expected to be a widely used application over 3G data services. The results show that even though the introduction of HSDPA significantly reduces the user-to-user voice delay, the performance is satisfactory only for selected devices. Overall, the end user experience is still significantly worse than with circuit switched solutions and is not acceptable. The current limitations with VoIP in HSDPA networks with a too large delay can be improved by using the RLC UNACK mode, potentially decreasing the jitter buffer size and reducing the terminal processing delay. In the longer term, HSUPA and several features in 3GPP Release 7 standards will bring further performance improvements in both user plan latency and system capacity.
... For instance ITU-T H.323 is the alternative protocol. However SIP has recently gained the increasing interest of universities standardization organizations and companies" [7]. ...
... "A SIP call setup is essentially a 3-way handshake between caller and callee. For instance, the main legs are INVITE (to initiate a call), 200/OK (to communicate a definitive successful response) and ACK (to acknowledge the response)" [7]. SIP communication messages between caller and callee are shown in Figure 4. ...
... E.721 recommendation is 0.75s for local, 1.5s for toll, and 2.0s for international connections, with 1.5s, 3.0s, and 5.0s as 95% values. Finally, "call end" (call-release delay) is the time it takes for the call to be terminated [18,19]. The signaling [16] and delay measurements for voice call setups when a PDP context is active and the terminal is registered to the IMS system are depicted in Figure 14. ...
... Since all of the network elements were located in a private network, the environment could be though of as providing local calls. Our results also show that an embedded mobile VoIP client experiences an increased delay compared to a PC client, such as the one measured by Curcio and Lunden [18] with a WCDMA network. The setup delays for VoIP calls might be impacted with additional delays in a cellular system in cases were there is no active PDP context, and also due to a required registration to the IMS. ...
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Third generation (3G) packet switched WCDMA networks with high-speed downlink packet access (HSPDA) are currently being deployed worldwide to provide wireless broadband connectivity. When introducing HSDPA in 3G networks the end user experience and system capacity with voice over IP applications impro ve considerably. When later on adding also high-speed packet uplink access (HSUPA), the system capacity and end user experience will improve even further. This paper analyzes with measurements the VoIP quality over current Release 5 HSDPA networks. VoIP is expected to be a widely used application over 3G data servic es. The results show that even though the introduction of HSDPA significantly reduces the user-to-user voi ce delay, the performance is satisfactory only for sel ected devices. Overall, the end user experience is still significantly worse than with circuit switched solu tions and is not acceptable. The current limitation s with VoIP in HSDPA networks with a too large delay can be improved by using the RLC UNACK mode, potentially decreasing the jitter buffer size and r educing the terminal processing delay. In the longe r term, HSUPA and several features in 3GPP Release 7 standards will bring further performance improvements in both user plan latency and system capacity.
... In regard to 3GPP based wireless accesses, (Kist & Harris 2002) provides simulations for transfer delays with 3GPP signaling, while (Fathi, Chakraborty & Prasad 2006; Pous et al. 2003) modeled signaling performance. Further, (Curcio & Lundan 2002) provides measurements for a WCDMA setup using laptop clients for local, international and overseas calls. Most of the mentioned research focuses on simulations, and does not consider some end user cases such as calls in wireless environments starting from different states. ...
... The results show that only HSDPA and WiFi are able to provide performance similar to circuit switched calls. Our results also show that the embedded VoIP client experiences an increased delay compared to a PC client, such as the one measured by (Curcio & Lundan 2002) with a WCDMA network. In addition, our results show the cases in which a 3G data connection (PDP context activation) has been previously established prior to registration or call setup. ...
... Experiments about CSD measurements have been carried out also in [37], where the CSD is indicated by Post-Dialing Delay (PDD) and evaluated in a 3G-based environment. In this case, the CSD is in the order of hundred of milliseconds, but, the authors rely on a network simulator, by neglecting all the concurrent effects present in real networks. ...
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... We plan to evaluate and possibly use them in future work. More study on signaling delay can be found in [17,18]. ...
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... In the network topology as shown inFigure 3, Node 1 and Node 5 represent, respectively, a SIP client (UAC) and a SIP server (UAS).Figure 3. Network topology in simulation scenario to simulate a number of clients and servers in each domain and a number of call requests outstanding simultaneously through a single connection between proxies. The security protocol in SIP is responsible for the security channel in hop-by-hop fashion rather than in end-to-end fashion [6]. The simulation does the same. ...
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... SIP-based video-telephony is one of the most challenging multimedia over IP applications. A test provided by Cursio and Lundan [34] These delays are shown inTable 3 below: ...
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... Curcio et al.[21]used a SIP emulator to measure the delay introduced by SIP transaction over lossy wireless LAN links. They focus on telephony applications and measure the delayt average F(t,r,s) in fall-off region[%]retry = 2 retry = 4 retry = 6 retry = 8The delay introduced by SIP transactions has also been analyzed for UMTS networks[22,23,24](analytically and through simulations), for UMTS satellite links[25](simulations), and for the fixed portion of the Internet[26](simulations). The influence of 802.11 packet loss parameters has been empirically studied in two other (i.e., non-SIP) application areas. Aguayo et al.[18]empirically investigated the influence of 802.11 packet loss parameters in the context of multi-hop ad-hoc networks. ...
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The Session Initiation Protocol (SIP) is a popular application-level signaling protocol that is used for a wide variety of applications such as session control and mobility handling. In some of these applications, the exchange of SIP messages is time-critical, for instance when SIP is used to handle mobility for voice over IP sessions. SIP may however introduce significant delays when it runs on top of UDP over lossy (wireless) links. These delays are the result of the exponential back-off retransmission scheme that SIP uses to recover from packet loss, which has a default back-off time of half a second.In this paper, we empirically investigate the delay introduced by SIP when it runs on top of UDP over IEEE 802.11b links. We focus on the operation of SIP at the edge of an 802.11b cell (e.g., to update a mobile host's IP address after a handoff) as this is where SIP's retransmissions scheme is most likely to come into play. We experiment with a few 802.11 parameters that influence packet loss on the wireless link, specifically with different link-level retransmission thresholds, signal-to-noise-ratios (SNRs), and amounts of background traffic. We conduct these experiments in a controlled environment that is free from interfering 802.11 sources.Our results indicate that (1) SIP usually introduces little delay except for an SNR range of a few dBs at the very edge of an 802.11 cell in which the delay increases sharply, and (2) that a maximum of four 802.11 retransmissions suffices to limit the delay introduced by SIP retransmissions. The first result is of interest to developers of SIP applications who have to decide at which SNR to initiate a handoff to another network. The second result allows network providers to optimize their 802.11b networks for delay sensitive SIP applications.
... The figure is deliberately abbreviated to emphasize the post-dialing period. The duration between the first INVITE message and the 180 Ringing response is defined as post-dialing period, and the time it takes is called post-dialing delay [10] or, most commonly, dial-to-ring delay. It is the time elapsed between the time instant when user finishes dialing and the time instant when caller hears ringing response from the terminal. ...
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... Experiments about CSD measurements have been carried out also in [37], where the CSD is indicated by Post-Dialing Delay (PDD) and evaluated in a 3G-based environment. In this case, the CSD is in the order of hundred of milliseconds, but, the authors rely on a network simulator, by neglecting all the concurrent effects present in real networks. ...
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... A dial-to-ring latency of 300ms is comparable to the setup time needed for an overseas call in a fixed network infrastructure [15]. The overhead is introduced by the slow Bluetooth connection and the various components (proxy, access point) involved. ...
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... On the other hand, Fathi et al. [4] consider a SIP-based VoIP system and evaluate the call setup latency in wireless fading channels. Curcio et al. [5] evaluate the SIP call setup latency using a 3G network emulator. All of these works consider cellular systems where a dedicated channel is assigned to a mobile user. ...
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... Experiments about CSD measurements have been carried out also in [37], where the CSD is indicated by Post-Dialing Delay (PDD) and evaluated in a 3G-based environment. In this case, the CSD is in the order of hundred of milliseconds, but, the authors rely on a network simulator, by neglecting all the concurrent effects present in real networks. ...
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... The delay introduced by SIP transactions has also been analyzed for UMTS networks [Banerjee03,Banerjee04,Curcio02] (analytically and through simulations), for UMTS satellite links [Kueh04] (simulations), and for the fixed portion of the Internet [Eyers00] (simulations). ...
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Behind the popularity of VoIP in these days, it may present significant security challenges in privacy and accounting. Authentication and message encryption are considered to be essential mechanisms in VoIP to be comparable to PSTN. SIP is responsible for setting up a secure call in VoIP. SIP employs TLS, DTLS or IPSec combined with TCP, UDP or SCTP as a security protocol in VoIP. These security mechanisms may introduce additional overheads into the SIP performance. However, this overhead has not been understood in detail by the community. In this paper we present the effect of the security protocol on the performance of SIP by comparing the call setup delays among security protocols. We implement a simulation of the various combinations of three security protocols and three transport layer protocols suggested for SIP. UDP with any combination of security protocols performs a lot better than the combination of TCP. TLS over SCTP may impose higher impact on the performance in average because TLS might have to open secure channels as the same number of streams in SCTP. The reasons for differences in the SIP performances are given.
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The IP Multimedia Subsystem (IMS) is the technology that will merge the Internet (packet switching) with the cellular world (circuit switching). It will make Internet technologies, such as the web, email, instant messaging, presence, and videoconferencing available nearly everywhere. Presence is one of the basic services that is likely to become omnipresent in IMS. It is the service that allows a user to be informed about the reachability, availability, and willingness of communication of another user. Push to talk over Cellular (PoC) is another service in IMS that is intended to provide rapid communications for business and consumer customers of mobile networks. In order to become a truly successful mass-market service for the consumer segment, the only realistic alternative is a standardized Push-to-talk solution providing full interoperability between terminals and operators. Instant Messaging (IM) is the service that allows an IMS user to send some content to another user in near real-time. This service works under IETF’s Message Session Relay protocol (MSRP) to overcome the congestion control problem. We believe the efficiency of these services along with the mobility management in IMS session establishment has not been sufficiently investigated.In this research work, we identify the key issues to improve the existing protocols in IMS for better system behaviour. The work is centred on the three services of IMS: (1) Presence Service, (2) Push-to-Talk over cellular and, (3) Instant Messaging and over the issue of (4) IMS session set up. The existing session establishment scenario of IP Multimedia Subsystem (IMS) suffers from triangular routing for a certain period of time when an end IMS user or terminal is mobile. In this thesis, the performance of three possible session establishment scenarios in a mobile environment is compared by using an analytical model. The model is developed based on the expressions of cost functions, which represents system delay and overhead involved in sessions’ establishment. The other problem areas in optimizing presence service, dimensioning a PoC service and analysing service rates of IM relay extensions in IMS are identified. A presence server becomes overloaded when massive number of IMS terminals joins a network to request presence facility. Performance models are developed in this research to mitigate such load during heavy traffic for the presence service. Queuing analyses for different cases are provided while instant messaging chunks go through two consecutive relay nodes. The specific factors such as blocking probability, stability conditions, optimized subscription lifetime etc. in IMS environment parameters have been investigated. We have also elaborated models to dimension a PoC service for service providers with regards to controlling PoC session access, optimal PoC session timer, path optimization and number of allowable simultaneous PoC sessions for given network grade of service.In a nutshell, the contribution of this dissertation are: (a) a proposed robust scheduler to improve performance of the IMS presence service, (b) several derived models to dimension IMS Push-to-talk over cellular service, (c) a new mechanism to reduce cost for the IMS session set ups in mobile environment and (d) evaluation of message blocking and stability in IMS Instant Messaging (IM) service by applying queuing theories. All of these analyses have resulted in recommendations for the performance enhancements with optimal resource utilization in IMS framework.
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This paper analyzes the session setup delay in the IP multimedia subsystem (IMS) with the CDMA2000 evolution data only rev. A (EV-DO rev. A) standard for wireless transmission. Session setup delay is particularly critical for interactive multimedia applications, such as gaming, push-to-X and voice over IP (VoIP), as it directly translates in user perception of service quality. Keeping signaling delay low, however, is a challenge in IMS due to the text-based nature of the session initiation protocol (SIP) for signaling, and, more significantly, due to the lossy and capacity constrained wireless links. To address this challenge, we analyze the session setup delay end-to-end, by taking into account key system properties across all layers, ranging from radio links to IMS signaling architecture. We present a model for cross-layer performance analysis and simulation, which includes the statistical properties of the EV-DO (rev. A) wireless channel, and also takes into consideration the properties of transport protocols (TCP, UDP) and SIP signaling (message size and compression). By means of analysis and simulations, we study the setup delay performance of a generic, multi-operator IMS communication scenario between two mobile users. We describe how session setup delay can be estimated and reduced in realistic IMS settings and we propose architecture alternatives to the basic IMS scenario. The results derived from this study show that the proposed methods can incrementally lead to a lower setup delay and less sensitivity to the radio transmission quality and frame error rate compared to the base IMS scenario
Chapter
These days we are witnessing a tremendous increase in the availability of personal communication devices that are able to provide ubiquitous connection. As a result, we can also see an increasing demand for mobile services where content is personalized based on user location and context. Mobile multiplayer gaming is today provided as a service and, of course, is not an exception. Unfortunately, differently from other legacy services, location- and context-based gaming strictly requires near-field communication to interact with nearby players in order to create teams and arenas. Since currently adopted technologies suffer from scalability (Bluetooth) or energy (WiFi) constraints, Opportunistic Networks (ONs) have already been addressed as a viable solution to involve potential players which are located in the surrounding area. Nevertheless, it is not yet clear how player experience will be affected by the increased delay and probabilistic message forwarding introduced by an ON. In this chapter the aforementioned phenomenon will be addressed in order to achieve a better understanding of the problem, and, thanks to a case study, guidelines for game designers will be provided to actually deliver a compelling and intriguing experience through an opportunistic game.
Conference Paper
Mobile IP(MIP) and SIP have been proposed to support mobility in the wireless internet working at different layers of the protocol stack. However MIP has some problems such as triangle routing, the need of each host’s home address, the overhead of tunneling and the lack of wide deployment. Thus we proposed a scheme for supporting mobility based on SIP in this research. A novel SIP system to provide a hierarchical registration has been designed according to this scheme. Our SIP system has been established by implementing JAIN technologies which follow next generation network standards to support the mobility of wireless terminal. This system successfully satisfied ITU-T recommandation.
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IMS (IP Multimedia Subsystem) is an architectural frame work which allows the convergence of Wireless, Wire line networks and internet thereby serving the use of a global platform for the delivery of IP multimedia applications. Thus IMS leads in the greater support of Next Generation Networks (NGN). The cause for IMS a high level technology's design and origin is mainly for the 3 rd Generation Partnership Projects (3GPP). The intension of the design and origin of such a subsystem is to enable the service providers to create and deliver value added services to the users on heterogeneous network. IMS is all about sending data (voice, video, files etc) over an instrument using absolutely any kind of networks. The system has been configured to the real IP of 127.0.0.1 using the open source tool of OpenIMSCore. IMS has been configured between the hosts using OPEN IMS CORE test bed on an Open Source platform and initiate the calls between several IMS clients and its registration was evaluated. The traffic parameters of SIP IMS parameter has been verified using Wireshark network analyzer.
Article
Session Initiation Protocol (SIP) has been selected as the official end-to-end signalling protocol for establishing multimedia sessions in the IP-based Universal Mobile Telecommunication Systems (UMTS) network. Since a satellite component has been identified to play an integral role in UMTS, there is a need also to support SIP-based session establishment over Satellite-UMTS (S-UMTS) to achieve end-to-end seamless IP-based terrestrial/satellite network integration. However, the transport of SIP over the satellite radio interface is not efficient due to the request/response nature of SIP and its large message size, coupled with the unreliable wireless links and the larger propagation delay. With little work carried out in this area, this paper presents the research on the efficient solutions to transport SIP session set-up signalling over the S-UMTS radio interface based on the Radio Link Control acknowledged mode (RLC-AM) mechanisms. Two schemes are also proposed to overcome the inefficiency of the current RLC-AM procedure when applied for session set-up over S-UMTS. The proposed enhancements are easy to implement and are fully backward compatible with the existing implementation of RLC-AM in UMTS.
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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.
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The session initiation protocol (SIP) is used as the signaling protocol in the IP multimedia subsystem (IMS) and the signaling is becoming computing intensive comparing to the current telecommunication network. The SIP is a text-based protocol with characteristics of unordered and verbose headers, variable-size message, and case-insensitive keyword. It imposes challenges for an efficient message processing. The property of SIP elements being able to process SIP messages quickly is critical for the performance of IMS networks. This article investigates the performance of SIP message processed in SIP servers, mainly focusing on improving message parsing by introducing a method named selective parsing for SIP message (SP4SIP). By modeling and analyzing a SIP server with a tandem Jackson network, it is concluded that parsing messages is the bottleneck of a SIP server performance, i.e., it is the most processing intensive activity in the system. To validate the approach, it has been implemented in a high-performance SIP server in the authors' lab. The results show that selective parsing for SIP message can indeed reduce processing time.
Conference Paper
Virtualization, in telecommunication, has been given great interest over the 20 past years. Migrating operating system instances between physical hosts is one of the most important features of this technology. It allows network administrators to resolve serious problem such us hardware failure, lack in server capacity, while facilitates the management and the upgrade of their systems. With the strong use of Internet Protocol Multimedia Subsystem (IMS) by most of third Generation (3G) network operators as a common infrastructure, live migration may be useful to resolve the overload problem that affects Call Session Control Function (CSCF) components. Our objective is to propose a High availability solution in order to avoid performance degradation. Thus, we aim to proof that when introducing the migration technique within the IMS architecture, we resolve the overload problem and guarantee Quality of Service (QoS) for IMS client. Based on previous work on high availability using virtualization, we describe, in this paper, our methodology and implementation of an experimental scenario using live migration. Then, we evaluate the registration and the session setup delays. The use of migration reduces considerably the delays by 40% for registration and by 38% for session setup. So, we obtain acceptable values compared to the thresholds.
Conference Paper
The performance of signaling protocols in circuit and packet switching networks is a significant parameter to evaluate the performance of any telecom operator. Today, users demand continuous high data rate, service availability, network reliability, mobility, QoS and minimum service initialization time. IP Multimedia Subsystem (IMS) standardized Session Initiation Protocol (SIP), which categorize as a signaling protocol for management and maintenance of new multimedia IP-based services. Session setup time for the interactive and real-time application has become major QoS parameter to assess the performance of these applications. In the context of this communication application, in this paper, we analyze the performance of session setup time of SIP protocol over IMS network with different access networks (GSM, WLAN and PSTN). We measure and compare mean delay of each SIP message during session setup over these access networks. We also measure the mean delay with respect to distance of nodes in the network. Our results show that SIP message size, processing, transmission, queuing delay, transition from packet to switched network, node mobility and wireless channel have a consequence on the reliability, performance and efficiency of the protocol.
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Abstract-A bidirectional wavelength-division-multiplexing (WDM) transport system base on two reflective semiconductor optical amplifiers (RSOAs) and optoelectronic feedback technique is proposed and experimentally demonstrated. One RSOA is employed as a broadband light source and another RSOA in combination with optoelectronic feedback technique is used as wavelengths reuse and data signal remodulation scheme. System architecture is further simplified by using only one RSOA at the optical node, and system performance is further improved by optoelectronic feedback technique. Impressive performances of bit error rate (BER) (<;10<sup>-9</sup>) were achieved for both 2.5 Gbps down-link and 1.25 Gbps up-link transmissions.
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In 3G, the server platform of IP multimedia subsystem (IMS) offers multimedia service to the multimedia clients through the exchange of authenticated key agreement (AKA) scheme. Since the IMS service is transmitted over public networks, the server and client of IMS shares a common session-key via the execution of AKA scheme to secure the communication service over a public network. Though there are some identical steps involved in the key authentication Scheme of IMS, the communication efficiencies of IMS server and client are still conciliating. Besides, it is having the issues of insecurity and session compatibility dealt with the real time transport protocol (RTP). To resolve the issues of IMS, this paper proposes three novel schemes: (1) The first scheme is multimedia server–client authentication scheme and its purpose is to mitigate the packet congestion on the networks; (2) The second scheme is traffic on background and its purpose is to analyze the signal congestion, bandwidth consumption and network utilization of the system; and (3) The last one is fault detection index and its purpose is to prevent the SIP vulnerabilities, like bye and invite. In addition, we integrate the proposed and existing authentication schemes, such as EPS-AKA, AP-AKA, UMTS-AKA, X-AKA, S-AKA and so on., in the IMS server and client environments to testify the metrics, namely round trip time, call response time, packet generation, flow analysis, RTP session, network utilization, throughput analysis, bandwidth utilization and flooding attack detection realistically.
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With the growing availability of personal communication devices, we are witnessing a tremendous increase in the demand for mobile services based on location and context. Mobile gaming as a service is, of course, no exception. Unfortunately, differently from other services, location- and context-based gaming strictly requires near-field communication to interact with nearby players to create teams and arenas. Since currently adopted technologies suffer from scalability (Bluetooth) or energy (WiFi) constraints, opportunistic networks (ONs) have already been addressed as a viable solution to involve a considerable number of players on a wider area. Nevertheless, it is not yet clear how player experience will be affected by the increased delay and probabilistic message forwarding introduced by an ON. In this paper, we address the aforementioned problems by studying, by means of simulations, the feasibility to deploy a contact-based game on top of the ON provided by the public transportation systems (PTSs) of three cities: Milan (Italy), Edmonton (AB, Canada), and Chicago (IL, USA). Furthermore, to provide playability and scalability considerations, we also study an opportunistic collaborative version of a famous standalone game. The focus on this specific game is functional to foster the use of the PTS itself. The contribution of this paper is twofold. Firstly, we provide simulation results hinting that deployment on a PTS is feasible when targeting users commuting inside the city. Secondly, we provide a number of considerations and guidelines for game designers to actually deliver a compelling and intriguing experience.
Conference Paper
This paper analyzes the signaling efficiency for multimedia call (session) establishment in all-IP wireless networking infrastructures based on the Internet Multimedia Subsystem (IMS) standard framework and CDMA2000 Evolution Data Only Rev. A (EV-DOrA) standard for wireless link transmission. We present a comprehensive, bottom up analysis of signaling delay for setting up multimedia sessions and we evaluate system architecture alternatives to reduce it. Call setup time is particularly critical for interactive applications, such as gaming, push-to-X and Voice over IP (VoIP), where user experience of time that takes to start or join a session directly translates in user perception of service quality. Our analysis takes into account a large set of system design parameters across all layers, ranging from radio link properties to IMS processing and specific characteristics of the Session Initiation Protocol (SIP). For an example signaling flow between two mobile users, we derive an important set of numerical results for call set up time under various IMS design scenarios that consider effects of SIP compression efficiency, choice of transport protocols (TCP, UDP) and radio link quality. Based on the results obtained, we propose and discuss new methods to improve the signaling efficiency and to reduce the call setup time.
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The modeling of Application Triggering Architecture (ATA) in the IP Multimedia Subsystem (IMS) is presented. The session setup delay and system throughput are employed as the measurement to investigate the performance of ATA and the Serving Call Session Control Function (S-CSCF). With theoretical analysis and simulation results, we find that, the number of the ASs (Application Servers), the use of the subsequent Filter Criteria (sFC) and the arrival rate have heavy impact on the session setup delay and the S-CSCF is the major bottleneck in IMS network. The results are useful in constructing IMS network. At last, we propose several possible solutions to reduce the session setup delay and decrease the load of the S-CSCF.
Conference Paper
This chapter presents a comprehensive introduction to the field of wireless systems and their applications. We begin with the fundamental principles of wireless communications, including modulation techniques wireless system topologies, and performance ...
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Next generation network (NGN) is a packet-based converged network to support session and non-session services in QoS-enabled broadband transport network. QoS based resource control must be defined to support differentiated services for various applications in NGN. This paper defined parallel control mechanisms for NGN resource control interfaces to minimize session and resource control delays. We simulated the existing sequential mechanism and our proposed parallel mechanism to measure and analyze control delays and completion ratios.
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Todaypsilas mobile satellite systems are built to support a packet switched all IP communications infrastructure based on IP multimedia subsystem (IMS) leveraging the benefits of solutions already available in terrestrial networks. IMS technology provides a SIP based open platform that enables operators to introduce multimedia applications quickly and in a cost effective manner. Although IMS is very well suited for terrestrial IP networks, it is challenged by high propagation delay satellite environment with low data rates to provide equivalent levels of quality of experience enjoyed by subscribers of circuit switched systems. In this article, we first provide an overview of Circuit Switched and IMS Packet Switched based call flows for mobile satellite systems. We then concentrate specifically on voice call setup time and show how Circuit Switched architecture provides significantly better performance. Lastly, we propose several advancements on SIP call signaling to improve the IMS Packet Switched call setup times.
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J2EE based SIP application server is a scalable middleware for containing and managing SIP application. In this paper, a kind of SIP accelerator named SIP offload engine (SOE) is presented to improve server performance through offloading SIP message parsing, SIP transaction processing, security and transport layer connections management. The SIP application can be deployed on SOE enabled SIP application server without any modification. The benchmarking results on typical SIP application scenario indicate that the throughput of SIP application server can be improved significantly. Furthermore SOE can be used as high performance HTTP and SIP converged proxy server.
Conference Paper
Session setup time is a critical factor for multimedia applications. Some of the factors that affect the session setup delay are length of SIP messages, bandwidth and packet loss in wireless link This paper aims at reducing the session setup delay in UMTS IP Multimedia Subsystem. We propose compression of SIP header by Text-based Compression using Cache and Blank (TCCB) approach and a caching scheme. In caching scheme, during registration, the address of serving call session control function (S-CSCF) of mobile users are cached at Interrogating-CSCF (I-CSCF). Typically, it takes 300 ms to find the address of called mobile user's Serving CSCF and it can be minimized using cache. A simulation has been carried out to analyze the effect of SIP header compression and caching scheme. The result shows that for 9.6 Kbps channel, a reduction of 25% is possible and for 19.2 Kbps channel 19% reduction can be achieved.
Article
Internet telephony has been the focus of much recent effort by ITU and IETF standards bodies, with initial, albeit small-scale deployment in progress. While Internet telephony voice quality has been studied, call setup delay has received little attention. This paper outlines a simulation study of Internet Telephony Call Setup delay, based on UDP delay/loss traces. The focus is signaling transport delay, and the variations arising from packet loss and associated retransmissions. Of particular interest are the differences arising from H.323 signaling, which uses TCP, and SIP, which can use UDP with additional error recovery. Results show that during high error periods, H.323 call setup delay significantly exceeds that of SIP. We also consider PSTN/Internet telephony interworking, and show that high blocking rates are likely if either H.323 or SIP are used across the public Internet. I. INTRODUCTION Internet telephony is experiencing significant growth, prompted initially by low-price l...
Signaling Flows for the IP Multimedia Call Control Based on SIP and SDP -Stage 3 (Release 5), TS 24
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  • Cn
3GPP TSG CN, Signaling Flows for the IP Multimedia Call Control Based on SIP and SDP -Stage 3 (Release 5), TS 24.228 v. 2.0.2, 2002-03.
Delay in Mobile Video-phones, Seventh International Workshop on Mobile Mul-timedia Communications
  • A Hourunranta
  • I D D Curcio
A. Hourunranta and I.D.D. Curcio, Delay in Mobile Video-phones, Seventh International Workshop on Mobile Mul-timedia Communications (MoMuC 2000), 23-26 October 2000, Tokyo, Japan, pp. 1B-3-1/1B-3-7.
Call Control Protocol Based on SIP and SDP -Stage 3 (Release 5), TS 24
  • gpp Tsg Cn
  • Ip
3GPP TSG CN, IP Multimedia Call Control Protocol Based on SIP and SDP -Stage 3 (Release 5), TS 24.229 v. 2.0.1, 2002-03.
An Offer/Answer Model with SDP, IETF RFC 3264
  • J Rosenberg
  • H Schulzrinne
Network Grade of Service Parameters and Target Values for Circuit-Switched Services in the Evolving ISDN
  • Itu-T Recommendation