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SIP call setup delay in 3G networks
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.
... 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 ...
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. ...
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. ...
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. ...
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. ...
Mobile telecommunications are converging towards all-IP solutions. This is the case of the Long Term Evolution (LTE) technology that, having no circuit-switched bearer to support voice traffic, needs a dedicated VoIP infrastructure, which often relies on the IP Multimedia Subsystem architecture. Most telecom operators implement LTE-A, an advanced version of LTE often marketed as 4G+, which achieves data rate peaks of 300 Mbps. Yet, although such novel technology boosts the access to advanced multimedia contents and services, telco operators continue to consider the VoIP market as the major revenue for their business. In this work, the authors propose a detailed performance assessment of VoIP traffic by carrying out experimental trials across a real LTE-A environment. The experimental campaign consists of two stages. First, we characterize VoIP calls between fixed and mobile terminals, based on a dataset that includes more than 750,000 data-voice packets. We analyze quality-of-service metrics such as round-trip time (RTT) and jitter, to capture the influence of uncontrolled factors that typically appear in real-world settings. In the second stage, we further consider VoIP flows across a range of codecs, looking at the trade-offs between quality and bandwidth consumption. Moreover, we propose a statistical characterization of jitter and RTT (representing the most critical parameters), identifying the optimal approximating distribution, namely the Generalized Extreme Value (GEV). The estimation of parameters through the Maximum Likelihood criterion, leads us to reveal both the short- and long-tail behaviour for jitter and RTT, respectively.
... We plan to evaluate and possibly use them in future work. More study on signaling delay can be found in [17,18]. ...
In this paper, we present the creation and characterization of an open source Internet Protocol Multimedia Subsystem (IMS) test bed. We built this test bed with the intention and motivation of doing research related to the performance of its various functional components as they cooperate to provide IMS features. The IMS specifications describe a set of functional components, the interfaces between those components, and the protocols and procedures to be used across those interfaces. The test bed is built based on the "Open IMS Core", which is an open source set of IMS functional components. We verify that the operation of the test bed is in conformance with the message sequence charts provided by the IMS standards. Moreover, we define four metrics that quantify the overhead associated with IMS implementations, and present the observed values of these metrics collected on the test bed under the conditions of no background traffic. This provides necessary and valuable data for us in order to study the performance when the test bed is under various background traffic loads.
... 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. ...
Despite the popularity of VoIP these clays, this method of communication may present significant security challenges in terms of privacy and accounting. Authentication and message encryption are considered essential mechanisms in VoIP for it 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 SIP performance. However, these overheads have not been understood in detail by the technical community. In this paper we demonstrate the effect of a security protocol on the performance of SIP by comparing the call setup delays that occur with various security protocols. We simulate the various combinations of three security protocols and three transport-layer protocols suggested for SIP. UDP with any combination of security protocols performs much better than the combination of TCP. TLS over SCTP may have more impact on performance on average. The reasons for the differences in SIP performances are given.
... 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: ...
SIP is receiving much attention these days and it seems to be the most promising candidate as a signaling protocol for the current and future IP telephony services. Realizing this, there is the obvious need to provide a certain level of quality comparable to the traditional telephone service signalling system. Thus, we identified the major costs of SIP, which were found to be delay and security. This thesis discusses the costs of SIP, the solutions for the major costs, and the development of a low cost SIP application. The literature review of the components used to develop such a service is discussed, the networks in which the SIP is used are outlined, and some SIP applications and services previously designed are discussed. A simulation environment is then designed and implemented for the instant messaging service for wireless devices. This environment simulates the average delay in LAN and WLAN in different scenarios, to analyze in which scenario the system has the lowest costs and delay constraints.
... 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. ...
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. ...
Session Initiation Protocol (SIP) is an application layer protocol that controls multimedia sessions of Third Generation Mobile
Networks. A major concern with SIP in 3G networks is ensuring the efficient use of the bandwidth constraints inherent in the
wireless medium. Because of the significant size of SIP messages, 3G terminals can suffer from substantial set-up delays.
IETF has proposed a general solution to this problem, named SigComp. This paper investigates the performance of SigComp, in
particular how much time saving the scheme is capable of offering. A novel scheme that can replace SIP for increasing time
savings and reducing the load of the SigComp layer is proposed. The scheme, referred to as Binary Hybrid SIP (BHSIP), is based
on binary sequences and plain text. This paper evaluates the performance of the scheme, and compares it with normal SIP. The
paper provides experimental results obtained when comparing the performance of SigComp/SIP versus SigComp/BHSIP by using an
emulated network environment, and discusses the benefits of using BHSIP.
... 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. ...
Mobile telecommunications are converging towards all-IP solutions. This is the case of the Long Term Evolution (LTE) technology that, having no circuit-switched bearer to support voice traffic, needs a dedicated VoIP infrastructure, which often relies on the IP Multimedia Subsystem architecture. Most telecom operators implement LTE-A, an advanced version of LTE often marketed as 4G+, which achieves data rate peaks of 300 Mbps. Yet, although such novel technology boosts the access to advanced multimedia contents and services, telco operators continue to consider the VoIP market as the major revenue for their business. In this work, the authors propose a detailed performance assessment of VoIP traffic by carrying out experimental trials across a real LTE-A environment. The experimental campaign consists of two stages. First, we characterize VoIP calls between fixed and mobile terminals, based on a data- set that includes more than 750,000 data-voice packets. We analyze quality-of-service metrics such as round-trip time (RTT) and jitter, to capture the influence of uncontrolled factors that typically appear in real-world settings. In the second stage, we further consider VoIP flows across a range of codecs, looking at the trade-offs between quality and bandwidth consumption. Moreover, we propose a statistical characterization of jitter and RTT (representing the most critical parameters), identifying the optimal approximating distribution, namely the Generalized Extreme Value (GEV). The estimation of parameters through the Maximum Likelihood criterion, leads us to reveal both the short- and long-tail behaviour for jitter and RTT, respectively.
... 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. ...
VoIP is born from the growing Internet infrastructure, which has over the years seen significant improvements in both bandwidth and end-to-end latency. In this paper, we explore making VoIP available on a mobile phone. For that purpose, we propose an architecture and describe the various components involved. Data entering and leaving the mobile phone is encapsulated in a wire-less Bluetooth connection. The bridge to the Internet is provided by a linux Bluetooth access point. The system is compatible with current VoIP standards using RTP and SIP for data and signaling transmission, respectively. It has been tested to work in combina-tion with several known softphones. Both RTP and SIP stack are built on top of a special IP emulation layer on the mobile phone, which has been developed to facilitate the migration of the appli-cation onto devices having a direct IP binding available for Blue-tooth such as BNEP (Bluetooth Network Encapsulation Protocol). Apart from presenting design and implementation details, the pa-per also provides measurement results with regard to delay and ses-sion setup time and discusses certain limitations of the application evolving from restrictions imposed by the mobile phone's program-ming platform.
... Where Quality of Service (QoS) issues for media transport in IP networks has been in the focus of the research community for many years, QoS provisioning specifically for signalling messages has received less attention. Signalling message delay and session initiation delay was addressed by Eyers [7] and Curcio [8]. Earlier work [9] outlined the need for QoS considerations on the SIP layer to provide equivalent telephony services, in particular, if SIP is to be used in 3GPP carrier-grade networks. ...
The Session Initiation Protocol (SIP) is widely ac-cepted as the IETF alternative the ITU-T H.323 tele-conferencing protocol to enable call and media ses-sion management and control. It is also used in carrier grade environments, such as the IP Multime-dia Subsystem (IMS) of the 3rd Generation Partner-ship Project (3GPP) in emerging Universal Mobile Telecommunications System (UMTS) networks. This paper proposes a framework that enables the Quality of Service (QoS) provisioning for signalling messages. The contributions are twofold: Firstly, it defines an overall framework to enable QoS provision-ing for SIP signalling messages in carrier grade net-works. Secondly, it outlines existing work that fits into the framework and identifies areas that require further investigation to implement the concepts in car-rier grade networks.
... 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. ...
In this letter, we consider session initiation protocol (SlP)-based voice over wireless local area networks (VoWLANs). We derive the analytical expression for the average call setup latency, and we observe the impact of the number of contending mobile nodes in a WLAN. Extensive simulation results are given to validate the analytical results.
... 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. ...
Mobile telecommunications are converging towards all-IP solutions. This is the case of the Long Term Evolution (LTE) technology that, having no circuit-switched bearer to support voice traffic, needs a dedicated VoIP infrastructure, which often relies on the IP Multimedia Subsystem architecture. Most telecom operators implement LTE-A, an advanced version of LTE often marketed as 4G+, which achieves data rate peaks of 300 Mbps. Yet, although such novel technology boosts the access to advanced multimedia contents and services, telco operators continue to consider the VoIP market as the major revenue for their business. In this work, the authors propose a detailed performance assessment of VoIP traffic by carrying out experimental trials across a real LTE-A environment. The experimental campaign consists of two stages. First, we characterize VoIP calls between fixed and mobile terminals, based on a data-set that includes more than 750,000 data-voice packets. We analyze quality-of-service metrics such as round-trip time (RTT) and jitter, to capture the influence of uncontrolled factors that typically appear in real-world settings. In the second stage, we further consider VoIP flows across a range of codecs, looking at the trade-offs between quality and bandwidth consumption. Moreover, we propose a statistical characterization of jitter and RTT (representing the most critical parameters), identifying the optimal approximating distribution, namely the Generalized Extreme Value (GEV). The estimation of parameters through the Maximum Likelihood criterion, leads us to reveal both the short-and long-tail behaviour for jitter and RTT, respectively.
... 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). ...
In this thesis, we consider the efficient distribution of live and scheduled multimedia content (e.g., radio and TV broadcasts) to mobile users via a ubiquitous wireless Internet. The objective is to design and develop a content delivery system that (1) enables content owners to deliver their multimedia content to a large number of heterogeneous receivers, and (2) enables receivers to continuously receive that content, independent of their location and the network they connect to.
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 network users in NGN. This paper defined parallel control schemes for NGN resource control interfaces to minimize session and resource control delays. We simulated the existing and proposed NGN control schemes to measure and analyze control delays and completion ratios. By arrival rate of 120 in two-phase resource control, we checked out that completion ratios of sequential and parallel schemes achieved 100%, and the control delay of parallel scheme was improved by about 21.5% compared to that of sequential scheme.
In mobile voice over IP (VoIP) systems, user mobility leads to the increased call setup latency to find out the current location of user, which has a negative impact on users call quality. In this letter, we propose a prefetching technique for user location to reduce the call setup latency. We develop an analytical model and investigate the impacts of mobility parameters and thresholds.
The accurateness of round-trip- and one way delay measurements for 2G and 3G networks rely to a much larger extent on employing a sound methodological framework than this is the case for other types of networks. Typical mobile access networks differ significantly from core networks, most prominently with respect to delay. In this paper we present payload-dependent delay measurement results for public 2G and 3G networks which illustrate that accurate IP-layer delay measurements in mobile networks must use high sample counts and randomness in start times for uplink and for downlink. Most notably this concerns ICMP round-trip delay measurements which, due to the synchronization of ICMP requests with the network clock when leaving the mobile network's uplink, fail to meet the random start time criterion for ICMP replies in the downlink (or vice-versa). This synchronization effect leads to significant clustering of one-way delay values for the reply leg and causes minor delay differences in the core network to have potential significant impact on ICMP round-trip delays. Therefore, highly accurate simulations and emulations must model uplink and downlink of time-slotted networks as two interrelated links based on a common timebase.
To determine the potential signaling traffic reductions, the session establishment procedures are investigated. The investigation shows that, the S-CSCF (Serving Call Session Control Function) is the major bottleneck in IMS (IP Multimedia Subsystem) and the existing 3GPP (the 3rd Generation Partnership Project) Service Triggering Algorithm (3GPP STA) increases largely the end to end session setup delay. To reduce the session setup delay and improve the system performance, a new Group based Service Triggering Algorithm (GSTA) is proposed. And then the modeling of 3GPP STA and GSTA are presented. Theoretical analysis and simulation results show that, GSTA can efficiently reduce the signaling traffic load of the S-CSCF, increase the throughput of the system and considerably reduce the session setup delay, improve IMS network quality of service.
To determine the potential signaling traffic reductions, the session establishment procedures were investigated. The investigation showed that, the S-CSCF (serving call session control function) is the major bottleneck in IMS (IP multimedia subsystem) network. And then the modeling of 3GPP (3rd Generation Partnership Project) service triggering algorithm (STA) in IMS was presented. The session setup delay was employed as the measurement to investigate the performance of the ATA (application triggering architecture) and S-CSCF. Analytical results showed that, the number of the ASs, the use of the subsequent filter criteria (sFC) and the arrival rate have heavy impact on the session setup delay. The results will be useful in constructing IMS network. At last, we proposed several possible solutions to reduce the session setup delay decrease the load of the S- CSCF and improves the QoS (quality of service) of IMS network.
Due to Internet Protocol (IP) is the most important protocol in Next Generation Networks. 3GPP utilizes Session Initial Protocol (SIP) based on IP as the base protocol for negotiating sessions in IP Multimedia Subsystem (IMS). Different from traditional circuit-switched network, the media traffic and signaling are delivered through IP transport layer in IMS. The media traffic may affect the signaling efficiency in core network because the traffic collisions and delivery packets in best effort. This paper proposes a novel path-migration mechanism for enhancing the traffic efficiency in IMS. The simulation results show that by adopting proposed path-migration mechanism, the interference and traffic collision can be reduced and the signaling efficiency in core network can be improved.
To determine the potential signaling traffic reductions, the session establishment procedures were investigated. The investigation showed that the serving call session control function (S-CSCF) is the major bottleneck in IP multimedia subsystem (IMS) network and the existing third-generation partnership project (3GPP) service triggering algorithm (3GPP STA) based on initial filter criteria (iFC) increases largely the end-to-end session setup delay (SSD). To reduce the SSD and improve the system performance, a call-state-based application triggering architecture (CATA) was proposed. In CATA, the call-state-based filter criteria and the call-state-based service triggering algorithm (CSTA) were proposed, and then the modeling of 3GPP STA and CSTA was presented. Theoretical analysis and simulation results showed that the CSTA avoids the additional signaling traffic load of the S-CSCF and application server effectively, which exists in 3GPP STA, reduces the SSD considerably, increases the throughput of the S-CSCF and improves the quality of service of IMS network. Copyright © 2008 John Wiley & Sons, Ltd.
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.
In this letter, we analyze the SIP transfer delay over multi-rate wireless networks where different transmission modes based on adaptive modulation code are supported. We consider a Nakagami m fading channel at the physical layer and automatic repeat request at the data link layer. Extensive simulation results are given to validate the analytical results and to demonstrate the effects of message sizes and channel conditions.
Since Internet Protocol (IP) is the most important protocol in Next Generation Networks (NGNs), 3rd Generation Partnership Project (3GPP) utilizes Session Initial Protocol (SIP) based on IP as the base protocol for negotiating sessions in IP Multimedia Subsystem (IMS). Different from traditional circuit-switched network, in IMS, the media traffic and signaling are delivered through IP transport. The media traffic may affect the signaling efficiency in core network, due to traffic collisions and best effort packets delivery. This paper proposes a novel path-migration mechanism for enhancing the traffic efficiency in integrated NGN-IMS. The simulation results show that the interference and traffic collision can be reduce by applying proposed path-migration mechanism and the signaling efficiency in core network can be improved with higher system capability and voice quality.
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.
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.
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
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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 ...
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.
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.
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.
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.
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
- gpp Tsg
- 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