A Survey And Analysis Of Handoff Techniques

Published Date: 02 Nov 2017

Abstract:

In the modern world, wireless network plays an important role in the field of communication. Many research works have been undergone in wireless communication. Next generation Wireless network is the one of the upcoming and optimistic approach. Here a multimedia based service is carried out. Handoff property is the, main concern in wireless network. In this survey we have analyzed many existing problems in next generation and wireless communicating. Many results are analyzed in handoff technique and many precision approaches show that handoff is the main parameter that has to be optimized. The packet transfer from source to destination should be safe irrespective of network, area, device and traffic. A survey of near 35 papers is done and the problem in the existing system is analyzed and a proposed methodology will be derived from the existing methodology. After examining all the wireless related details, a finding is obtained, i.e. SIP- Session Initiation Protocol provides a better way of thinking. SIP generally works in the principle of transaction. It is used to create, manage and terminate the session based on 2 way communication protocols. SIP generally provides user to located areas and user presented hotspots where the packet transfer can be done easily. SIP generally works in the principle of receiving individual manages and then establishing a link be6ween source end to the destination end. After linking the packet are transferred to the destination and then an acknowledgment is received after the packet is transferred. So in general SIP works in the transaction based policy. In this work a survey of nearly 32 SIP related papers are done and problem are identified based on survey.

I. Introduction

SIP is signalling protocol that can create, change, and terminate multimedia sessions in as Internet Protocol based Networks. SIP supports variation capability structures, like: i) Inviting participants to existing sessions, ii) Adding and removing media from a session, iii) Supports personal mobility. SIP supports five methods of establishing and terminating multi-media communications: User location: find out location of the target endpoint, User availability: Check of the willingness of the called party to engage in communications, User capabilities: Discovering of the media and media parameters to be used, Session setup: "Ringing", establishment of session parameters at both called and calling party, Session management: This includes transfer and termination of sessions, modifying session parameters, and invoking services.

SIP Messages

SIP messages are basically two types like requests and responses.

Request

The SIP request is basically six types as follows:

REGISTER: User agent to specify its current position.

INVITE: Invite between two user agents. Generally send by user agent clients.

ACK: Approves an INVITE message.

CANCEL: Terminates the request.

BYE: Terminate the session.

OPTIONS: is used to requests the abilities of a caller or server.

Response

The response methods are collected in the following class:

1xx: informational responses like 100 (Trying), 180 (Ringing).

2xx: indicates success of request like 200(OK).

3xx: redirection of the call like 300 (Multiple choices).

4xx: request failure like 400 (Bad Request) and etc.,

5xx: server error like 500 (Server Internal Error) and etc.,

6xx: global failures like 600 (Busy Everywhere).

SIP Architecture

From architecture viewpoint, there are two basic components of SIP: SIP user agent (UA) and SIP network server. As illustrated in Figure 1, the SIP server is an intermediate device.

Figure 1: SIP architecture

SIP User Agent (UAC): A SIP user agent (UA) is logical thing that starts and replies calls between two nodes. UA is divided into two types:

User Agent Client (UAC) is SIP client that sends INVITE message to SIP server call.

User Agent Server (UAS) that respond to SIP requests and answers to SIP server call.

UA can be a mobile phone, software or any device.

SIP Call Connection:

A SIP call between two user agents over a SIP server.

Figure 2: SIP Call connection

II. Related Work

Henning Schulzrinne et al., [1] proposed that the Session Initiation Protocol (SIP) provides advanced signaling and control functionality for a wide variety of multimedia services. Based on a location-independent name and then negotiate session characteristics, SIP can efficiently and scalably place resources. SIP was used in applications ranging from Internet telephony, control of networked devices and conferencing to instant messaging, event notification. They summarized the main protocol features and describe a range of extensions currently being discussed within the Internet Engineering Task Force.

With ftp, the notion of controlling a data stream with a control stream dates back to 1972. Telephony signaling protocols progressed from channel-associate signaling to out-of-band signaling. Q.BICC tried to allow ISUP to set up connections between IP endpoints. In 1996, the International Telecommunications Union (ITU) first published the H.323 suite of protocols, with the ISDN-derived signalling mechanism, and has updated the specification since then. SIP was related to Internet e-mail which was a messaging mechanism. It shares the ability to deliver information without knowing the precise network location of the recipient. With end-to-end notification of success or failure, and designed for sub second delivery times SIP messaging was synchronous. Setup of large-scale sessions using multicast was considered by ICEBERG project.

Nilanjan Banerjee et al., [2] presented the most challenging problems towards the system integration of 4G wireless networks was providing seamless mobility support. Application-layer mobility management protocol like the SIP has been considered as the right candidate for handling mobility in the heterogeneous 4G wireless networks, because of transparency to lower layer characteristics. SIP was able to provide support for not only terminal mobility but also for service mobility, personal mobility and session mobility. However, in heterogeneous environment, the performance of SIP, operating at the highest layer of the protocol stack, was only as good as the performance of the underlying transport layers.

To achieve different goals, several mobility protocols was proposed [2] for Wireless Internet and targeted different layers of the network. The dependency of the mobility protocols in the underlying layers was reduced when they operate higher in the protocol stack; even each of them had the same goal of providing location transparency. No single network specific mobility protocol is expected to work for all because varieties of wireless network technologies were believed to co-exist in the next generation networks. Tremendous efforts were required for the design of a uniform mobility protocol that will work on all different networks. The only way out seems to apply the mobility management functionality in the application layer, where there is smallest amount of dependency on the lower layers. SIP, widely accepted as a signalling protocol but capable of providing mobility hold at the application layer, satisfies this condition. They analyzed the handoff performance of SIP in a IP-based 4G network with Wireless LAN (WLAN) access networks and Universal Mobile Telecommunication System (UMTS). The handoff to a UMTS access network introduced a minimum delay for 128 kbps channel was 1.4048 s, while for handoff to a WLAN access network the minimum delay is 0.2 ms, shown by Analytical results. The minimum delay was unacceptable in the former case for streaming multimedia traffic and in order to reduce the handoff delay to a desirable maximum limit of 100 ms, it requires the use of soft-handoff and advanced resource reservation techniques.

Kalyan Basu et al., [3] presented that the future-generation wireless networks have been envisioned as the combination of various wireless access networks, including both wireless local area networks and wireless wide area networks. Because of transparency to lower-layer characteristics, greater scalability and ease of deployment, the right candidate for managing mobility in heterogeneous wireless networks was the application- layer-based Session Initiation Protocol. SIP involves application-layer transport and processing of messages, which introduced considerable delay. They analyzed the delay associated with vertical handoff in WLAN-UMTS internetwork using SIP, as case study of the performance of mobility management protocols in mixed wireless networks. WLAN-to-UMTS handoff gives unacceptable delay for supporting real-time multimedia services, and this is due to transmission of SIP signalling messages over erroneous and bandwidth-limited wireless links shown by analytical results. UMTS-to-WLAN handoff experiences very less delay, mainly contributed by the processing delay of signalling messages at the WLAN servers and gateways. Former case requires reducing the delay by the deployment of soft handoff techniques, faster servers and efficient host configuration mechanisms required in the latter case.

Future-generation wireless networks were envisioned as the combination of various wireless access networks. In such a varied network environment, faultless mobility support is the basis of providing wireless services to mobile users wandering between various wireless access networks, which was uninterrupted. SIP, most widely accepted signaling protocol, which was capable of providing mobility support on the application layer, which needs very less amount of dependence on the lower layers. They performed a case study of vertical handoff delay analysis in WLAN-UMTS inter-network using SIP as the terminal mobility management protocol. Numerical analysis has shown that the WLAN-to-UMTS handoff due to error-prone and bandwidth-limited wireless links incurs much larger delay than the UMTS-to- WLAN handoff. In order to comply with the maximum limit of the handoff delay for supporting delay-sensitive applications, soft handoff techniques need to be applied for SIP-based terminal mobility management in heterogeneous wireless IP networks.

Samir Chatterjee et al., [4] presented the next generation of enterprise networks was undergoing major changes as a excess of new applications, architectures, and services started to roll out inside businesses. In global communications network, the world of video, voice/telephony and data are "converging". Purpose of their paper was twofold. The design, analysis, and performance of a SIP - based videoconferencing desktop client was first developed and deployed over Internet2, was presented. A guideline for managing SIP-based services to be deployed within enterprises was planned by them to perform secondly, which addresses several challenges in each layer such as directory service integration issues, network address translator (NAT)/FW issues, and interoperability issues, is proposed. Through network traffic measurements findings of extensive SIP/NAT traversal analysis were reported. The lessons learned from both, management challenges with enterprise deployment as well as the design of a new SIP-based voice/video client were highlighted. The work [4] was based on substantial experience in dealing with SIP-based multimedia services deployment within enterprise. While highlighting the lessons learned from the process, they described the design and architecture of a new SIP-based video client. They also raised important deployment issues in various stack levels and presented the experimental analysis of the solutions that were available and lessons learned. They hope that the technical and managerial guidelines provided can be useful to network administrators and managers who are contemplating on deploying SIP-based solutions. They did not focus much on securing SIP communication in their paper. They saw lot of challenges as a future work in completely securing this SIP environment. The mechanisms of using certificate authorities (CAs), digital certificates, S/MIME for end-to-end encryption, and managing trusts between domain CAs (federated identity management) will be critical as they moved toward a more robust production environment.

Ing-Chau Chang et al., [5] they know that the traditional approaches such as the Hierarchical Remote Subscription (HRS) and Bi-directional Tunnelling with Mobile Multicast (BT-MoM) integrate the Mobile IP protocol with the IP multicast technique but they planned for supporting the mobile host (MH) to be the multicast receivers on wireless networks. However, they inherit intrinsic problems to handle mobility and suffer important delays when the MH moves out of the multicast tree ,tree should be rebuild, that results in serious playback interruption and quality of service (QoS) degradation for the on-going multimedia services especially when the MH are dispersed over different kinds of wireless networks. They [5] proposed a cross-layer framework and the Hierarchical Multicast Session Initiation Protocol (HMSIP), which lengthened the IETF Session Initiation Protocol (SIP) with the concept of multicast SIP session and combined the underlying IP Multicast QoS routing protocol and the RSVP resource reservation technique, to upkeep mobile multimedia multicasts. Their outline meaningfully reduces deployment costs and handoff delays and hence achieved end-to-end QoS support for MHs on varied wireless networks. The mathematically [5] analyse handoff delays of the HRS, BT-MOM and HMSIP schemes as a further proceedings. Furthermore, simulation results presented that the HMSIP can knowingly achieve much more stable and lower handoff delays than those of the BT-MOM and HRS only with a slight proportion of additional bandwidth than the bandwidth-optimal HRS, thus supporting higher numbers of handoffs without disturbing multimedia playback of the mobile multicast activity. Trade-offs between costs for deploying HMSIP Proxies in the outline and corresponding performances of the HMSIP families were discussed. Finally, multimedia QoS enhancements of MHs were shown to highlight advantages by integrating RSVP in HMSIP outline.

For supporting multiple MHs in heterogeneous wireless networks to receive high quality of real-time multimedia streams, they proposed the cross-layer HMSIP basis to assimilate the SIP protocol, IP multicast and RSVP resource reservations in their proposed to provide a low deployment cost scheme that decreases the handoff delay for session and mobility managements, consumed link bandwidth and satisfies multimedia end-to-end QoS for the MH, as compared to the two traditional multicast BT-MOM and HRS. From the simulation results, they concluded that the HMSIP achieves significant multicast performance developments over the BT-MOM and HRS schemes on the heterogeneous wireless networks.

Hanane Fathi et al., [6] presented that the wireless networks beyond 2G aim at supporting real-time applications such as VoIP. They proposed that, before a user can start a VoIP session, terminal at the end-user has to establish the session using signalling protocols such as SIP and H.323 in order to negotiate media factors. The term session setup time was defined as the time interval to perform the session setup. It can be affected by the quality of the wireless link, measured in terms of frame error rate (FER), which resulted in retransmissions of packets lost and increased the session setup time. So, such protocols should have a session setup time which was optimized against loss. One way to do so is by choosing the suitable retransmission timer and the underlying protocols. In their paper, they focussed on SIP session setup delay and proposed optimizing it using an adaptive retransmission timer. They also evaluated SIP session setup performances as a function of the FER with various underlying protocols such as user datagram protocol (UDP), radio link protocols (RLPs), transport control protocol (TCP)). For 19.2 Kbps channel, the SIP session setup time can be up to 7s with TCP and 6.12s with UDP when the FER is up to 10 percentages. The use of RLP (1, 2, 3) and RLP (1, 1, 1, 1, 1, 1) places the session setup time down to 3.4s underneath UDP and 4s underneath TCP for the same FER and the same channel bandwidth. They also compare SIP and H.323 performances by an adaptive retransmission timer: SIP outperforms H.323, especially for a FER higher than 2 percentages.

In their paper [6], they have proposed a novel adaptive transmission timer that is adjustable to the size of signalling packets involved in the session establishment. Using analytical model, they assessed the average SIP session setup depending on the FER of the wireless link and the processing power of the servers and source/destination stations (queuing delays). The choice of UDP(User Datagram Protocol) or TCP(Transmission Control Protocol) to transport SIP messages effects the session setup time for FER higher than 2 percentage. To use UDP in its place of TCP can make the session setup 10 percentage shorter for FERs higher than 4 percentages. Session setup delay was considerably improved by low-layer retransmission mechanisms, such as RLP. With RLP the session setup delay remains small (4-7s). RLP (1, 2, 3) outdoes RLP (1, 1, 1, 1, 1, 1) only for FER higher than 3-4 percentage in the environments with high FER. With the adaptive timer, SIP gives a smaller delay than H.323 for FERs higher than 2 percentages. But, for an FER less than 2 percentage, H.323 is as per formant as SIP due the timer adjustable to the size of messages (H.323 has more messages but much smaller than SIP). Therefore, in general the adaptive timer was efficient for optimizing the performance of signalling protocols. The performance of SIP using the adaptive timer could be improved by using some compression schemes to decrease the size of the SIP messages. Also, error correction mechanisms or hybrid ARQ schemes could progress the performance of VoIP session setup time by correcting the SIP messages and avoiding retransmissions on the wireless link.

Qi Wang et al., [7] presented that, in the all-IP wireless networks beyond the third generation, mobility management can be successfully achieved by applying jointly Mobile IP (MIP) and the Session Initiation Protocol (SIP). Yet, an efficient combination of both protocols remains an open research issue. MIP and SIP almost operated independently by Conventional hybrid MIP-SIP mobility architectures, resulting in important redundant costs. Their article investigates the representative hybrid MIP-SIP architectures and explores the joint optimizations between MIP and SIP for a more cost-efficient mobility support whilst utilizing their balancing power. Two novel design approaches were presented. The first approach was to produce maximum system efficiency, which concludes in a tightly integrated architecture, that combines the redundant mobility entities in MIP and SIP. The other approach leads to a loosely integrated architecture, where necessary interactions were introduced between MIP and SIP mobility servers while their physical entities were kept complete. Major mobility procedures, including session setup, location update and handoff were discussed in those architectures. The analytical results demonstrated that both proposed architectures outperform characteristic hybrid MIP-SIP architectures in terms of straightforward reduced signalling costs.

They have [7] investigated mobility management in all-IP wireless networks where MIP and SIP coexist. To support both UDP and TCP mobility effectively, mobility management based on joint MIP and SIP protocols were emerging. To resolve the excessive costs due to the redundancies found in the hybrid MIP-SIP architectures, they proposed the tightly and loosely integrated architectures. In the tightly integrated architecture, redundant MIP and SIP entities and operations of similar functionality were merged and the cost saving was maximized. In the loosely integrated architecture, two schemes were devised to establish the desired interactions between MIP and SIP servers. In Scheme I, only MIP HA tracks the location of an MH, and SIP HS uses MIP HA as a location service. In Scheme II, MIP HA updates SIP HS. All interactions are based on MIP messages. The analytical results show that the proposed architectures improve the system efficiency by meaningfully reducing the signalling costs compared with the traditional hybrid MIPSIP schemes. Regarding the two approaches, the tightly integrated one may prove most cost-efficient in a long run, while the loosely integrated architecture, particularly Scheme I, may be chosen in the short-to-medium term, since this approach does not change the physical entities or constrain the physical locations of the entities whereas being capable of achieving similar costs saving. Finally, the principles of the proposed approaches were applicable to both MIPv4- SIP and MIPv6-SIP, though the MIPv4 context has been demonstrated for the purpose of comparison.

Min-Xiou Chen a et al., [8] presented that most of Voice over Internet applications were based on the SIP. Their paper addresses issues of allowing a user to continue to communicate with a remote party while changing terminals over multiple devices. Specifically, they proposed SIP extension header to improve the Call Transfer mechanism and hide the changing of the terminal from the remote party. They also propose a mechanism to solve the problem of the user needed to terminate all devices separately when a session is divided over multiple devices. Finally, the proposed mechanisms were implemented using Sip-Communicator, an open source of SIP Split a SIP session over multiple devices will become an important application in the near future. In their paper, they proposed a number of extensions for the user agent to get the ability to split a session. First, they proposed a SIP extension header "Mobility to improve the Call Transfer mechanism and make it clear as crystal to the remote party. Then, they proposed the theory of "Association" in the CN to solve the crisis of the user having to cease all devices independently when a session was tearing over multiple devices. Furthermore, they made an valuation to show that sending a REFER request to the local devices is better than sending it to the CN. At last, they employed a SIP user agent with the capability to divide a session over multiple devices, and they tested it in our experimental environment. The CN in their scenarios were measured as the remote party using only a single device lacking any additional actions. Yet, the CN might divide a session over multiple devices as well. Their protocol is not completely suitable for this situation. At the similar time, a user may want to assemble multiple sessions to a single device. They did not address that issue either. So, they plan to examine how to use the Split session mechanism freely at any party and collect sessions from multiple devices to a single device in the future.

Swetkecher et al., [9] presented a concept for sip, Mobility managing protocols functioning from different layers of the standard protocol stack (e.g., link, network, transport, and application layers) have been suggested in the last numerous years. These protocols attain different handoff concert for dissimilar kinds of applications. In this paper, mobile applications are assembled into five dissimilar classes, Class A over Class E, centered on their mobility managing requirements. Analytical models are advanced to explore the handoff concert of the prevailing mobility managing protocols for these application classes. The analysis displays those applications of a particular class practice different handoff concert when different mobility managing protocols are used. Handoff performance assessments of unlike mobility management protocols are conceded out to resolve on the apt mobility managing protocol for a exact application class. The outcomes of calculated analysis believe the practice of transport layer mobility managing for Class B and Class C applications, Mobile IP aimed at non- real -time Class D and Class E applications, and Session Initiation Protocol- centered mobility managing for real-time Class D and Class E applications. Besides, over analytical modelling, the constraints that influence the handoff concert of mobility managing protocols are well-known. These parameters can be used to scheme new application-adaptive methods to improve the handoff concert of the prevailing mobility managing protocols.

To condense, our analysis shows that the handoff enactment of a mobility management protocol be contingent on the subsequent factors: i) Type of application: Diverse applications use different transport layer protocols. As the functioning principles of dissimilar transport layer protocols are unlike, they respond contrarily to the handoff. Hence, the concert of a certain mobility managing protocol is unlike for different kinds of applications. For instance, as deliberated earlier, the handoff dormancy of Mobile-IP-based handoff is greater for applications by means of TCP than applications exhausting UDP. This is as, after packets are lost throughout the handoff, TCP went over retransmission timeouts earlier retransmitting the missing packets. ii) Link layer frame error possibility: Our inquiry displays that the handoff expectancy, end-to-end packet transportation deferral, and packet damage throughout handoff are subject to on the link layer frame error possibility pf Þ, both while no RLP is used and once RLP is used. iii) Signalling delay: Handoff potential and packet damage during handoff be determined by on the signalling deferral among the network units that are intricate in a handoff, e.g., MH and HA in the event of Mobile IP and MH and CH in the incident of SIP and TCP migrate.

Link layer admittance technologies: As witnessed in our study, diverse categories of link layer access technologies such as the practice of RLP also stimulus the statistical value of handoff constraints. Besides, the link layer admittance deferral that is unlike for different access technologies also impacts the handoff concert. Centered on our handoff performance exploration, we sponsor the use of TCP-Migrate for applications exhausting TCP, i.e., Class B and Class C applications. SIP is apt for real-time applications exhausting UDP. Still, SIP is identical solitary for real-time applications; hence, Mobile IP can be used for non-real-time applications that practice UDP. In summary, dissimilar mobility managing protocols functioning from diverse layers of the conventional protocol stack are fit for different classes of applications. The custom of application-adaptive mobility itself is not adequate to support continuous mobility management. This is exposed in our study where we witness that the handoff performance be subject to profoundly on link layer FER, the deferral among different network units that are intricate in the handoff, and the wireless admittance technology. Therefore, we believe information distribution among diverse layers to enhance the concert of mobility management. This cross-layering method will remove the undesirable properties of altered constraints such as link layer frame error rate and signalling deferral on the handoff enactment of mobility managing protocols.

Yu-Chee Tseng [10] as portable devices are attaining more recognition, maintaining Internet connectivity anytime and anywhere becomes crucial, mainly for mobile and also the vehicular networks. Network mobility (NEMO) and also the Internet Protocol mobility are achieving more and more significance. In their paper, they develop Session Initiation Protocol (SIP)-based mobile network architecture to hold up NEMO for vehicular applications. They proposed to outline a mobile ad hoc network (MANET) by the mobile hosts (MHs) in a vehicle or a group of vehicles. The MANET is linked to the outside world through a SIP-based Mobile Network Gateway (SIP-MNG), which is equipped with one or numerous external wireless interfaces and a quantity of inner IEEE 802.11 interfaces. Then the external interfaces of the SIP-MNG hold up Internet connectivity by combining user traffic to and from the Internet. In adding up, utilizing the session information carried by SIP signalling, the SIP-MNG supports resource management and call admission control for the MHs. Though, wireless access earns charges, power consumption, and overhead of mobility management. As a result, it was enviable to allow the SIP-MNG to detach its external interfaces when required. To warranty that users inside the mobile network will not lose any arriving request, they offer a push mechanism through short message service to stir up these wireless interfaces in an on-demand manner. They exemplify the complete signalling to bear such a mechanism. The proposed system is fully well-matched with existing SIP standards. Their real prototyping skill and some experimental results are also reported.

Their paper [10] has proposed SIP-based mobile network architecture to support networking services on the roads. With multiple wireless interfaces, a SIP-MNG can offer dynamic bandwidth to internal users based on their bandwidth requirements. In addition, by permitting multiple sessions to share one interface, their system can help public transportation operators’ users to save Internet access fees. Moreover, through their system, vehicles can provide Internet access to passengers with support of cluster movement. By understanding SIP signaling, our CAC and RM mechanisms within the SIP-MNG can guarantee QoS for users. In addition, through SMS and session control, they have recommended a push mechanism to permit the SIPMNG to stay offline when there is no calling activity and to be "woken up" when necessary.

Their [10] push approach can protect call charges and energy while preserving worldwide reachability of users. In their architecture, they do not modify the existing SIP client–server protocol architecture. A prototype had been developed, and a number of experimental results had been offered. For IEEE 802.11, WCDMA, and PHS networks, they demonstrated that it is feasible to allow multiple stations to share one interface. It is also shown that, by cellular interfaces, the call setup time and handoff delay are longer than that by 802.11 interfaces, because connecting to the Internet via a cellular interface had to go through more networks. For their push mechanism, based on the present technologies, the call system time is in the range of 20 s, which is somewhat long. So, they have designed the push server to temporarily pick up the session and put on the REFER scheme to transfer the session to the user within the SIP-MNG. The wireless interface re-joining time takes longest. Even though that was not in the scope of their work, they believe that a lot of research outcomes can help decrease the reconnection periods.

Stefano Salsano et al., [11] described that the ITU-T definition of next generation networks includes the capability to make use of multiple broadband transport technologies and to maintenance general mobility. Next generation networks must incorporate several IP-based access technologies in a seamless way. In their article, they first describe the necessities of a mobility management system for multimedia real-time communication services; then, they reported a study of the mobility management schemes proposed in their recent literature to perform vertical handovers between diverse networks. Based on that analysis, they proposed an application-layer solution for mobility management that is based on the SIP protocol and that satisfies the most important desires for a proper implementation of vertical handovers. They also implemented their proposed solution, testing that in the field, and demonstrating its overall feasibility and its interoperability with different SIP servers and terminals. SIP provides well-designed application-layer mobility support that resolves the difficulties related with lower-layer mobility protocols in next-generation heterogeneous wireless access networks. Though, the handoff delay in SIP may be considerable, thus causing substantial packet loss, which extremely disturbs the excellence of video or voice streams. In order to relieve the problem of packet loss, in their article they have offered a SIP-based mobility architecture for soft handoff in next-generation wireless networks. A test bed has been set up to measure the effectiveness of the suggested architecture. The experimental outcomes display that the architecture is proficient of ensuring zero packet loss and controlled delay jitter.

Beheet Sarikaya et al., [12] introduces a new paging technique to track and awaken a mobile node (MN) attached to an access point (AP) in a wireless LAN network after a SIP INVITE message is started by a caller. A tracking agent (TA) keeps track of the mobiles’ handoffs among the APs. A paging agent (PA) activates the TA to page the mobile once a SIP INVITE is acknowledged for one of its users. The context transference feature of their paging protocol permits the paging messages to supply the station context in order to allow quicker session restoration. The AP then does on-link paging in the wireless link. SIP extensions are necessary to trigger the PA to start paging MNs to inform their inactive status using an extended SIP REGISTER technique. Tracking protocol is investigated to associate and hard- and soft-state methods for message rate, state inconsistency ratio and the overall cost. The simulation model they developed enables us to evaluate the traffic introduced by the tracking protocol and the cache (state) size. Paging protocol is investigated for the transmission delays and CPU processing times in the SIP session setup by means of paging. Replication of the paging through context transport is used to show the gains in re-authentication.

They have [12] introduced a new approach called SIP paging and tracking. SIP pagings are done with the PA component to attentive a latent MN after a SIP INVITE message was acknowledged. SIP pursuing is done with a TA at the SIP layer that obtains position updates from the ARs based on relations made to the APs. Paging is started through the PA, and the TA express the paging demand to the AP to which the mobile was related last. The AP practices on-link paging to awaken the mobile. MNs are required to register their latent mode status with the SIP proxy server using the extensions they introduced. The PA converses with the SIP proxy using a new SIP message. They examined two forms of the tracking protocol, HS and SS for state inconsistencies, signalling amounts and the overall cost. Outcomes show how the factors of the SS protocol need to be personalised so that the cost is lesser than the HS protocol. The results illustrate that if the packet error rates are lesser, greater values can be selected for SS refresh timer. They simulated the SS tracking protocol to determine paging cache size and the tracking traffic rate. They determined that their centralized TA architecture provides improved performance results than the replicated TA architecture. SIP paging is investigated in order to determine the processing load at the SIP proxy. Paging delay is analysed and the delay is exposed to increase with the packet error rate and the number of hops among the APs and the TA and influences the delay strongly. SIP paging transmission delay analysis can be extended assuming TCP transport for the messages. Duplicating the Pas and TAs has to be considered for fault accepting operation of SIP paging. Implementation of SIP paging and tracking protocols are also left as future work. Consideration of cellular on-link paging techniques such as UMTS and their integration with SIP paging needs further examination.

Shun-Ren Yang et al., [13] the Universal Mobile Telecommunications System (UMTS) all Internet-Protocol (IP) network supports IP multimedia services during the IP Multimedia Subsystem (IMS). Their paper proposed a mobile quality-of-service (QoS) framework for varied heterogeneous IMS interworking. To decrease the handoff interruption time, the following frameworks support the IMS mobility based on the idea of Session Initiation Protocol (SIP) multicast. In their approach, the mobility of a User Equipment (UE) is modelled as a transition in the multicast group membership. By the thought of dynamic shifting of the multicast group’s members, the stream of real data packets can be switched to the new route as hurriedly as possible. To overcome mobility impact on service guarantees, UEs want to make QoS resource reservations in progress at neighbouring IMS networks, where they might trip through the lifetime of the ongoing sessions. These places become the leaves of the multicast tree in their approach. To attain more efficient use of the limited wireless bandwidth, their approach allows UEs for the short term exploit the inactive bandwidths booked by other UEs in the present IMS/ access network. Analytic model and simulation model were developed to inspect our resource reservation scheme. The results point out that scheme yields similar performance to that of the previously planned channel assignment schemes.

Their paper [13] proposed a mobile QoS framework for heterogeneous IMS interworking. To decrease the handoff disruption time, that framework hold up the IMS mobility based on the idea of SIP multicast. In their approach, the mobility of a UE is modeled as a transition in the multicast group membership. Even though this method needs a lot more background processing and use of bandwidth on wired links, they can remove the need for rerouting the data path during handoffs along with IMS networks. They developed an analytic model for their proposed resource reservation scheme. The above mentioned analytic model vary from the active mobile network handoff models owing to the introduction of PR and TR more to the point the traditional CR reservation. The analytic approach was authorized against the simulation model. Based on the simulation experiments, they examined the po, pf , and pnc performance of the resource reservation algorithm.

Fang-Yie Leu [14] in a varied heterogeneous wireless environment, flawless mobility is the base of network hold up with which mobile users who wander between or along with various wireless access networks are able to completely enjoy uninterrupted wireless services. While users are in a collection of transportation vehicle, e.g., a bus or a train that supply network service, the vehicle can be viewed as a network which is helping users as it moves from one spot to another. The movement of a network is called network mobility (NEMO). Based on the network mobility protocol, Mobile IPv6 as planned by the Internet Engineering Task Force (IETF) in 2005 has some fundamental disadvantage, such as header overhead and the pinball problem.

In their paper [14], they proposed a novel hybrid method for network mobility called Hybrid-NEMO, which provides a soft handoff system at the transport layer basically using SIP and SCTP protocols to ensure a lossless packet-transmission environment and less handoff-delay variation, which are dangerous in providing QoS voice and multimedia applications. Performance evaluation and Experimental validation performance evaluation were also conducted in this study. Seamless mobility supports are the basis for providing uninterrupted wireless services to mobile users roaming between or among various wireless access networks. However, seamless mobility approaches that have been proposed have their individual problems, e.g., relatively longer handoff delay sand/or higher packet-loss rates. The fundamental problems of network mobility protocols based on Mobile IPv6 are header overhead and pinball problem. OptiNets RO and SIP-NEMO have been proposed to improve on MIPv6-NEMO, and they actually can eliminate some of the problems that arise with MIPv6-NEMO (Cho et al., 2006). But packet loss during handoff still exists, which severely affects the quality of voice and video streams because a hard handoff mechanism is deployed. In order to alleviate the problem, in their paper they proposed a novel network mobility architecture that exploits the multi-homing characteristic of the SCTP and deploys a soft handoff scheme for next-generation wireless networks. However, an SCTP-CMS needs more egress interfaces than other central management servers which apply other NEMO schemes. In other words, the power consumption of a central management server using Hybrid-NEMO is greater than that of other three NEMO schemes. In their paper, they also described in detail node, network and nested handoffs. Simulations were performed to evaluate the efficiency of the architecture. The experimental results show that the architecture was capable of ensuring zero packet loss and stabilizing delay jitters, particularly when the transmission error rate=0. Basically, the mobility issue was two-fold, involving a handoff decision and a handoff scheme. Although the former was not addressed in this study, in the future, we will deal with handoff decisions, security issues and apply the concurrent multi-path.

Franco Callegati et al., [15] presented novel application aware network architecture for evolving and emerging IT services and applications. It was proposed to enrich an optical burst switching network with a session control layer that can close the gap between network control and application requests. The session control layer was implemented using the Session Initiation Protocol, giving birth to SIP-OBS architecture. Their article discussed the important added value of that architecture, and showed that it may support a number of end-to-end resource discovery and reservation strategies (for both non-network and network resources). Finally, it presented a testbed implementation where this approach was experimentally validated. This article describes a possible extension to the conventional control plane of an OBS network with functionalities that makes it able to communicate with end-user applications and understand their needs. This has been achieved by introducing a session control layer implemented by SIP on top of the conventional network control plane. The proposed architecture was applied as part of a fully functional OBS testbed, and its possibility was proven in a series of experiments culminating in running a fully functional multimedia application.

Doris Bao et al., [16] proposed a SIP automatic debugger tool. That is a software instrument that will be used to verify the compliance of Voice over Internet Protocol (VoIP) devices, such as VoIP gateways and soft phones to the SIP specifications, and then to test the interoperability of VoIP equipment produced by different manufacturers. Different tools are existing in the market to conduct a interoperability and compliance. However, they often had features limited to packet capturing and decoding, or they are simulation tools which often require a complex developing phase to define the behavior of each test. The proposed tool, can be inserted into an SIP network and are capable of analyzing and observing in an spontaneous way, the communication steps. It was operated by executing three subsequent phases. In the first phase, the SIP messages that are flowing in the network are captured. Then, second phases are in charge of grouping SIP messages into dialog and transactions. Finally, a third phase operated by comparing the message flow with a set of predefined rules. Rules have been classified into two groups. Rules belonging to the first group, are called static rules that have been obtained by the direct analysis of SIP specifications. Rules belonging to the second group are called dynamic rules that have been obtained by experience with SIP compliance and interoperability testing (with the support of SIP testing specialists). When verification of a number of rules is unsuccessful, an output is described by representing the rule that failed and a list of possible fault reasons. The tool had been validated in a laboratory network in different scenarios. Some sample test cases, which had been extracted by these scenarios and show the capability of the SIP automatic debugger tool in finding compliance and interoperability faults, are also presented in their paper.

In their paper [16], an SIP automatic debugger tool designed to be used in the test and validation process of SIP devices has been presented. The tool allowed the analysis of SIP conversations (offering additional features for the automatic detection of faults) that occurred during the communication. These features provided significant advantages in terms of optimizing the effort involved in SIP product testing. Independent from the operator’s skill, the SIP automatic debugger tool makes it easier to analyze the test results, to spot and fix crucial bugs, and to examine the generated reports. Some scenarios of SIP calls were presented to show how the proposed tool was able to find faults in the interoperability between SIP devices and verify the compliance to the standard. Further work will proceed in the direction of the selection of a complete test suite, which will allow the verification of compliance to RFC 3261 and all related SIP specifications. In addition, the core of the tool can be improved to allow easier definition of the dynamic rules, which could directly be added by the user at runtime.

Lin Liu et al., [17] describe that in recent years Voice over Internet Protocol (VoIP) has become a popular multimedia application over the Internet. At the same time serious safety issues in VoIP have started to emerge. The SIP is a predominant signalling protocol for VoIP. It is used to establish, preserve and terminate VoIP calls, playing a crucial role in VoIP. Their paper is aimed at developing a Coloured Petri Net (CPN)-based approach to analysing security susceptibility in SIP; with the ultimate goal of achieving a formal and comprehensive security assessment of SIP specification, and producing a platform for evaluating counter measures for securing SIP.

In their paper [17] they presented a method for modelling the behaviour of SIP and their safety threats using CPNs, and discussed suitable techniques for analysing the CPNs for SIP security issues and investigation. The CPN models and the analysis techniques will then become the platform for analysing the behavior of SIP that is enhanced with proposed security countermeasures.

Swetkecher et al.., [18] presented a concept for sip, this work influences SIP transference and mobility mechanism to transmission session data among two Web browsers. In addition, a Web browser can currently act as an adaptive User Agent Client to search the Internet and make voice requests as a SIP client. It is a new work that uses SIP to transfer session data among Web browsers and borrows SIP Mobility kinds to introduce new facility namely, content sharing and session hand-off, to the Web browsing practice. Stated to as a SIP-based HTTP session mobility service, it deals personal mobility to end users, and simplifies session mobility in Web browsing. Whereas content sharing refers to the capability to sight the same Web resource on two Web browsers and does not need moving session data, session hand-off denotes to the migration of a Web session with its session facts (cookies, hidden form features and rewritten URL) to extra Web browser. Results displayed that the combination of SIP into a Web browser does not reduce the concert of a Web browser. Consequences also exhibited that the service could not work on all websites as the Same Origin Policy (SOP) used by Web browsers to handover cookies. The hybrid-based architectural scheme suggested and implemented here is matched with other existing Web session relocation schemes. On the service commercialization, if the confidentiality and security of session data could be definite by the implementers, a flat rate could be occasionally charged irrespective of the varying session data dimensions. In another sense, it could be concentrated as a Value Added Service (VAS) to clients. This work helps advance collaboration and mobility between the Web users. It also inspires adaptive UACs. In this instance, the Web browser can also be used as a SIP client to make voice requests and extensible to make other functionalities. On the service commercialization, should there be a need to install this service in a client-server location; a flat rate could be charged occasionally irrespective of the varying session data extents. On the other hand, it could be rendered as a Value Added Service (VAS) to clients. It is highly advised that an S/MIME-supported SIP stack is used over and done with the implementation to confirm confidentiality and security of session data over a big network. A problem was still encountered during the HTTP session mobility check. The exact Web browser’s state could not be replicated when session handoff was passed out on Mash-ups, AJAX-based and FRAME/IFRAME-based Websites. A credible solution to this tricky is to capture the Web browser’s running state at the basis UAC and replicate at the destination UAC. This key is highly technical but could be attained by the developers of Web browsers. Other areas that could be discovered include implementing a plan control to block undesirable Web session transfer demand. Such limitation could be centered on a domain name or a SIP address. Whereas a session-based cookie perishes in a short time of idleness, a persistent cookie can deliver access to a Website over a extensive period. A session management mechanism could also be combined so that a Web session handover request can be detained for a long time without deceasing when the endpoint SIP address cannot be prolonged or a Web server usages a session-based cookie that expires in a small time. These policy control and session managing functionalities could be advanced and combined into a SIP Application Server (AS). The SIP AS could be arranged on the IP Multimedia Subsystem (IMS) thus making the facility available on the IMS. An address book of links could also be combined so that two or additional people can be complicated in a content distribution process.

Arslan Munir et al., [19] proposed that the third-generation partnership project (3GPP) and 3GPP2 have standardized the IP multimedia subsystem (IMS) to provide ubiquitous and access network-independent IP-based services for next-generation networks via merging the Internet and cellular networks. standardizing the application layer Session Initiation Protocol (SIP), by 3GPP and 3GPP2 for IMS, was responsible for IMS session establishment, transformation and management. The IEEE 802.16 worldwide interoperability for microwave access (WiMax) assured to provide high data rate broadband wireless access services. In their paper, they proposed two novel interworking architectures to combine WiMax and third-generation (3G) networks. Moreover, they analyzed the SIP-based IMS registration and session setup signaling delay for 3G and WiMax networks with specific reference to their interworking architectures. Finally, they explore the effects of different WiMax-3G interworking architectures on the IMS registration and session setup signaling delay.

In their paper [19], they analyzed the SIP-based IMS registration and session setup signaling delay in 3G and WiMax access networks. They also analyzed the effects of novel WiMax-3G interworking architectures on the IMS signaling delay. Their numerical analysis revealed that the tightly coupled architectures have lower IMS signaling delays than loosely coupled architectures. It can be concluded that a tightly coupled system was more appropriate for restricting the IMS signaling delays to acceptable limits. However, deployment of the tightly coupled architecture requires more effort than the deployment of loosely coupled architecture, and hence, a definite trade-off exists between performance efficiency and implementation cost. Numerical data analysis indicated that the IMS registration and session setup signaling delay in WiMax networks was much less than the IMS registration and session setup signaling delay in 3G networks. Their numerical results encourage the deployment of WiMax-3G interworking architectures with the IMS infrastructure support.

Zong-Hua Liu et al., [20] proposed that the Mobile Virtual Private Network (MVPN) had developed to secure mobile user’s communication between untrusted external networks and protected internal private network. The IETF’s solution, however, leads to some problems like where to put the external Home Agent (HA), how the external HA can be trusted, and overhead of three tunnels.

In this paper [20], they proposed an alternative SIP-based MVPN. The problems inherited from Mobile IP were not problems in their architecture, because the proposed architecture is based on SIP. There is no need to tunnel a packet three times as that in IETF MVPN. In this paper, they also proposed analytical models to evaluate and compare the performance of the proposed SIP-based MVPN with the IETF MVPN. The results showed that the proposed SIP-based MVPN can reduce packet delivery cost knowingly. It was especially suitable for real-time applications.

Rosario G. Garroppo et al., [21] projected that the overload was a challenging problem for a SIP server because the built-in overload control mechanism based on generating rejection messages could not prevent the server from collapsing due to congestion. In this scenario, their paper presents an overload mechanism combining a remote and a local solution. The local part of the overload control mechanism was based on the appropriate queueing structure and buffer management of the SIP proxy. The remote overload control mechanism was based on feedback reports provided to the upstream neighbors by the SIP proxy. These reports permit the traffic regulation necessary to avoid the critical condition of the overload. Their paper's main contributions were the design of key components of a remote control mechanism, the use of a prediction technique in the remote control loop and the proposal of a new approach for dynamic load estimation.

Their paper [21] investigated the problem of SIP server overload, by combining local and remote overload control mechanisms. The main contribution is about the remote overload control mechanism, while local one has been discussed in a previous work. The remote overload control mechanism is based on a feedback loop from the receiving entity to the sending entities. On the receiving entity there is a controller that estimates the available resources based on a dynamic load estimation algorithm. Then, a feedback value is calculated. This feedback value is reported to the sending entities to obtain traffic regulation. Their paper has presented a novel proxy load estimation method called Transaction approach. This method is based on a novel algorithm for the available resources estimation that takes into account the number of active state machines. The second novel contribution of the paper was the enhancement of Transaction approach with a prediction algorithm to obtain a faster reaction to traffic increasing. The simulation results point out that the Transaction approach with prediction based resource allocation, leads to the best performance in terms of voice sessions successfully established.

Yujiao Wang et al., [22] presented a concept for sip, the elementary content of the session origination protocol and its protocol's distinctive are analyzed in the paper, joining the model of expanse education system, a distance education scheme based on SIP protocol is suggested, and then presents a model of method and gives a brief explanation of each module of the model, A detailed description about the application of each model is given.

With the firm improvement of multimedia technology and network communication technology, SIP is certain to develop the most significant multimedia session control protocol in the following generation network. With modest, malleable, extensible, and attainable features, SIP protocol can simply meet the strains of multimedia session in contemporary distance education, setting communications among learners and tutors free from spatial limitations. The technology can significantly increase learners’ proficiency, providing new chances for the progress of distance learning in China.

Ivan Vidal et al., [23] presented a concept for sip, in latest years, the progress and placement of new wired and wireless admittance network technologies have complete the ubiquitous Internet realism. Workers can admittance anyplace and anytime to the wide-ranging set of value-added Internet facilities, which are transported by resources of the IP protocol. In this framework, 3GPP is presently developing the IP Multimedia Subsystem (IMS), as a significant part that agrees to progress from the pervasive access to the Internet services to a next generation network classical, by providing a set of vital amenities such as session control, QoS, accusing and service combination. Nevertheless, numerous open matters still need deliberation before the forthcoming Internet becomes real, such as supportive user mobility in IP networks. Though mobility upkeep in the Internet is getting much attention, IMS networks existing intrinsic accuracies that require further scrutiny. The solutions suggested so far for IMS do not sustenance mobility evidently to the end-user applications, or address the problematic by familiarizing multifaceted changes to the IMS substructure. This paper benevolences TRIM, an architecture for translucent IMS-based mobility. TRIM maintenances mobility in IMS networks obviously to the end-user applications, which are uninformed of the assignment management actions executed among the mobile node and the network. We have achieved several experimentations with a TRIM prototype, using a factual IMS test bed with 3G and WLAN admittance networks, legalizing the tender for UDP and TCP centered applications.

This paper [23] proposes TRIM, design to deliver mobility sustenance in IMS-based networks. TRIM is centered on SIP signaling, but contrasting other methods to offer mobility sustenance in IP networks scentered on SIP, TRIM marks mobility translucent to applications, which do not need to do any action to support an alteration of admittance network by the mobile node. This functionality is analogous to the one providing by Mobile IP, but the compatibility of Mobile IP and IMS necessitates alterations to the IMS provisions, a prerequisite that TRIM does not have. We have verified a prototype of TRIM in a test bed merging 3G and IEEE 802.11 admittance technologies, and an IMS essential. In the experimesntations they uncovered that TRIM appropriately maintenances mobility both for UDP and TCP user traffic. There are numerous lines of upcoming research for TRIM. TRIM onwards the user traffic over a network part in the mobile node home network, which is parallel to Mobile IP deprived of route optimization. We need to explore an optimization to abide this network part in or close to the network existence stayed by the mobile node. In addition they would like to learn the performance of handovers while only one network boundary is accessible in the mobile node and soft deliveries are not conceivable. A method to improve concert could be to raise optimizations centered on buffering in the midway network element.

Vijay K.Gurbani et al., [24] to encrypt the media stream in the SIP in order to Exchange cryptographic keys had demonstrated complicated. To exchange keys successfully was the test at the same time as preserving the features of the protocol (e.g., forking, re-targeting, request recursion, etc.), reducing key exposure to unintentional parties, eliminating voice clipping, maintaining end-to-end key privacy and through interfacing with PSTN. In their paper, they survey about three key management protocols SDES, ZRTP and DTLS-SRTP. These three protocols had been proposed for media keying, and estimate them for exercise with SIP. To support in the valuation, they first extract a interior feature set from SIP. Then they survey every key management protocol in detailed manner. They also carry on analyzing the interior of the three protocols in opposition to this feature set to explain their weaknesses and strong point.

They had evaluated three media key management protocols SDES, ZRTP, and DTLS-SRTP namely in SIP against a core characteristic set to determine their suitability. A summary of their evaluation was also given. Their analysis suggested that the key management protocols that function in the media layer are really right media keying protocols although their operational differences. To identify the users and derive media keying material DTLSSRTP was used, which was a certificate-based system while ZRTP uses a mixture of a short authentication string and key continuity to verify the users and a DH exchange to obtain media keying material. They draw reader’s interest to restrain MiTM hit possibilities on ZRTP. The SDES key management protocol, which functions in the signalling plane, had natural support for certified interception, which was too hard to accomplish with DTLS-SRTP and ZRTP. It is possibly appealing to speculate whether an entirely new protocol can be planned to meet the needs of SIP media keying. Such a protocol should, out of requirement, stability providing security guarantees while sustaining the legacy characteristics of the SIP protocol. But devising such a novel protocol is not a panacea; for one, budding a new key exchange protocol with provably protected properties is a non-trivial assignment, and more crucially, it is not the case that the current media keying protocols fall short in provided that security assurances — TLS and its derivatives are safe if one thinks a public-key infrastructure. Without such an infrastructure, ZRTP make use of a SAS to validate the validity of the communicating endpoints. In addition, these protocols had been successfully used to key SIP media stream.

Vittorio Ghini et al., [25] presented a distributed architecture for the provision of flawless and receptive mobile multimedia services. The above revealed architecture permit its user applications to utilize simultaneously all the wireless network interface cards (NICs) a mobile terminal which is capable of doing. In particular, as mobile multimedia services are usually applied or implemented using the UDP protocol, their architecture facilitates the broadcast of every UDP datagram in the course of the "most suitable" (e.g. most responsive, slightly loaded) NIC among those available at the time a datagram is transmitted. They term this operating mode of their architecture Always Best Packet Switching (ABPS). ABPS permit the use of policies for load balancing and recovery purposes. In essence, the architecture that they propose consists of the following two principal components: (i) a fixed proxy server, which take action as a relay for the mobile node and make possible communications from/to this node not considering of possible firewalls and NAT systems, and (ii) a proxy client running in the mobile node in charge for maintaining a multi-path tunnel, created out of all the node’s NICs, as mentioned above fixed proxy server. They show how the architecture supports multimedia applications based on the SIP and RTP/RTCP protocols, and keep away from the typical hindrance introduced by the two way message/response handshake of the SIP signaling protocol. Tentative results instigated from the implementation of a VoIP application on top of the architecture. They suggested to show the efficiency of their approach.

In their paper [25] they have offered a distributed architecture for the provision of flawless and responsive mobile multimedia services. All the MN’s NICs can be simultaneously subjugated, in the sense that each datagram is transmitted through the most suitable NIC in use at the time, based on the ABPS operating mode. The architecture is based on the use of a proxy client which is installed on the MN and a fixed proxy server that communicates with the client. The provided experimental results demonstrated the effectiveness of their solution. Their approach optimized the interaction/communication between a MN and its