The implementation of the Session Initiation Protocol (SIP)-based Voice over Internet Protocol (VoIP) and multimedia over MANET is still a challenging issue. Many routing factors affect the performance of SIP signaling and the voice Quality of Service (QoS). Node mobility in MANET causes dynamic changes to route calculations, topology, hop numbers, and the connectivity status between the correspondent nodes. SIP-based VoIP depends on the caller's registration, call initiation, and call

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... on processes. Therefore, the SIP signaling performance has an important role for the overall QoS of SIP-based VoIP applications for both IPv4 and IPv6 MANET. Different methods have been proposed to evaluate and benchmark the performance of the SIP signaling system. However, the efficiency of these methods vary and depend on the identified performance metrics and the implementation platforms. This survey examines the implementation of the SIP signaling system for VoIP applications over MANET and highlights the available performance enhancement methods. Keywords-SIP; VoIP; MANET; Peer-to-Peer; Back-to-Back User Agent (B2BUA); IMS I. This review is focused on research in SIP signaling over MANET and the performance enhancement approaches for SIP-based VoIP applications. In this paper, the current stateof-the-art, results, gaps, the merits and demerits of the four types of SIP signaling systems over MANET mentioned here and the performance enhancement methods for SIP signaling over MANET are discussed in detail. Finally, two open issues have been identified and highlighted for future investigations. A. SIP Signaling System SIP is an Internet Engineering Task Force (IETF) standard for signaling protocol released as RFC 3261 [1]. SIP is commonly used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP). SIP is used in initiating, managing, and terminating multimedia sessions such as voice calls over IP based networks. This session can be either a two-way call, which is either unicast or collective multimedia calls, which is multicast. These features have made SIP a better choice for providing VoIP services in the last few years. SIP is an application layer protocol, which serves five main functions for multimedia calls [1]. These functions are: User Location, User Availability, User Capability, Session Setup, and Session Management. User Location is used to determine the location of the end user, while User Availability examines the willingness of the end user to participate in the call session. User Capability supports the applications compatibility with different communication systems and users to determine the required methods and standards for the requested multimedia 192 | P a g e www.ijacsa.thesai.org applications. Session Setup provides the resources to setup and establish the communication. Finally, the Session Management function supports the call management services in different ways such as adding, transferring and modifying the session parameters.