Analisis Performansi SIP (Session Initiation Protocol) Over SCTP (Stream Control Transport Protocol) dengan OPENSIPS untuk Komunikasi Multi-User

ABSTRAKSI: Dalam beberapa tahun terakhir, Session Initiation Protocol (SIP) yang dikembangkan oleh Internet Engineering Task Force (IETF) telah mendapatkan popularitas yang signifikan di arena Voiceover-IP (VoIP) dan bersaing dengan Internet Multimedia protokol H.323. SIP juga dipilih oleh Generation Partnership Project Ketiga (3GPP) sebagai protokol standar untuk sinyal kendali pelayanan dalam Generasi Ketiga (3G) jaringan nirkabel.SIP adalah protokol kontrol komunikasi mampu berjalan pada lapisan transportasi yang berbeda, misalnya, Transportasi Control Protocol (TCP), User Datagram Protocol (UDP), atau Streaming Transmission Control Protocol (SCTP). Saat ini sebagian besar aplikasi SIP beroperasi melalui protokol transport UDP yang bersifat connectionless dan unreliable. Dalam lingkungan lossy seperti jaringan nirkabel dan jaringan internet padat, masalah yang dihadapi antara lain pesan SIP dapat hilang atau terkirim keluar dari urutan (delivered out of sequence). Aplikasi SIP kemudian harus memancarkan kembali (retransmit) pesan yang hilang dan kembali mengurutkan paket yang diterima (reordered). Pemrosesan tambahan ini menyebabkan overhead yang dapat menurunkan kinerja dari aplikasi SIP. SCTP, yaitu sebuah protokol transport yang menyediakan acknowledge, error-free, transfer non-duplikasi pesan, yang diusulkan untuk menjadi alternatif untuk UDP dan TCP. Fitur multi-streaming dan multi-homing SCTP sangat menarik untuk aplikasi yang memiliki kinerja yang ketat dan persyaratan keandalan yang tinggi.Pada tugas akhir ini didapatkan pengukuran untuk mengetahui hasil perbandingan performansi dari transport protocol UDP dan SCTP yang diukur dari delay yang keduanya menunjukkan hasil yang tidak jauh berbeda,masih sesuai dengan batas-batas yang ditetapkan oleh ITU yaitu sekitar 150-400 ms. Pengukuran jitter menunjukkan keduanya memiliki performansi yang melebihi standar dari CISCO yaitu kurang dari 30 ms, diakibatkan adanya buffer jitter pada saat pengujian. Dan sedangkan saat diukur dengan parameter packet loss menunjukkan bahwa SCTP lebih unggul dalam hal menangani packet loss sekitar 10-20%.Kata Kunci : VoIP, SIP,UDP, SCTP, multistreaming,delay,jitter,packet loss.ABSTRACT: In recent years, Session Initiation Protocol (SIP) developed by the Internet Engineering Task Force (IETF) has gained significant popularity in the Voiceover- IP (VoIP) arena and is competing with the Internet Multimedia protocol H.323. SIP is also selected by Third Generation Partnership Project (3GPP) as a standard signaling protocol for service control in Third Generation (3G) wireless network.SIP is a communication control protocol capable of running on different transport layers, e.g., Transport Control Protocol (TCP), User Datagram Protocol (UDP), or Stream Control Transmission Protocol (SCTP). Today’s SIP application is mostly operating over the unreliable transport protocol UDP. In lossy environment such as wireless networks and congested Internet networks, SIP messages can be lost or delivered out of sequence. The SIP application then has to retransmit the lost messages and re-order the received packets. This additional processing overhead can degrade the performance of the SIP application. To solve this problem, researchers are looking for a more suitable transport layer for SIP.SCTP, a transport protocol providing acknowledged, error-free, non-duplicated transfer of messages, has been proposed to be an alternative to UDP and TCP. The multi-streaming and multi-homing features of SCTP are especially attractive for applications that have stringent performance and high reliability requirements.In this final measurements to determine the results obtained performance comparison of UDP and SCTP transport protocol is measured from the delay which both showed results that are not much different, still in accordance with the limits set by the ITU at around 150-400 ms. Jitter measurements show they have a performance that exceeds the standards of CISCO which is less than 30 ms, caused the jitter buffer at the time of testing. And while the current is measured by the packet loss parameter indicates that the SCTP is superior in terms of dealing with packet loss of about 10-20%.Keyword: VoIP, SIP,UDP, SCTP, multistreaming, delay,jitter,packet loss

Signalling Transport Protocols

SCTP is a newly developed transport protocol tailored for signalling transport. Whereas in theory SCTP is supposed to achieve a much better performance than TCP and UDP, at present there are no experimental results showing SCTP s real benefits. This paper analyzes SCTP s strengths and weaknesses and provides simulation results. We implemented SIP on top of UDP, TCP and SCTP in the network simulator and compared the three transport protocols under different network conditions

Signalling Transport Protocols

SCTP is a newly developed transport protocol tailored for signalling transport. Whereas in theory SCTP is supposed to achieve a much better performance than TCP and UDP, at present there are no experimental results showing SCTP s real benefits. This paper analyzes SCTP s strengths and weaknesses and provides simulation results. We implemented SIP on top of UDP, TCP and SCTP in the network simulator and compared the three transport protocols under different network conditions

The Impact of SCTP on Server Scalability and Performance

The Stream Control Transmission Protocol (SCTP) is a newer transport protocol, having additional features to TCP. Although SCTP is an alternative transport protocol for the Session Initiation Protocol (SIP), we do not know how SCTP features influence SIP server scalability and performance. To estimate this, we measured the scalability and performance of two servers, an echo server and a simplified SIP server on Linux, comparing to TCP. Our measurements found that using SCTP does not significantly affect on data latency: approximately 0.3 ms longer for the handshake than that for TCP. However, server scalability in terms of the number of sustainable associations drops to 17-21%, or to 43% of TCP if we adjust the acceptable gap size of unordered data delivery

Delay-centric handover in SCTP

The introduction of the Stream Control Transmission Protocol (SCTP) has opened the possibility of a mobile aware transport protocol. The multihoming feature of SCTP negates the need for a solution such as Mobile IP and, as SCTP is a transport layer protocol, it adds no complexity to the network. Utilizing the handover procedure of SCTP, the large bandwidth of WLAN can be exploited whilst in the coverage of a hotspot, and still retain the 3G connection for when the user roams out of the hotspot’s range. All this functionality is provided at the transport layer and is transparent to the end user, something that is still important in non-mobile-aware legacy applications. However, there is one drawback to this scenario - the current handover scheme implemented in SCTP is failure-centric in nature. Handover is only performed in the presence of primary destination address failure. This dissertation proposes a new scheme for performing handover using SCTP. The handover scheme being proposed employs an aggressive polling of all destination addresses within an individual SCTP association in order to determine the round trip delay to each of these addresses. It then performs handover based on these measured path delays. This delay-centric approach does not incur the penalty associated with the current failover-based scheme, namely a number of timeouts before handover is performed. In some cases the proposed scheme can actually preempt the path failure, and perform handover before it occurs. The proposed scheme has been evaluated through simulation, emulation, and within the context of a wireless environment

IP-Based Mobility Management and Handover Latency Measurement in heterogeneous environments

One serious concern in the ubiquitous networks is the seamless vertical handover management between different wireless technologies. To meet this challenge, many standardization organizations proposed different protocols at different layers of the protocol stack. The Internet Engineering Task Force (IETF) has different groups working on mobility at IP level in order to enhance mobile IPv4 and mobile IPv6 with different variants: HMIPv6 (Hierarchical Mobile IPv6), FMIPv6 (Fast Mobile IPv6) and PMIPv6 (Proxy Mobile IPv6) for seamless handover. Moreover, the IEEE 802.21 standard provides another framework for seamless handover. The 3GPP standard provides the Access Network and Selection Function (ANDSF) to support seamless handover between 3GPP – non 3GPP networks like Wi-Fi, considered as untrusted, and WIMAX considered as trusted networks. In this paper, we present an in-depth analysis of seamless vertical handover protocols and a handover latency comparison of the main mobility management approaches in the literature. The comparison shows the advantages and drawbacks of every mechanism in order to facilitate the adoption of the convenient one for vertical handover within Next Generation Network (NGN) environments. Keywords: Seamless vertical handover, mobility management protocols, IEEE 802.21 MIH, handover latenc

Taxonomy and analysis of IP micro-mobility protocols in single and simultaneous movements scenarios

The micro-mobility is an important aspect in mobile communications, where the applications are anywhere and used anytime. One of the problems of micro-mobility is the hand-off latency. In this paper, we analyse two solutions for IP micro-mobility by means of a general taxonomy. The first one is based on the Stream Control Transmission Protocol (SCTP), which allows the dynamic address configuration of an association. The second one is based on the Session Initiation Protocol (SIP), which is the most popular protocol for multimedia communications over IP networks. We show that for the SCTP solution, there is room for further optimisations of the hand-off latency by adding slight changes to the protocol. However, as full end-to-end solution, SCTP is not able to handle simultaneous movement of hosts, whose probability in general cannot be neglected. On the other hand, the SIP can handle both single and simultaneous movements cases, although the hand-off latency can increase with respect to the SCTP solution. We show that for a correct and fast hand-off, the SIP server should be statefull

Roaming Real-Time Applications - Mobility Services in IPv6 Networks

Emerging mobility standards within the next generation Internet Protocol, IPv6, promise to continuously operate devices roaming between IP networks. Associated with the paradigm of ubiquitous computing and communication, network technology is on the spot to deliver voice and videoconferencing as a standard internet solution. However, current roaming procedures are too slow, to remain seamless for real-time applications. Multicast mobility still waits for a convincing design. This paper investigates the temporal behaviour of mobile IPv6 with dedicated focus on topological impacts. Extending the hierarchical mobile IPv6 approach we suggest protocol improvements for a continuous handover, which may serve bidirectional multicast communication, as well. Along this line a multicast mobility concept is introduced as a service for clients and sources, as they are of dedicated importance in multipoint conferencing applications. The mechanisms introduced do not rely on assumptions of any specific multicast routing protocol in use.Comment: 15 pages, 5 figure