Session Initiation Protocol Attacks and Challenges

In recent years, Session Initiation Protocol (SIP) has become widely used in current internet protocols. It is a text-based protocol much like Hyper Text Transport Protocol (HTTP) and Simple Mail Transport Protocol (SMTP). SIP is a strong enough signaling protocol on the internet for establishing, maintaining, and terminating session. In this paper the areas of security and attacks in SIP are discussed. We consider attacks from diverse related perspectives. The authentication schemes are compared, the representative existing solutions are highlighted, and several remaining research challenges are identified. Finally, the taxonomy of SIP threat will be presented

Efficient and flexible password authenticated key agreement for Voice over Internet Protocol session initiation protocol using smart card

Providing a suitable key agreement protocol for session initiation protocol is crucial to protecting the communication among the users over the open channel. This paper presents an efficient and flexible password authenticated key agreement protocol for session initiation protocol associated with Voice over Internet Protocol. The proposed protocol has many unique properties, such as session key agreement, mutual authentication, password updating function and the server not needing to maintain a password or verification table, and so on. In addition, our protocol is secure against the replay attack, the impersonation attack, the stolen-verifier attack, the man-in-the-middle attack, the Denning–Sacco attack, and the offline dictionary attack with or without the smart card

VERIFIABLY SECURE SESSION INITIATION PROTOCOL REQUESTS

Techniques are described for reducing the amount of spam and congestion on Session Initiation Protocol (SIP) devices and endpoints to significantly improve customer User Experience (UX). This may be packaged as a web Application Programming Interface (API) that provides an “anti-spam as a service” for other web-based clients

Analisa dan Implementasi Session Initiation Protocol dan<br /> Real-Time Transport Protocol Pada Telephony Over<br /> Internet Protocol<br /> Analisys and Implementation of Session Initiation<br /> Protocol and Real-Time Transport Protocol to Telephony<br

ABSTRAKSI: SIP (Session Initiation Protocol) ialah protokol kontrol sinyal pada application layer yang membentuk, memodifikasi, dan menghentikan sesi multimedia seperti internet multimedia conferences, internet telephone calls, dan multimedia distribution. Protokol SIP merupakan protokol berbasis teks dan dapat dikembangkan dengan fitur dan layanan tambahan seperti layanan pengontrolan panggilan dan ketersediaan user, instant messages, mobility, dan interoperability dengan sistem telephony. Adanya teknologi ini diharapkan dapat memenuhi kebutuhan akan koneksi panggilan ke berbagai wilayah selama masih dalam cakupan jaringan internet. Protokol RTP (Real-Time Transport Protocol) membentuk fungsi yang tepat pada transportasi jaringan end-to-end system untuk aplikasi transmisi data real-time seperti audio, video, dan data simulasi melalui layanan jaringan multicast atau unicast. Penerapan Session Initiation Protocol sebagai protokol kontrol pensinyalan dan Real-Time Transport Protocol sebagai protokol yang mengatur pengiriman media pada system yang akan diimplementasikan. Penggunaan teknologi Internet Telephony dengan menggunakan protokol SIP dan RTP dintegrasikan dengan sistem administrasi akan lebih praktis dalam proses pembangunan dan pemeliharaan aplikasi tersebut. Analisis pada SIP dan RTP dilakukan dengan mengaplikasikan penggunaan Type of Service dan codec yang digunakan oleh klien dalam melakukan koneksi pada system sehingga akan didapat behavior pesan dan call flow SIP dan delay, jitter, dan konsumsi bandwidth pada RTP.Kata Kunci : Internet Telephony, SIP, RTP, linux.ABSTRACT: The Session Initiation Protocol is an application layer control signaling protocol for creating, modifying and terminating sessions, these include internet multimedia conferences, internet telephone calls and multimedia distribution. SIP is text-based protocol and can be developed with additional feature and service, these include call control service, presence, instant messages, mobility, and interoperability with other telephony system. This technology can fulfilled the needed of call connection to other area in internet coverage. The Real-time Transport Protocol provides end-to-end network transport functions suitable for applications transmitting real-time data such as audio, video or simulation data, over multicast or unicast network services. By using Session Initiation Protocol for control signaling protocol and RTP for controlling media transport on a system that will be implemented. Internet Telephony technology with SIP and RTP will be integrated with administration system will be more effective in the building and maintenance of the system. Analize processing to SIP and RTP with using of type of service and codecs that used by client in order to connecting to the system so it can be obtain SIP message behavior and call flow and RTP delay, jitter, and bandwidth compsumptions.Keyword: Internet Telephony, SIP, RTP, linux

Covert Channels in SIP for VoIP signalling

In this paper, we evaluate available steganographic techniques for SIP (Session Initiation Protocol) that can be used for creating covert channels during signaling phase of VoIP (Voice over IP) call. Apart from characterizing existing steganographic methods we provide new insights by introducing new techniques. We also estimate amount of data that can be transferred in signalling messages for typical IP telephony call.Comment: 8 pages, 4 figure

Security in Peer-to-Peer SIP VoIP

VoIP (Voice over Internet Protocol) is one of the fastest growing technologies in the world. It is used by people all over the world for communication. But with the growing popularity of internet, security is one of the biggest concerns. It is important that the intruders are not able to sniff the packets that are transmitted over the internet through VoIP. Session Initiation Protocol (SIP) is the most popular and commonly used protocol of VoIP. Now days, companies like Skype are using Peer-to-Peer SIP VoIP for faster and better performance. Through this project I am improving an already existing Peer-to-Peer SIP VoIP called SOSIMPLE P2P VoIP by adding confidentiality in the protocol with the help of public key cryptography

ANALISIS IMPLEMENTASI VOICE OVER INTERNET PROTOCOL (VOIP) PADA JARINGAN WIRELESS LAN BERBASIS SESSION INITIATION PROTOCOL(SIP) ANALYSIS OF IMPLEMENTATION VOICE OVER INTERNET PROTOCOL (VOIP) ON WIRELESS LAN BASED ON SESSION INITIATION PROTOCOL(SIP)

ABSTRAKSI: Dalam perkembangan telekomunikasi, VoIP (Voice over IP) bukan menjadi hal yang dikesampingkan lagi dengan berbagai kendalanya. Pengembangan komunikasi data juga mulai mengarah ke jaringan wireless. Wireless LAN (WLAN) pada mulanya didesain untuk komunikasi data. Komunikasi suara dicoba dikembangkan pada jaringan data. Sifat informasi suara yang harus real time dan reliable menjadi suatu pertanyaan khusus bagi jaringan data wireless. Apakah komunikasi suara pada jaringan WLAN masih memenuhi kelayakan kualitas yang dapat diterima? Penelitian ini menganalisis faktor kualitas suara pada jaringan data wireless dengan memperhatikan rentannya sinyal suara terhadap distorsi. Faktor delay, packet loss, jitter, dan throughput menjadi parameter yang dianalisis. Sehingga kelayakan komunikasi suara pada jaringan data wireless bisa dinilai. Untuk menguji parameter-parameter kualitas suara, penulis merancang suatu topologi jaringan data wireless yang terdiri dari entitas jaringan VoIP berbasis SIP (Session Initiation Protocol). Penulis juga merancang beberapa skenario pengujian untuk mendapatkan nilai-nilai parameter tersebut. Diantaranya yaitu uji jangkauan, uji codec, dan uji access point. Untuk lebih mengetahui performa SIP, penulis juga membandingkan dengan protokol H323. Dari hasil pengujian dan analisa mengatakan bahwa kualitas suara pada jaringan WLAN masih layak dan sesuai dengan standar ITU. Dari hasil percobaan diperoleh bahwa one way delay end to end VoIP SIP yang terukur sebesar 59.634 ms yang masih dalam rentang standar terbaik ITU 0-150 ms. Jitter hasil pengukuran mempunyai rata-rata 0.921 ms. Packet loss rata-rata sebesar 0 %. Throughput juga mencapai 100 %. Jangkauan komunikasi VoIP ini mencapai 40 meter dalam keadaan LOS (Line of Sight) dan 15 meter maksimal untuk kondisi non LOS. Semua parameter hasil penelitian menunjukkan bahwa VoIP masih memungkinkan untuk diimplementasikan pada jaringan wireless. Parameter level SNR yang diterima juga mempengaruhi throughput sistem. Performa SIP dalam WLAN lebih baik 0,2 % dibanding dengan H.323 dilihat dari MOS yang terukur.Kata Kunci : VoIP,SIP,WLANABSTRACT: In the telecommunication, VoIP is not to be ignored by any problem that it belongs. Data communication development also directed to the wireless network. Wireless network is designed for data communication early. But, recent, the voice communication is tried to be implemented in the data network. Voice information characteristic that has to be real time and reliable become a specific question for the wireless network. Whether voice communication on WLAN can be accepted? This research tried to analyze voice quality factor on WLAN by noticed the weakness off data distortion on wireless network. Delay, packet loss, and throughput factor become parameters which be analyzed. So voice communication is proper on WLAN or not. The experiment started by designing a network which is contained by SIP network entities. Scenario is designed to take any value of parameter. Scenarios were reach test, codec test, and access point test. To analyze SIP performance, author also compare SIP with H323. From the test result and analysis, it is mentioned that voice quality on WLAN still fulfill the proper and ITU standard. From the test result one way delay for SIP VoIP is 59.634 ms which is fill on range best ITU standard 0-150 ms. Jitter has an average 0.921 ms. Packet loss has an average 0 %. Throughput also reach 100 %. Coverage reached 40 meters for LOS (Line of Sight) condition and 15 meters for non LOS. All of test parameter show that VoIP has proper level to be implemented in wireless area. SNR level also affected throughput on voice communication. In WLAN SIP have better performance 2 % than H.323 according to MOS test.Keyword: VoIP, SIP,WLA

ALEX: Improving SIP Support in Systems with Multiple Network Addresses

The successful and increasingly adopted session initiation protocol (SIP) does not adequately support hosts with multiple network addresses, such as dual-stack (IPv4-IPv6) or IPv6 multi-homed devices. This paper presents the Address List Extension (ALEX) to SIP that adds effective support to systems with multiple addresses, such as dual-stack hosts or multi-homed IPv6 hosts. ALEX enables IPv6 transport to be used for SIP messages, as well as for communication sessions between SIP user agents (UAs), whenever possible and without compromising compatibility with ALEX-unaware UAs and SIP servers