<![CDATA[Abstrak Session Initiation Protocol (SIP) merupakan protokol standar multimedia sebagai produk dari Internet Engineering Task Force (IETF) dan telah digunakan sebagai standar VoIP. Protokol ini menggabungkan teknologi seluler dan dunia internet. Sebuah sesi dalam jaringan SIP dapat berupa panggilan suara, e-mail, pesan teks, atau video streaming. Keunggulan SIP adalah operator jaringan dapat menggunakannya untuk mengontrol semua bentuk komunikasi dalam jaringan, bukan hanya suara. Protokol SIP berasal dari Simple Mail Transport Protocol (SMTP) dan Hypertext Transport Protocol (HTTP). Pada penelitian ini layanan Video Streaming, Audio Streaming, Voice over IP (VoIP) dan Conference Call dalam jaringan lokal SIP menggunakan Graphical Network Simulator (GNS3) dianalisis dengan mengamati Quality of Service (QoS) yaitu nilai delay, jitter, packet loss dan throughput dengan Wireshark. Dilakukan juga variasi faktor – faktor yang mempengaruhi QoS yaitu variasi jenis file, ukuran file, panjang kabel Lokal Area Network (LAN) dan gangguan pada pemutaran/playback video dan audio streaming. Dari hasil pengujian yang dilakukan, diperoleh bahwa pada layanan video dan audio streaming, QoS/kinerja layanan dipengaruhi oleh jenis file, ukuran file dan gangguan pada playback. QoS video dan audio streaming secara umum sudah memenuhi standar ITU G.114 baik dilihat dari besar delay, jitter, packet loss maupun throughput-nya. Pada layanan VoIP dan Conference Call, QoS dipengaruhi oleh variasi panjang kabel LAN dan secara umum sudah memenuhi standar. Kata Kunci : QoS, SIP, GNS3, Video Streaming, Audio Streaming, VoIP, Conference Call Abstract Session Initiation Protocol (SIP) is a multimedia standard protocol as a product of the Internet Engineering Task Force (IETF) and has been used as a standard VoIP. This protocol will merge together the cellular and the Internet worlds. A session in SIP networks can be anything like voice call, e-mail, text message, or video stream. Excellence of SIP is network operators are able to use SIP to control all forms of communications within networks, not just voice. The SIP protocol derived from the Simple Mail Transport Protocol (SMTP) and the Hypertext Transport Protocol (HTTP). In This research, Video Streaming, Audio Streaming, Voice over IP (VoIP) and Conference Call service on the local SIP network using the Graphical Network Simulator (GNS3) is analyzes by observing Quality of Service (QoS)that is delay, jitter, packet loss and throughput with Wireshark. Also performed variety of factorsaffect the QoS that is the variation of file format, size, Lokal Area Network (LAN) cable long and disturbance on video and audio streaming playback. From the results of tests performed, it found that in the video and audio streaming services , QoS/service performance is affected by file type, file size and disturbance on playback. QoS or performance video and audio streaming service has been meet ITU G.114 standard with good views of it delay, jitter, packet loss and throughput value. On VoIP and Conference Call, QoS is affected by length variation of LAN cable and generally already meet the standards. Keywords: QoS, SIP, GNS3, Video Streaming, Audio Streaming, VoIP, Conference Call]]>
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