IMPLEMENTASI WEBRTC UNTUK LAYANAN VIDEO CALL MENGGUNAKAN SKEMA REDUNDANCY GATEWAY

GUNTUR BHIMA ADYTAMA, PUTRA (2019) IMPLEMENTASI WEBRTC UNTUK LAYANAN VIDEO CALL MENGGUNAKAN SKEMA REDUNDANCY GATEWAY. Undergraduate Thesis thesis, Institut Telkom Purwokerto.

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Abstract

ABSTRACT The development of telecommunication technology is growing rapidly. Technology is formed to facilitate in communicating in particular real-time communication. One of the real-time communication in the form of video calling service. One of the existing video call service applications is Skype. With the application, users must install the app in order to communicate. The installation process takes time so less efficiency in usage, it takes technology to facilitate users, one of which is webRTC technology. WebRTC or Web Real-Time communication is a technology with communication capabilities requiring the installation of special applications to be able to use it. On webRTC Technology applied Redundancy Gateway Network for communication line backups. The type of protocol used in Redundancy gateway is the Hot Standby Router Protocol (HSRP). HSRP is one of the protocols that can be implemented tolerance to interference in the Gateway interface. If any of the interfaces have problems, then in the other interfaces can be used as a substitute for a package to be forwarded to the destination. QoS measurements in the Gateway Redundancy network are based on scenarios with normal network conditions and problematic network conditions using a 5-minute measurement time variation of 10 minutes, 15 minutes dan20 minutes. The highest value of the throughput parameter is at a 20 minute time measurement with the result of a value in normal network conditions of 5899.76 Kbps and the result of a value in a problematic network condition of 5523.44 Kbps. The highest value of the delay parameter is Measuring 20 minutes with the result of the value in normal network conditions that is 1.42 MS and the result of the value in problematic network conditions namely 1.55 Ms. Jitter Parameter on normal network conditions in the range (0,011-0,012) MS and in problematic network conditions in Range (0.21 – 0.24) Ms. The highest value of packet loss parameter is in 20 minutes time measurement with the result of the value in normal network condition that is 0.0075% and problem network condition is 0.0081%. Based on the value of testing process QoS parameters according to the concept of the network of Redudancy Gateway and the results are still under the standardization ITU-T Keywords: webRTC, redundancy gateway, Hot Standby Router Protocol (HSRP), Quality of Service (QoS)

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