实现WebRTC与SIP之间的互联网关,实现实时音视频通信

S. P, Pradhyumna P, Mohana
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引用次数: 2

摘要

需要将实际或实时的视听通信基础设施整合到网络中,并将其与web一起使用,这引发了一种发展,导致引入了许多新技术。在当今的融合网络中,实时通信至关重要。今天,基于Ip的服务,如协作视频通话、视频会议、会议、聊天、消息和外观,都非常有名和广泛使用。这种服务彻底打破了全世界的通信界限。我们生活的许多方面现在都受到这些技术的影响,包括学校教育。其中最重要的是会话发起协议(SIP)和Web实时通信(WebRTC)。如果两台使用不同服务提供商的计算机希望彼此通信,它们需要像SIP这样的VoIP信令协议来实现这一点。网关是作为WebRTC和SIP之间的中介的元素。本文介绍了将SIP和WebRTC这两种互联网关键技术融合在一起的技术要素,在两者之间搭建桥梁。
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Internetworking Gateway between WebRTC to SIP to Integrate Real-Time Audio Video Communication
The need to integrate actual or real-time audio-visual communications infrastructure in networks and their uses, along with the web, sparked a development that resulted in introduction of numerous new technologies. In today's converging networks, real-time communication is critical. Today, Ip - based services such as collaborative video calls, videoconferencing, conferencing, chatting, message, and appearance are highly famous and widespread utilized. Such services completely dismantled communications boundaries throughout the world. Many parts of our life are now influenced by these technology, including schooling. One of the most essential are the Session Initiation Protocol (SIP) and Web Real-Time Communication (WebRTC). If 2 computers using different service providers wish to communicate with each other, they need a VoIP signalling protocol like SIP to do so. Gateway is the element that works as an intermediary between WebRTC and SIP. This paper describes technology of the elements of merging these two key internet technologies, SIP and WebRTC, to build a bridge between them.
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