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http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1


https://github.com/sipwise/rtpengine


http://www.opensips.org/html/docs/modules/2.1.x/rtpengine


WebSocket is a protocol that provides full-duplex communication between web clients and servers over TCP connections. Using the WebSocket protocol, browsers can connect to web servers and exchange data, regardless the type or nature of the application protocol. RFC 7118 leveraged this protocol in order to allow browsers to make VoIP calls using the SIP protocol.

This document describes how to use OpenSIPS as the core component of a SIP platform that connects both SIP clients (over UDP, TCP or TLS) as well as browser based clients (using SIP over WebSockets). While OpenSIPS handles the SIP signalling part, media is handled by RTPengine, a high performance media proxy that is able to handle both RTP and SRTP media streams, as well as bridging between them.

This tutorial is inspired from



http://oversip.net/



  • The current solution for using WebRTC with OpenSIPS is by using a gateway between them, such as OverSIP
  • The goal of the discussion is to enlist and evaluate the advantages and disadvantages of integrating WebRTC in OpenSIPS
  • At the end of the meeting we should determine whether the current approach offers a complete solution for WebRTC, or we should integrate it directly in OpenSIPS.
조회 수 :
11570
등록일 :
2015.04.04
11:43:34 (*.160.89.217)
엮인글 :
http://webs.co.kr/index.php?document_srl=365288&act=trackback&key=122
게시글 주소 :
http://webs.co.kr/index.php?document_srl=365288
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