Kamailio Webrtc Support

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webrtc The Kamailio SIP Server Project

    https://www.kamailio.org/w/tag/webrtc/
    In 2005, OpenSER project spawned from SER and had to change the name to Kamailio in summer of 2008 due to trademark claims. During the autumn of 2008, Kamailio and SER teams decided to join back the projects and use further the Kamailio name.

GitHub - havfo/WEBRTC-to-SIP: Setup for a WEBRTC client ...

    https://github.com/havfo/WEBRTC-to-SIP
    Mar 11, 2019 · Setup for a WEBRTC client and Kamailio server to call SIP clients - havfo/WEBRTC-to-SIP. Setup for a WEBRTC client and Kamailio server to call SIP clients - havfo/WEBRTC-to-SIP. Skip to content. Why GitHub? ... To be able to support running HTTP(S) and TURN on the same port (443), we need a newer version of nginx that supports streams the way ...

Kamailio IMS installation guide and WebRTC support · Issue ...

    https://github.com/kamailio/kamailio/issues/860
    Nov 16, 2016 · We are working now in the EU project reTHINK and would like to use kamailio ims for interop between webrtc clients and ims world. For the moment, we are evaluating the status of IMS components and WebRTC support. Thus, we would like to know:

Asipto - Kamailio Expertise – SIP, VoIP, WebRTC

    https://www.asipto.com/sw/
    Next Kamailio Advanced Training in Berlin, Germany - March 9-11, 2020! Do not miss the chance to learn how to build scalable real time communication systems! Authentication, authorization and accounting, load balancing, least cost routing, nat traversal, advanced call …

Getting Started WebRTC

    https://webrtc.org/start/
    WebRTC is a free, open project that enables web browsers with Real-Time Communications (RTC) capabilities via simple JavaScript APIs. The WebRTC components have been …

Kamailio WebRTC SIP Server Telecom R & D

    https://telecom.altanai.com/2018/09/04/kamailio-webrtc-sip-server/
    Sep 04, 2018 · The purpose of this article if to demo the process of using Kamailio + RTP Engine to enable SIP based WebRTC call to a traditional SIP UA like Xlite. Kamailio Will thus provide not only call routing but also NATing , TLS and websocket support for webrtc endpoints. For this bridging of SRTP from WebRTC…

Support Asipto - Kamailio Expertise – SIP, VoIP, WebRTC

    https://www.asipto.com/sw/support/
    The services are provided directly by the top level experts, not by a outsourced support team, nor by a call center service. In addition, ASIPTO offers various Kamailio SIP Server training courses to prepare in-house support team, ready to take Level 1 and 2 support queries and maintain your VoIP platform in a day-by-day basis.

Development - Using Kamailio as a webrtc gateway

    http://sip-router.1086192.n5.nabble.com/Using-Kamailio-as-a-webrtc-gateway-td144276.html
    > Hi, > > I need to implement a WebRTC gateway for an existing conference bridge. The > clients application can be a JsSIP client (SIP over websocket or JSON over > websocket). The WebRTC gateway has to support Signaling and ICE and DTLS. > > Can I …

Schedule – Kamailio World Conference 2019

    https://www.kamailioworld.com/k06/schedule/
    Kamailio has superb SIP and WebRTC support and can protect, balance and scale FreeSWITCH insuperable conferencing, video MCU, media switching/mixing, SIMPLE message processing capabilities. We will see how to build and implement such a complete platform, touching both on …



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