WebRTC SIP gateway v.1.0



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Description

WebRTC SIP is a gateway to convert WebRTC calls from browsers to SIP and inverse turning your browser into a phone with audio, video and SMS capabilities. The gateway allows your web browser to make and receive calls from/to any SIP-legacy network or PSTN for example call to X-Lite or any mobile/landline phone.

The supported protocols includes WebRTC, SIP, UDP/RTP/RTCP, WebSocket, ICE, STUN, TURN, DTLS and SRTP.

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  • Release Date:02/25/2016
  • License:Shareware
  • Category:Science

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Versions History
Version Date Released Release Notes
1.0 25.02.2016 WebRTC compatibility fixes, DTLS and SRTP enhancements, RTP routing speedup
Screen Shot

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WebRTC SIP gateway
System Requirements

WebRTC SIP gateway requires WinVista, WinVista x64, Windows2003, Windows Vista Ultimate, Windows Vista Ultimate x64, Windows Vista Business, Windows Vista Enterprise, Windows Vista Business x64, Windows Vista Enterprise x64, Windows Server 2012.

WebRTC SIP gateway v.1.0 Copyright

Shareware Junction periodically updates pricing and software information of WebRTC SIP gateway v.1.0 full version from the publisher using pad file and submit from users. Software piracy is theft, Using crack, password, serial numbers, registration codes, key generators, cd key, hacks is illegal and prevent future development of WebRTC SIP gateway v.1.0 Edition. Download links are directly from our publisher sites. Links WebRTC SIP gateway v.1.0 from Bittorrent, mediafire.com, uploadfiles.com, hotfiles.com rapidshare.com, megaupload.com, netload.in, storage.to, depositfiles.com and other files hosting are not allowed. The download file is obtained directly from the publisher, not from any Peer to Peer file sharing applications such as Shareaza, Limewire, Kazaa, Imesh, eDonkey, eMule, Ares, BearShare, Overnet, Morpheus, BitTorrent Azureus and WinMX.

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