About WebRTC SIP gateway
WebRTC SIP is a gateway to convert WebRTC calls from browsers to SIP and inverse turning your browser into a regular softphone.
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.
Both the signaling and media conversions are supported.
Compatible with any WebRTC implementation such as SIPML5, JSSIP and SIP.JS implementing RFC7118.
Beside WebRTC to SIP and SIP to WebRTC, WebRTC to WebRTC and SIP to SIP calls are also supported.
Previous Versions
Here you can find the changelog of WebRTC SIP gateway since it was posted on our website on 2016-04-21 18:04:51.
The latest version is 1.0 and it was updated on 2024-04-23 08:40:16. See below the changes in each version.
WebRTC SIP gateway version 1.0
Updated At: 2016-02-25
Changes: WebRTC compatibility fixes,
DTLS and SRTP enhancements,
RTP routing speedup
Disclaimer
External Download
We do not host WebRTC SIP gateway on our servers. We did not scan it for viruses, adware, spyware or other type of malware. This app is hosted by the software publisher and passed their terms and conditions to be listed there. We recommend caution when installing it.
The external download link for WebRTC SIP gateway is provided to you by apps112.com without any warranties, representations or guarantees of any kind, so access it at your own risk.
If you have questions regarding this particular app contact the publisher directly. For questions about the functionalities of apps112.com contact us.
