CryptoVoIP WebRTC to SIP Gateway
Add WebRTC and call from browser capabilities for your SIP backend, compatible with any SIP server and includes all the necessary component for seamless protocol conversion including built-in auto TLS certificate, STUN, TURN and auto codec conversion on demand. Runs as a transparent proxy, no any changes are required in your SIP server.
SALIENT FEATURES
The CryptpVoIP WebRTC gateway provides seamless integration to your existing SIP and VoIP Infrastructure.
One point solution
Reliable webrtc-sip conversion, TURN and STUN support, flexible routing, supports voice, video, instant messages, call transfer etc.
User-friendly interface
Intuitive UI design, light-weight and reactive web pages
Security
Completely secured, supports TLSv1.2+
Performance
Best in class gateway with proven benchmarkingBest in class gateway with proven benchmarking
HD quality video
Uses video-codecs VP8/VP9
Easy integration
Easy to use
Superior voice quality
Uses audio-codecs OPUS/G.729/G722/Speex
Compatibility:
- On the server side it is compatible with all PBX/VoIP server/SIP trunk/proxy/gateway/carrier supporting the SIP protocol such as Asterisk, 3CX, Broadsoft, Brekeke, Yate, FreePBX, Elastix, Trixbox, Voipswitch, FreeSWITCH, Cisco, Siemens, Huawei, NEC, Mitel and others.
Multiple SIP server support (route calls to one ore more SIP server, accept calls from one ore more sip servers).
Direct SIP peers are also supported. - On the client side you can use any library implementing WebRTC and SIP over WebSocket as specified in RFC 7118, compatible with ).
WebRTC stacks present in browsers like Chrome, Firefox, Edge, Opera and others, WebRTC plugins for IE or Safari or native librariessuch as PJSIP.
All the common WebRTC SIP clients and JavaScript WebRTC libraries are supported such as CryptoVoIP WebRTC SIP client, SIPML5, JSSIP, JS and others.
Works from smartphone, tablet or desktop, using any operating system (Windows, Linux, MAC, Android, iOS