EZWEBPHONER-8200 WebRTC Phone Server
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Provide WebRTC to SIP conversation in a seamless manner.
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Ezvoicetek EZWEBPHONER-8200 is a WebRTC to SIP gateway sever which enables VOIP customers extend their service to browsers based applications in a seamless manner. EZWEBPHONER-8200 is based on the standard WebRTC protocol and can be running under any browsers which support WebRTC, including audio and video calls. It plays a bridge to SIP Server or IP-PBX in between WebRTC and SIP. Together using with Ezvoicetek IP contact center solution, it builds a very flexible plug and play contact center. It serves customer anywhere of the world.
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Highlights and Benefit
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• WebRTC to SIP
• DTLS/SRTP Encryption
• SIP RFC 3261
• SIP MWI/BLF and Phone Book
• NAT Traversal
• Support SIP Audio Transcode
• Support 3-way Conference
• Support 3PCC Calling form Simple Call Control
• Websocket API and Sample Code
• Support Click to Call
• Multi-Language Web Interface
• Optimized based on 64 bits Linux and IPV6 Ready
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Product Specifications
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System Requirements |
• INTEL/AMD CPU (Intel® 64)
• CentOS/RHEL 8
• MYSQL Database
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Convergence Technologies |
• SIP RFC-3261/2543
• RTP/RTCP
• DTLS/SRTP
• HTML5/Websocket
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WebRTC Features |
• Easy Javascript Websocket API
• Support NAT Traversal
• Support WebRTC G.711 Audio Codec
• Support VP8/VP9 Video Codec
• Support RTP Timeout Detect
• SIP Audio Transcode
• Support 3-ways Conference
• Support 3PCC Calling
• Support Click to Call
• Support phone book integration with EZVMS-6800 or EZPBX-2000
• Sample Source Code
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SIP |
• SIP Register
• SIP MWI/BLF
• SIP Message
• SIP Call Control
• Incoming Call Answer/Reject
• Outgoing Call
• Call Waiting/Hold/unHold
• Call Transfer
• 3-Ways Conference
• 3PCC Calling
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System Capabilities |
• Max Register : 2000
• Max Concurrent Calls : 512
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Security |
• HPPTS TLS Encryption Security
• Secure Websocket Encryption
• Media Encryption using SRTP
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Audio Transcode |
• SIP side RTP Transcode
• G.729
• G.722
• G.711A
• G.711U
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Reports |
• SIP User Register Statistic
• Audio/Video Call Statistic
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Management |
• System Real Time Status
• Real Time Register Status
• Real Time Call Status
• Multi-Language Support
• Web Provisioning Access Log
• Easy Web GUI (HTTP/HTTPS)
• Real Time System Monitor & Tracing
• Scheduled Update Task
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Application |
• IP Contact Center Web Softphone
• Outbound Dialing System
• Click to Talk Web Calling
• 3PCC Simple Call Control
• IP-PBX/Softswitch Web Softphone
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Screenshots
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• Call Control |
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• Service Parameter |
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• Click To Call |
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• Register Statistic |
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• Call Statistic |
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• Video Call Statistic |
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• Call History |
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• Register Status |
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• Call Status |
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