EZWEBPHONER-8200 WebRTC Phone Server
Provide WebRTC to SIP conversation in a seamless manner.
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.
 
Highlights and Benefit
• 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
 
Product Specifications
System Requirements
• INTEL/AMD CPU (Intel® 64)
• CentOS/RHEL 8
• MYSQL Database

Convergence Technologies
• SIP RFC-3261/2543
• RTP/RTCP
• DTLS/SRTP
• HTML5/Websocket

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

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
System Capabilities
• Max Register : 2000
• Max Concurrent Calls : 512

Security
• HPPTS TLS Encryption Security
• Secure Websocket Encryption
• Media Encryption using SRTP

Audio Transcode
• SIP side RTP Transcode
• G.729
• G.722
• G.711A
• G.711U

Reports
• SIP User Register Statistic
• Audio/Video Call Statistic

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

Application
• IP Contact Center Web Softphone
• Outbound Dialing System
• Click to Talk Web Calling
• 3PCC Simple Call Control
• IP-PBX/Softswitch Web Softphone
 
Screenshots
• Call Control
Call Control
 
• Service Parameter
Service Parameter
 
• Click To Call
Click To Call
 
• Register Statistic
Register Statistic
 
• Call Statistic
Call Statistic
 
• Video Call Statistic
Video Call Statistic
 
• Call History
Call History
 
• Register Status
Register Status
 
• Call Status
Call Status
 
Service Provider Solution
IP Contact Center Solution
Enterprise Service Solution
SIP Proxy Server
IP Centrex Server
Enterprise IP-PBX
IP Contact Center Server
SIP IVR Application Server
VoIP Monitor Server
WebRTC Phone Server