Webphone (formerly Mizu Webphone) 4.8

Embed a VoIP phone into your website and allow your users to make calls to landline or mobile networks much easier, from a user-friendly interface
WebPhone is a tool based on VOIP phone that can be run even from webpages. Based on the industry standard SIP protocol, it is compatible with all VOIP devices and services. It can call any other SIP soft phone (for free charge) or any landline or mobile number via a VOIP service provider of your choice. WebPhone can be used for adding a VoIP application into your website.

WebPhone is very similar with a regular VoIP softphone software, with the following differences:
· it can be easily embedded in webpages
· it can be easily customized: o you can change it after your needs by modifying its parameters including the display language o you can create any kind of user interface with simple HTML/CSS o you can control the call logic as you wish with the JavaScript API o use a skin or run it in the background

Just copy the webphone.jar to your webserver and refer to it from anywhere from your html with the “applet” tag. Set at least the “serveraddress” applet parameter to the IP or domain name of your VoIP server.

Main features:

  • SIP and RTP stack (compatibile with standard VOIP servers like Cisco or Asterix)
  • Standard Java applet (no installation required. runs directly from browsers)
  • VoIP calls with auto QoS
  • Supports multiple transport protocols: UDP, HTTP tunnel, encrypted UDP, TCP tunnel, TCP, TLS, SOCKS and HTTP proxy traversal
  • NAT/Firewall support
  • IM (chat RFC 3428), SMS and presence capability
  • RFC’s: 2543, 3261, 2976, 3892, 2778, 2779, 3428, 3265, 3515, 3311, 3911, 3581, 3842, 1889, 2327, 3550, 3960, 4028, 3824, 3966, 2663, 3022 and others.
  • Multiple supported methods: INVITE ,reINVITE, ACK,PRACK, BYE, CANCEL, UPDATE, MESSAGE, INFO, OPTIONS, SUBSCRIBE, NOTIFY, REFER
  • Call parking, early media, local ring-back, PRACK and 100rel, replaces
  • Audio improvements: Stereo output, PLC, AEC, noise and silence suppression and AGC
  • DTMF (INFO method in signaling or RFC2833)
  • Redial, call hold, mute, forward and transfer (attended and unattended)
  • Conference calls (built-in RTP mixer)
  • Call park and pickup, barge-in
  • Unlimited lines
  • Balance display, call timer, inbound/outbound calls, Caller-ID display
  • Voice recording (local and/or ftp upload), custom audio streaming
  • Voicemail (MWI)
  • Signaling and media tunneling and encryption
  • VPN tunneling
  • Click to call
  • Server side integration using PHP, .NET, J2EE , etc
  • Integration with any webpage or third party application
  • JavaScript API
  • Built with your own brand-name

last updated on:
January 25th, 2014, 11:47 GMT
file size:
3.2 MB
price:
$500.00
 
developed by:
MizuTech LTD
license type:
Demo 
operating system(s):
Windows 2K / XP / Vista / Vista 64 bit / 7 / 7 64 bit / 8 / 8 64 bit
category:
C: \ Internet \ Telephony / SMS & GSM

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user rating 3

4.3/5
 

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1 Screenshot
Webphone (formerly Mizu Webphone) - Webphone allows you to connect to your SIP account and start making calls.
What's New in This Release:
  • New Opus codec
  • New OGG recording
  • New Video module
  • New AEC module
read full changelog

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