Java VoIP Library Description

v.8.3 is available     

The Mizu Java VoIP SDK (JVoIP) is a compact and flexible SIP library which consists of one single jar file of ~2 MB and it can be used in many ways:

  • java VoIP library: add VoIP to you java app (or any JVM based) or create your own Java VoIP SIP client
  • standalone VoIP desktop application: as a compact convenient dialer as a Java SIP Softphone
  • console/command line VoIP: flexible java SIP client for any automation with endless configuration capabilities
  • VoIP applet: embedded to your website (you can create a custom HTML/CSS design and access from JS)

Cross platform: runs on all OS with Java SE support (Windows, Linux, MAC, others).
It can be used directly from any JVM language such as Java, Kotlin, Closure, Scala, Groowy, JRuby or Jython.
The API can be accessed from: Java, JavaScript or any others via UDP, TCP or HTTP (clear text, URL, JSON, XML).
Compatible with all SIP server, softswitch or IP-PBX such as Asterisk, Freeswitch, FreePBX, Cisco and others.
The settings can be specified from: API, command line, config file, URL or sent via SIP signaling.

With this Java VoIP SDK you have a full featured SIP/media stack in a single jar file, easy to integrate or embed into your desktop or web java application. Create your custom java voip client, integrate it with callcenter software or embed in VoIP devices such as IP-PBX or gateways so users will have a fully functional VoIP softphone without the need to download any other third-party software. You can create your own custom SIP client, java softphone or use it to add VoIP call capabilities into any software not directly related to VoIP (such as games or CRM’s) or to perform any kind of VoIP automation (auto dialer, auto answer machine, etc).
 

Download SIP library for Java

Download

Example

 //Include the JVoIP.jar to your project (both the package and the main class are named "webphone")
import webphone.*;

//Get a webphone instance
webphone wobj = new webphone();

//Set parameters (replace uppercase words)
wobj.API_SetParameter("serveraddress", "VOIP_SERVER_IP_OR_DOMAIN");
wobj.API_SetParameter("username", "SIP_USERNAME");
wobj.API_SetParameter("password", "SIP_PASSWORD");

//Initialize the sip stack
wobj.API_Start();    

//Make a call (replace DESTINATION with a SIP username, extension, phone number or SIP URI)
wobj.API_Call(-1, "DESTINATION");

//You can also register, accept incoming calls, send chat message and perform other operations.
//Download a working example from here: Java SDK Test.
//See the documentation for more details.

Features

  • Standard SIP client for voice calls (in/out), chat, conference and others
  • SIP/media stack compatible with any VoIP server or client (Asterisk, FreeSWITCH, any PBX, softswitch, gateways, ATA, softphones, IP Phones, X-Lite and many others)
  • Protocols: SIP, RTP, SRTP, UDP, TCP, TLS
  • Peer to peer encrypted media (between JVoIP instances; no server support required for this; can be disabled)
  • SIP server failover (to configured backupserver or after TLS record)
  • NAT/Firewall support: stable SIP and RTP ports, keep-alive, rport support, UPnP, proxy traversal, fast ICE/fast STUN protocols and auto configuration
  • 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 many other SIP related RFC's are implemented
  • Supported methods: REGISTER, INVITE, reINVITE, ACK, PRACK, BYE, CANCEL, UPDATE, MESSAGE, INFO, OPTIONS, SUBSCRIBE, NOTIFY, REFER
  • Audio codec: PCMU, PCMA, G.729, GSM, iLBC, SPEEX, OPUS
  • Video codec: H264, H263, H261, MPEG1, MPEG4, MPEG2, VP8, Theora (optional "as-is")
  • HD Audio: Wideband, ultra-wideband and full-band codecs (speex, opus)
  • Audio enhancements: stereo output (will convert mono sources to stereo) , PLC (packet loss concealment), AEC (acoustic echo canceller), noise suppression, silence suppression, AGC (automatic gain control) and auto QoS
  • Conference calls (built-in RTP mixer, unlimited parties)
  • Voice recording (local file, FTP, HTTP)
  • Audio streaming (play audio files or streams to remote peers, get local/remote audio streams to your app, easy text to speech integration)
  • DTMF (NTE RFC 2833 / RFC 4733, SIP INFO method in signaling or in-band )
  • IM/Chat (RFC 3428), group chat, BLF, SMS (via API) and Presence capability
  • Offline message and message queue with late delivery when peer is not reachable
  • Redial, call hold, mute, forward and transfer (attended and unattended)
  • Balance display, call timer, inbound/outbound calls, Caller-ID display, Voicemail (MWI)
  • Additional features: call parking, barge-in, early media, local ring-back, PRACK and 100rel, replaces
  • Pure Java SIP implementation, including the media stack (you don't have to play compiling native code to various platforms or to deploy large libraries)
  • Unlimited lines (for both simultaneous calls and conference) and multi-account support

Usage
The Java client can be used in a flexible way, suitable for multiple purposes:

  • Use it as a Java SIP library to implement any VoIP solution (VoIP framework for Java)
  • Use it from console as a command line SIP client (it is a powerful SIP client tool with endless configuration options)
  • Use it as a standalone Java SIP application (it has it's own simple dialer GUI)
  • Custom Java callcenter VoIP client (integrate with your call center frontend with voice recording and scripting support)
  • Implement a custom Java SIP client
  • Implement a Java softphone
  • Speaking bot, Voice Assistant
  • Implement remote working / remote office software
  • Add VoIP call capabilities into any JVM application using as a Java VoIP toolkit (Java, Closure, Scala, Kotlin, Groowy, JRuby or Jython)
  • Automate any VoIP tasks, such as auto-dialer, music/voice streams from file, SIP tests, voice quality tests, auto respond or any other SIP automation
  • Java VoIP client for your desktop or web application (you can also integrate into custom hardware such as a mini IP-PBX)
  • Initiate/receive SIP calls from Java, send/receive chat, make conference calls or use any IP-PBX function from Java

 

What's New

The latest stable version (v.8.3) have been released at September 14, 2020 representing a quality upgrade above the previously released v.8.0 build which contains many new features and all cumulative upgrades since 2019 May for the SIP/media stack. For the details, have a look at the version history.
New stable versions are usually built monthly, but demo is published on our website only for major releases usually twice a year. Contact us if you need an unrestricted trial for the latest stable release.

 

Pricing

 
  Demo Basic Advanced Gold
Demo Limiations: yes - - -
Max. nr. of Clients: 50 50 unlimited unlimited
Max. nr. of Servers: 1 1 4 10 or more
Features: advanced standard advanced all
Email Support: presales support         4 requests / 2 hours      15 requests / 8 hours         40 requests / 24 hours      
Dev. Support: - - 1 hour included 8 hours included
Priority Support: - - yes yes, high
Maintenance Upgrades:      - 1 year 2 years 4 years
Price: - $990 $1,900 $3,900
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All prices are in USD for life-time license. Delivery time: one workday.
We also accept wire-transfer; payments can be done also in EUR after xe.com exchange rates.


info@mizu-voip.com

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