Java VoIP Library Description

The Mizu Java VoIP SDK (JVoIP) is a compact and flexible SIP library which consists of one single jar file of 1 MB and it can be used in many ways:

  • java VoIP library: add VoIP to you java app or (or any JVM based) or create your own JavaVoIP SIP client
  • standalone VoIP desktop application: as a compact convenient dialer
  • console/command line VoIP: for any automation
  • VoIP applet: embedded to your website (you can create a custom HTML/CSS design and control for JS)
Runs on all OS with Java SE support (Windows, Linux, MAC, others).
The API can be accessed from: Java, JavaScript or any others via UDP, TCP or HTTP (clear text, URL, JSON, XML).
The settings can be specified from: API, command line, config file, URL or sent via SIP signaling.

With this Java VoIP SDK you have an easy to use full featured SIP/media stack in a single jar file, easy to integrate or embed into your desktop or web java application. For example it can be integrated with callcenter software or embedded in VoIP devices such as 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 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

Example

//Include the JVoIP.jar to your project (both the package and the main class is 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: JVoIP 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 (Cisco, Asterix, 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)
  • NAT/Firewall support: stable SIP and RTP ports, keep-alive, rport support, 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 others
  • 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)
  • Voice recording (local, ftp, http), custom audio streaming
  • DTMF (INFO method in signaling or RFC2833)
  • IM/Chat (RFC 3428), SMS and presence capability
  • 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
  • Unlimited lines
  • Flexible usage: as a java voip library, from command line or as a standalone java SIP client

What's New

The latest version (v.6.0) have been released at January 11, 2017 with the following changes:
  • new: public universal API accessible via UDP/TCP/HTTP (clear text/XML/JSON or URL parameters)
  • new: ability to receive the media streams from the SDK
  • new: multiple simultaneous registrations (multi-account capability via the “extraregisteraccounts” setting)
  • new: UPNP NAT (better NAT handling behind UPNP capable routers)
  • new: reinvite with private media ip if no incoming audio with stun enabled (handling stun external port mismatch)
  • new: more settings such as allowcallredirect, disablesamecall, checkvolumelevel and usecommdevice
  • new: chat composing notifications and send via API_SendChatIsComposing
  • new: detailed call details (API_GetLastCallDetails)
  • new: recorded voice upload via http post (previously only FTP was available)
  • new: auto call forward on no answer (“callforwardonnoanswer” setting)
  • improvements: SIP stack handle warning and diversion headers
  • improvements: stun and turn (earlier public IP discovery, doesn’t use on local LAN’s)
  • improvement: various improvements for SIP and mediaench and more native audio features on Windows OS
  • improvement: local conference RTP mixer
  • improvement: VoIP logs in a separate low-priority thread
  • improvement: better proxy detection for UDP/TCP/TLS
  • improvement: various improvements for direct encrypted peer to peer media handling
  • fix: DNS SRV record timeout handling
  • fix: AEC for wideband speex and opus
  • fix: audio device list on Windows
  • fix: ptime settings for G.729
  • fix: reject double outbound calls to same destination
  • fix: more than 40 minor fixes and improvements

Pricing


Demo Basic Advanced Gold
Trial Limiations: yes - - -
Max. nr. of Clients: 100 400 unlimited unlimited
Max. nr. of Servers: 1 1 4 10 or more
Audio Codec: all G.711, GSM + G.729, Speex all
Conference: yes no yes yes
Other features: all all all all
Email Support: presales support 5 requests / 2 hours 15 requests / 8 hours 40 requests / 24 hours
Dev. Support: 0 0 1 hour included 8 hours included
Priority Support: - - yes yes, high
Maintenance Upgrades:
1 year 2 years 4 years
Price: $0 $990 $1,900 $3,900
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All prices are in USD. You can find the other payment options here. Delivery time: one workday.


jvoip@mizu-voip.com