Run it as a standalone app:

//Launch with the built-in user interface (you might pass any parameters or create a wpcfg.ini file with the parameters):
just double click on the JVoIP.jar file or launch as java -jar JVoIP

//Launch as a command line app from your terminal:
java –jar JVoIP.jar iscommandline=true serveraddress=VOIP_SERVER_IP_OR_DOMAIN username=USERNAME password=PASSWORD callto=DESTINATION autocall=true loglevel=1
(Ask for the headless version if you wish to test on a headless OS such as a linux server and the demo doesn't start)

Use it as library for your app:

//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. There are 100+ configuration parameters)
//wobj.API_SetParameter("proxyaddress", "OPTIONAL_OUTBOUND_SIP_PROXY");
wobj.API_SetParameter("serveraddress", "VOIP_SERVER_IP_OR_DOMAIN");
wobj.API_SetParameter("username", "SIP_USERNAME");
wobj.API_SetParameter("password", "SIP_PASSWORD");
//see the "Parameters" chapter in the documentation for more settings

//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("DESTINATION");

//You can also register, accept incoming calls, send chat messages, get notifications and perform many other operations.
//Example code: JVoIPTest.java.
//See the documentation for more details.

• 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)
• Compatible with all JRE/JDK since Java 1.5 (including both ancient and latest Java versions)
• Protocols: IPv4/IPv6, UDP/TCP/TLS/WS/WSS, SIP/SIPS, RTP/SRTP
• 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
• IMS/3GPP/VoLTE (basic compatibility and features such as USSD or 3GPP SMS)
• 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, G.722.1
• Video codec: H264, H263, H261, MPEG1, MPEG4, MPEG2, VP8, VP9
• HD Audio: Wideband, ultra-wideband and full-band codecs (speex, opus, g.722.1)
• 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/SIPREC in wav/mp3/gsm/ogg/RTP format)
• Media streaming (play audio/video streams/files to remote peers, get local/remote audio streams to your app or to external apps/cloud services)
• DTMF (NTE RFC 2833 / RFC 4733, SIP INFO method in signaling or in-band )
• IM/Chat (RFC 3428), group chat, BLF, SMS (including 3GPP SMS support) and presence capability
• Offline message and message queue with late delivery when peer is not reachable
• Redial, call hold, MOH (music on hold), mute, forward and transfer (attended and unattended)
• AI Integration: LLM chat and/or real-time speech-to-speech/SST, STT, TTS
• Balance display, call timer, inbound/outbound calls, Caller-ID display, voicemail (MWI)
• Additional features: call parking, barge-in, early media, 3PCC, ED-137, 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 VoIP client (integrate with your CRM or call center frontend with voice recording and scripting support)
• Implement a custom Java SIP client, a Java softphone or any VoIP work-flow
• LLM AI: SIP-AI CLI, Java SIP AI library, SIP AI Agent, OpenAI realtime API integration, STS speech to speech, TTS
• Voicebot, Voice Assistant, AI receptionists, Intelligent IVR, AI SIP extension
• Push to Talk, Intercom solutions for smart homes, hotels or nursing care
• Automotive (SAAS Solution for Automotive and other Industries)
• IoT solutions, transport, aviation, navigation, military/defense and other industries
• Remote working / remote office software implementations
• 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 on devices like Raspberry PI)
• Initiate/receive SIP calls from Java, send/receive chat, make conference calls or use any IP-PBX function from Java

The latest stable version (v.9.6.26056) have been released at June 18, 2026 with AI integration and many other new features. See the version history for the details.

All prices are in USD for life-time license. Delivery within maximum one workday by email.

* "Max. Ports" means the maximum number of JVoIP instances (when used as a client UA), the maximum number of connected clients (when used as a server) and also the maximum number of simulatenous calls across all instances.
* "Max. Servers" means different SIP servers (softswitch / IP-PBX / SIP proxy) configured for the SDK as registrar and for outbound/inbound calls (usually by the "serveraddress" parameter). Applied only when used as a client. The Gold version can be used with unlimited servers as described here.


info@mizu-voip.com