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(-1, "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, 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)
• 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 callcenter VoIP client (integrate with your call center frontend with voice recording and scripting support)
• Implement a custom Java SIP client, a Java softphone or any VoIP work-flow
• Speaking bot / voicebot, Voice Assistant, AI integration (OpenAI/ChatGPT), text to speech, Push to Talk, Intercom, smart home solutions
• 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

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


info@mizu-voip.com