CommUniWise: java push-to-talk sip softphone
http://github.com/wisekrakr/Sip_dev_pushToTalk
An open source sip phone with push to talk capabilities. CommUniWise provides an object-oriented Java API for embedding two-way audio. This is a pure client-side solution and requires zero server-side logic on your part.
CommUniWise is a software phone (softphone) compatible with the following specifications:
- RFC 3261 (SIP),
- RFC 4566 (SDP),
- RFC 3550 (RTP),
- RFC 3551 (RTP Audio/Video profile),
- RFC 2617 (Digest Authentication),
- ITU-T G.722 (PCMU, PCMA)
This software has been developed using Oracle Java Development Kit version 7.
These are the dependencies used in the project:
- Commons-cli
- Jain-sip-api
- Jain-sip-ri
- Jain-sdp
- Jain-sip-sdp
- Jain-sip-tck
- Log4j
- Commons-lang3
In program arguments use the following:
- -ip : Your IP address
- -i : Name of your audio input device to be used
- -o : Name of your audio output device to be used
- -u : Your username
- -d : The domain/server registered on
- -p : Your password
- -e : The extension that will be called immediately after registering/logging in.
For example, :
- -ip
127.0.0.1
- -i
Microphone (Best Mics V2)
- -o
Speakers (Big Boi Speakers)
- -u
wisekrakr
- -d
asterisk.local
- -p
1101101
- -e
666
- -d
asterisk.local
SIP account configuration settings:
- Username: name used to register on the domain
- Domain: domain name (like: asterisk.)
- Password: sip account password
- Realm: * (done automatically)
- Proxy Address: * (done automatically)
- SIP Registrar: asterisk server address (server IP or DNS name)
For example, if you have SIP account
666@asterisk.local
with password1101101
, configuration settings you would use:
- Display Name:
666@asterisk.local
- Username:
wisekrakr
- Password:
1101101
- Realm:
asterisk
- SIP Registrar:
asterisk.local
It will register/log you in automatically, with the help of commons cli. When registration is successful, the program will call the extension.
David Buendia Cosano davidiscodinghere@gmail.com