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                              CommUniWise: java push-to-talk sip softphone
                             http://github.com/wisekrakr/Sip_dev_pushToTalk
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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.

LICENSE

SPECIFICATION

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)

PREREQUISITES

This software has been developed using Oracle Java Development Kit version 7.

MAVEN DEPENDENCIES

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

USAGE

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 password 1101101, 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.

AUTHOR

David Buendia Cosano davidiscodinghere@gmail.com

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Open source push-to-talk sip phone

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