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This project will allow native sip for android device.

It relies on the pjsip sip stack project.

Supported features :

  • SIP for calls and instant messages
  • Android integration with rewriting and filtering rules
  • Codecs : pcmu/a (aka g711u/a); speex; g722; gsm; iSAC; SILK; G729; AMR (depending on device) and as extra plugin : OPUS; g726; g722.1; codec2
  • Echo cancellation (with various backends : webRTC, speex, simple)
  • Auto speaker/earpiece option
  • API provided for other apps plugins.
  • Conference, transfer.
  • Secure call with TLS transport for SIP and SRTP or ZRTP for media.

OR

Features in alpha test :

If you are using x86 or mips CPU arch you can download dedicated version here : Stable versions download

If you want to participate to alpha testing of the application, use nightly builds and have the latest features :

If you encounter any issue, please join and ask on the user group : http://groups.google.com/group/csipsimple-users (csipsimple-users@googlegroups.com). This mailing list will be helpful when you don't know how to configure the application with your server.

If you think that it is a bug, use the issue section of this website. To fill a new issue:

  1. Go in the issue list and search (using searching tool) if your issue is not already referenced.
  2. If not you can open a new issue.

For any other question not related to a bug/enhancement request, you can directly contact the project developers : use this email address developers[at]csipsimple[dot]com or use the google group : http://groups.google.com/group/csipsimple-users (csipsimple-users@googlegroups.com) to address all active developers.

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