I recently started writing a pjsip/pjsua2 binding for node that is available on npm[1] and github[2]. I would have loved to have gone pure javascript for a sip stack, but the currently available solutions were inadequate for the project I need this for. Also pjsip is the basis for a/the new SIP channel driver used by Asterisk 12+, so it must be good right? ;-)
Right now most of the core functionality is there, but most of my testing has been with dealing with incoming calls as a SIP trunk. I'm working on adding real automated tests to the project using SIPp[3]. Here are some of the things that you can do currently: * Make and receive calls * Play either individual or a playlist of wav files (ulaw, alaw, or pcm) * Record audio to wav file (ulaw, alaw, or pcm) * Hook up audio streams from different calls (e.g. create your own conference or record a mix of streams to wav) * Adjust volume levels of audio streams * Detect/Send DTMF digits * Hold/un-hold * Call transfer [1] npm install sipster [2] https://github.com/mscdex/sipster [3] http://sipp.sourceforge.net/ -- -- Job Board: http://jobs.nodejs.org/ Posting guidelines: https://github.com/joyent/node/wiki/Mailing-List-Posting-Guidelines You received this message because you are subscribed to the Google Groups "nodejs" group. To post to this group, send email to [email protected] To unsubscribe from this group, send email to [email protected] For more options, visit this group at http://groups.google.com/group/nodejs?hl=en?hl=en --- You received this message because you are subscribed to the Google Groups "nodejs" group. To unsubscribe from this group and stop receiving emails from it, send an email to [email protected]. For more options, visit https://groups.google.com/d/optout.
