On Thu, Oct 6, 2016 at 10:01 AM, marek cervenka <[email protected]> wrote:
> > Michael, >> >> What would be amazing is for you to tell us which features you are >> missing (or were missing when you tried) >> >> If we start a working group around PJSIP migration then these points will >> help drive that forward. >> >> Dan >> >> > > feedback on marketing features over chan_sip (and not only marketing!) > * 95% parity with chan_sip with examples (its possible drop some > functions for technical reasons) > * good webrtc compatibility with jssip ,simpl5 with actual examples > * pieces needed for support good voice over bad networks (opus, plc, > remb,...) (i know the part of the thing is in media stack) > * REST API for managing endpoints (hide the backend for newcomers from web > world) > * support for sipcapture.org/statsd (its already done!) > * and in general ... better architecture, stability, scalability, ... ;) > > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev > Thank you Marek! Much appreciated
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