[asterisk-dev] Viva Chan_Sip, may it rest in peace

[Dan Jenkins]

On Thu, Oct 6, 2016 at 10:01 AM, marek cervenka <cervajs2 at gmail.com> 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, ... ;)
>
>
>
>
>
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Thank you Marek! Much appreciated
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