[Freeswitch-dev] Questions regarding the architecture

[Paul Tinsley]

The reason that I started putting stuff on voip-info and will continue to
put it there is that you have to login to get to trac.  I do think it's too
much to ask to have people register to get documentation on a project that
they are trying to figure out if they care about being part of.  And no-one
will even know there is any documentation unless they are registered,
nullifying most of the usefulness of the content.

On 9/7/06, Brian West <brian.west at mac.com> wrote:
>
>
> On Sep 7, 2006, at 2:26 PM, Alex Guan wrote:
>
> Anthony,
>
>
> Great answers indeed.  Thanks a lot!  Is there a documentation system that
> I can save it to ... like a wiki?  These information are too valuable to
> be buried in the mail archive.
>
>
>
> We have trac.freeswitch.org or voip-info.org (I think we prefer it being
> on trac)
>
>
> It took me a while but I think I have come to agreement for pretty much
> every point you made, except one.  The only thing I felt suspicious is the
> busy waiting in which the originator polls check_channel_status() a million
> times after the originatee starts ringing but before it answers.   Can't
> we wait for a mutex or condition waken up by CALL_ANSWERED?  Imagine if
> you have a thousand incoming calls, there will be 1000 while loops spinning
> the processor.
>
>
> Anthony will need to answer this one.  ;)
>
>
> Another issue I noticed is that hold/resume is not working properly.  There
> was no 200OK for the re-INVITE.  I tried to debug further but somehow
> couldn't get into mod_exosip.c using gdb.  (DDD and Eclipse behaved the
> same.  Does anybody have the same issue?)  I kind of guess the issue is
> what kind of SDP is the SIP phone sending for hold.  The old way is to set
> an all zero O field, as opposed to SENDONLY or INACTIVE.  Anyways, if
> mod_exopsip.c is going way, we should probably just wait for mod_sofia.
>
>
>
> We still have work to do on sofia-sip to bring it up to speed with all the
> features we want.
>
> Two more basic questions:
>
> 1. Licensing.  A main problem I saw with Asterisk is it's GPL/Digium dual
> licensing scheme.  I don't see any serious business development could be
> possible under this license.  I think Mark and his company have made a big
> mistake on this subject.  (And their business model as well.)  What's the
> main rationale for you/other founders to choose MPL?  Do you favour/oppose
> to commercial development using freeswitch in their proprietary systems?
>
>
>
> The MPL allows you to commercialize the code.  But Anthony can elaborate
> on this subject more than I can.  I don't want to speak on his behalf.
>
> 2. Memory usage.  I am seeing freeswitch using 25M bytes of memory with
> just one basic SIP call.  It's a little surprising.  Should I turn on the
> -Os to see if we can save some space?
>
>
>   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+ COMMAND
>
> 25295 root       0 -20 27040  25m 3696 S    0  2.5   0:00.12 lt-freeswitch
>
>
>
>
>
> You can try it out and see.  Also look at what modules you load.  When I
> start freeswitch up on my mac I use 6 or 7 megs of ram.
>
> Thanks,
>
> Alex
>
>
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