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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