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@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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