On 21:00, Thu 10 Feb 05, Cyrill R?ttimann wrote: > Hello, > Keep alive on the server is nat=yes in the sip.conf? > yes. > > Not well suited means, that you have to define the internal ip on the > asterisk server (sip.conf) and the external ip on the sip.conf. If you > are running several servers, that is a burden. There is no single > sip.conf in the cvs to feed all servers. Every sip.conf is unique per > server. In addition, clients have (most of them) to use STUN-Servers. > If you compare to the iax protocol, you have a unique iax.conf on all > server and no STUN-Server's. > I dont use STUN server. I just map the 5060 with static-port and all works perfectly. Indeed, the IAX conf is way easier, but not that many hw phones use the IAX protocol. I was just replying cause I hear a lot of ppl yell "DONT use SIP when doing NAT, it will BREAK things". And that's just plain crap. ok, it takes 2 different pf.conf lines. > > > > > >Then about the bandwidth: > >What codec are you using ? If you use ulaw/alaw the soekris > >should keep up at 20 calls. Try to stay away from iLIBc, it > >is huge but doesn't sound any better then ulaw. > > g711u (raw, I think). We provide box-phones with g711u or g729a if the > customer has limited bandwith available. > But we have to handle hundreds of calls at the same time and to test, > we have to take the codec with the biggest bandwith ... > agreed. Dont want to upgrade everytime you get a new customer right ? To be honest I never tested beyond 50 concurrent calls. I do know those 50 calls went smoothly thru a cheap Realtek based card plugged into our good old Pentium 133 MMX. > > Thanks, > > Cyrill
-- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence."
