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Re: NAT state not deleted after IP change (DHCP)
On 21:00, Thu 10 Feb 05, Cyrill R?ttimann wrote:
> Keep alive on the server is nat=yes in the sip.conf?
> 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.
Michiel van Baak
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"Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence."