I couldn't resist to comment on some lines here. First of all, if you enable keepalive from the sip proxy to the phones (both SER and asterisk can do that) the state from server to phone won't be expiring. That way it is also not true that SIP is not well suited for firewalled connections. I have a asterisk server behind a natting firewall on location A and some phones on different residential places and both Linux and OpenBSD firewalls and all is working smoothly. Just be sure to limit the RTP port range so it's easy to forward it to the internal VOIP server.
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.
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 ...
Thanks,
Cyrill
