Here are the variables I used to set up my SPA 2102 ATA.
Set the DHCP on your home router to give your Linksys ATA the same IP address all the time (if you can).
Forward ports 5060 and 5061 from the internet to the IP address of the ATA.******** (5061 is for the second line if you every use it)
login to the ATA
Click the admin tab
Clink advanced tab
Click the voice tab
Under "System", set a admin password.
Click the SIP tab (you need to have clicked "advanced" to see this)
Change RTP packet size: 0.010 (This sends a packet every 10ms second vs the default of 30ms)
NAT Keep Alive Intvl: 15*
Click Regional Tab
Ring and Call Waiting Tone Spec, Ring Waveform : Sinusoid** (This is the north american standard)
Ring Voltage : 90** (lower voltage causes old mechanical ringer phones to sound like a dead cat)
Set time Zone: (GMT-05:00)
Daylight saving time rule: start=3/8/7/2:00;end=11/1/7/2:00;save=1
Under Line 1
NAT Mapping Enabled:NO
NAT Keep Alive Enabled: NO
NAT Keep Alive Msg:$PROXY
NAT Keep Alive Dest:$PROXY
Network Jitter Level: low
SIP port: 5060
Proxy: montreal2.voip.ms*** (This has better uptime the toronto2.voip.ms)* (needs to match your account setting on website)
Display Name:* <Enter the name for outgoing call name display>
User ID: <Account from voip.ms>
Password: <Voip.ms provided>
Register Expires:60************ (This means that the phones registers and updates your IP address to the VOIP server every 60 secs)
Preferred Coded: G711u
In addition, he gave me the following information:
* For a voip connection, the amount of bandwidth is not the limiting factor but the jitter (change in latency). You can test out the following website, when you are running your applications to see if there is any problems on your line.* http://myspeed.visualware.com/indexvoip.php**
The standard codec for voip is G7.11 (64Kbps).* Its actually about 90Kbps when you factor in everything.