History: Inbound Calls Directly to your LinkSys or Sipura

NOTE: This page is a "Work in Progress", and not ready to use yet...

• Does your registered provider not allow inbound SIP URI calls?
• Do you want to cut down on latency/echo, by bypassing your VoIP provider on inbound VoIP calls?
• Do you simply like the idea of allowing calls directly into your VoIP adapter?

If you said yes to any of the above, then this FAQ page may be for you. Here is a description of how to let your LinkSys/Sipura model adapter accept calls directly from a SIP URI (internet VoIP address), bypassing all VoIP providers in the process. Here's how to do it:

• Setup your adapter for use behind a NAT router
  • Setup STUN on your adapter (NOTE: STUN settings are on the SIP tab)
    • Handle VIA received: no
    • Handle VIA rport: no
    • Insert VIA received: no
    • Insert VIA rport: no
    • Substitute VIA Addr: yes
    • Send Resp To Src Port: yes
    • STUN Enable: yes
    • STUN Test Enable: no
    • STUN Server: stun.fwdnet.net:3478
      • NOTE: You can replace the above STUN server with any STUN server you like...
    • EXT IP:
      • NOTE: Leave this setting blank, STUN will figure this out for you...
    • EXT RTP Port Min:
      • NOTE: Normally you can leave this blank, but you can set this if you have a specific need
    • NAT Keep Alive Intvl: 45
      • NOTE: Use an value SHORTER than the "timeout" value in your router.
  • set "NAT Mapping Enable: yes"
• Set "Ans Call Without Reg: yes" on your adapter settings
• Make sure your adapter is on the default SIP port
  • i.e. "SIP Port: 5060"
• Make sure that SOMETHING is set for "User ID:"
  • NOTE: If your adapter is "registered" with a VoIP provider, this will be your real "User ID"
• Make sure NOTHING is in "Outbound Proxy:" field on your adapter
  • NOTE: This field is not normally needed if/when you have STUN setup (as above)
• Forward UDP port 5060 to your adapter
  • NOTE: This may be easier if you use a static LAN address for your VoIP adapter
• Set up a "dynamic DNS" service for your LAN
  • NOTE: The free service from "no-ip.com" works fine for this

If all of the above is setup correctly, then anyone on the internet can directly call your LinkSys/Sipura VoIP adapter by calling "sip:userid@dyamic_dns_address". For example, if your userid is "12345", and your dynamic DNS entry is "myaddress.no-ip.com", then your SIP URI is "sip:12345@myaddress.no-ip.com".

NOTE: One useful purpose of this, is to point a free http://ipkall.com number to your Sipura. You do this by logging into your IPKall account, and filling in your "UserID" info (12345 in this example) for "SIP Phone Number:" and your dynamic DNS entry (myaddress.no-ip.com in this example) for the "SIP Proxy:" field. After saving these changes (and waiting the necessary hour for them to take effect), then calling your IPKall number will directly ring your VoIP phone (bypassing any service provider, including "Free World Dialup").