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garys_2k
join:2004-05-07
Farmington, MI
 ·Future Nine Corpor..
·Vonage
| Re: How multiple ATAs can be behind same NAT router? Another thing to keep in mind is that all SIP traffic, and the RTP traffic that the SIP traffic sets up, originates FROM the ATA which is behind the router. Port forwarding should not be needed for any ATA. It's the router's job to keep track of which internal IP sent a request out to which external IP and route the reply back -- it is the essence of NAT.
You can have multiple PCs browsing the same web page and the router will keep track of which machine just asked for the page to be refreshed, it's no different. ATA traffic is UDP but those incoming packets are not unsolicited. | |
|  artisticcheese
join:2004-11-09
Carrollton, TX
·Future Nine Corpor..
·VoiceStick
| Re: How multiple ATAs can be behind same NAT router? said by garys_2k  :Another thing to keep in mind is that all SIP traffic, and the RTP traffic that the SIP traffic sets up, originates FROM the ATA which is behind the router. Port forwarding should not be needed for any ATA. It's the router's job to keep track of which internal IP sent a request out to which external IP and route the reply back -- it is the essence of NAT. You can have multiple PCs browsing the same web page and the router will keep track of which machine just asked for the page to be refreshed, it's no different. ATA traffic is UDP but those incoming packets are not unsolicited. How this will work for incoming call? | |
| | | | | artisticcheese
join:2004-11-09
Carrollton, TX | Re: How multiple ATAs can be behind same NAT router? I hear first call ringing but when I pick up I hear dead air. What most likely cause for this? I have 2 different VOIP Providers behind router and both of those use SIP UDP 5060. | |
| | | JTS33
join:2003-05-03
USA
| said by priller :Your VoIP provider's SIP proxy knows the address and port you registered from, so it just sends the INVITE to you on that port. In my example above, the INVITE comes to me on 1061 ... is translated back to 5060 and is passed along to the ATA. -- Please don't screw with port forwarding. If for some reason you have a problem receiving calls, the problem is that the NAT translation in your router is timing out. This is resolved by adjusting the timeout in your router. If that isn't an option, then request that your provider increase the registration interval or enable NAT keepalive on the ATA. Router: Airlink101 AR430W SuperG Wireless Router
ATA: Linksys PAP2 v1 flashed to SPA1001, STUN enabled.
My Internet connection is Dynamic IP, and I noticed when my ISP assigns me a new IP address, registration to GizmoProject would fail and I would not get any incoming calls.
Using the "DHCP Release" and "DHCP Renew" functions in the router to force a new dynamic IP, I spent a day messing around with the ATA settings, and registration would still always fail when my WAN IP changed.
Then I figured out that if I rebooted the router after the WAN IP address changed, GizmoProject registration did not fail.
Just curious if there is a technical explanation of what may be causing this?
Is it partially due to something on GizmoProject's end, or is it solely my router? | |
| | garys_2k
join:2004-05-07
Farmington, MI
·Future Nine Corpor..
·Vonage
| said by artisticcheese :said by garys_2k :Another thing to keep in mind is that all SIP traffic, and the RTP traffic that the SIP traffic sets up, originates FROM the ATA which is behind the router. Port forwarding should not be needed for any ATA. It's the router's job to keep track of which internal IP sent a request out to which external IP and route the reply back -- it is the essence of NAT. You can have multiple PCs browsing the same web page and the router will keep track of which machine just asked for the page to be refreshed, it's no different. ATA traffic is UDP but those incoming packets are not unsolicited. How this will work for incoming call? Here's how it works...
Your ATA sends the SIP server an "I'm here" message, the SIP server sends back a "Got it, thanks" reply. This happens regularly, and it should happen regularly enough to keep your router's tables current. My Vonage ATA does this 3 to 4 times per minute, plenty often enough.
When a call comes in the SIP server (which is still current to the ATA in the table) sends the ATA a message "Hey, go take a call at this IP address." The ATA sends back an OK to the SIP server and then initiates contact to the RTP server's address that the SIP server delivered to it.
That goes something like, ATA: "Hey, RTP server, heard you have a call for me -- here's my phone number." The RTP server sends back "Yeah, let's get it going. Start the bell ringing and here's the caller ID info to throw on the wire."
The call proceeds entirely on the RTP server but communication with the SIP server is maintained. Once the call is over the ATA and the SIP server put things away and the regular traffic starts over again.
As for handling different ATAs, the router's NAT table remembers which internal IP is associated with which SIP and RTP traffic. All of that traffic is initiated OUT from inside the LAN so the router, if it doesn't get mixed up (which some cheap ones can, especially if they're handling tons of addresses as can happen with file sharing) sends the packets to the proper IP on both sides. | |
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