artisticcheese
join:2004-11-09
Carrollton, TX
 ·Future Nine Corpor..
·VoiceStick
| reply to jensjewels
Re: How multiple ATAs can be behind same NAT router?
said by jensjewels  :Use the SIP Port field. THe only SIP port field refers to server port I shall be connecting to not client port. |
|
priller
join:2000-10-20
Gainesville, VA |
BTW ... What router are you using? |
|
artisticcheese
join:2004-11-09
Carrollton, TX | ActionTec (provided by Verizon for FIOS) |
|
joako
Premium
join:2000-09-07
/dev/null | reply to artisticcheese
What provider are you connecting to and what VoIP ATA are you using?
--
Am Heimcomputer sitz' ich hier, und programmier' die Zukunft mir |
|
artisticcheese
join:2004-11-09
Carrollton, TX | Gizmo Project and VOicestick.
One ATA is SPA 2002 and another one CISCO IP Phone 7960 |
|
christcorp
Premium
join:2001-05-21
Cheyenne, WY
 ·Bresnan Online
·VOIPo
1 edit | reply to artisticcheese
I tried reading all the responses and got a little side tracked on some of the responses. So, if my answer is a repeat of someone else's, then my apologies.
You can have as many ATAs behing a NAT router as you want as long as you have enough bandwidth to handle it. Because the ISP and backhaul doesn't QOS yours or anyone else's traffic on the internet, a good NAT router with QOS will help at least with the upload. This is only important if you plan on using 3 or 4 voip adapters at the same time making calls; or if you also plan on surfing a lot while making all these calls.
But, as far as connecting ATA's and having them work, go until your heart is content. Or until you run out of IP addresses in your subnet. (Basically, 254 devices). As far as PORTS go, that doesn't matter at all. You could have 10 Voip adapters on your router and they can all use port 5060 and there wouldn't be any problems. Why? Because when the traffic comes in and out of your router, it isn't based on the port primarily. It's based on the IP address. Once it hits the IP address, THEN it worries about the port.
I.e. 4 different voip adapters. IP addresses of 192.168.1.101 through 192.168.1.104. Each of them can use port 5060 for SIP signaling because when the packet comes in, it's looking for the IP address. Once it finds that, it opens the port. Each IP address in IPv4 uses 65535 ports. It doesn't matter if it's the same port number. Imagine you lived in APARTMENT #4. I ALSO live in APARTMENT #4. My I live at 100 Main St. and you live on 200 Elm st. The APARTMENT number is the port, but the street address is the IP address.
Hope this helps. If I just repeated myself from what someone else said, I apologize. It's been a long night. Later... Mike....
P.S. If you have a 2 LINE VOIP ADAPTER; like a PAP2; then the 1 IP address is running to voip circuits. In that case, you have to make sure the 2 different voip channels are on different ports. But individual adapters don't matter. |
|
joako
Premium
join:2000-09-07
/dev/null
 ·AT&T U-Verse
| reply to artisticcheese
Not sure about the Cisco but for the Linksys set the following, no need to forward ports. Everything else leave default except for whatever settings are needed for your ITSP.
Not related to NAT but for best audio quality also set:
RTP Packet Size: 0.020 (default is 0.030)
--
Am Heimcomputer sitz' ich hier, und programmier' die Zukunft mir |
|
DogFace05
join:2005-12-09
Cary, NC
| reply to christcorp
said by christcorp :P.S. If you have a 2 LINE VOIP ADAPTER; like a PAP2; then the 1 IP address is running to voip circuits. In that case, you have to make sure the 2 different voip channels are on different ports. But individual adapters don't matter. That's an all too common misconception. The reality is that Linksys/Sipura ATAs will happily run with both lines sharing the same port number. Try it and you'll see. I used to make the same assumption until low and behold, one day by accident I ended up with both lines of my adapter set to the same port, and to my surprise it was accepted just fine. As it turns out, the lines are identified and differentiated by their associated user id, so they do not each need to have their own distinct port number. |
|
jensjewels
join:2006-01-09
Austr | reply to artisticcheese
This is a great thread. I for one am learning lots from it  |
|
artisticcheese
join:2004-11-09
Carrollton, TX
·Future Nine Corpor..
·VoiceStick
| reply to joako
said by joako :Not sure about the Cisco but for the Linksys set the following, no need to forward ports. Everything else leave default except for whatever settings are needed for your ITSP. Interesting settins. On my linksys pretty much everything after NAT settings were not set. So all this "VIA" settings supposed to make working behind single IP address easier for NAT router, right?
The only issue why I started the thread is odd behaviour I was getting with new VOIP providers with 2 ATAs behind my router. If I don't use port forwarding then Gizmo Project account will not even ring when incoming call will come in. On the other hand if I do have static port forwarding set to Gizmo Project ATA then incoming calls to Linksys will come in, phone will ring but I would not hear the other party on first time they call, next time they will call and connection establish just fine so I assume mapping for RTP is not done correctly on first call becouse of some weirdness with all this NAT mapings.
I'll try settings below and see if I can use both VOIP adapaters without any port forwarding in place. |
|
B
Premium,MVM
join:2000-10-28
| reply to jensjewels
said by jensjewels :This is a great thread. I for one am learning lots from it Fooey! That's exactly what I came in here to say!
-- B
--
In a realm outside causality and function |
|
DracoFelis
Premium
join:2003-06-15
| reply to artisticcheese
said by artisticcheese :Gizmo Project and VOicestick. One ATA is SPA 2002 and another one CISCO IP Phone 7960 I can't answer for the Cisco (as I don't know their setup), but for the SPA-2002 your client (ATA) port is set on: the "Line x" tab, field "SIP Port:", under the "SIP settings". For example, the ATA line shown in the attached graphic is currently set to client (ATA) port 5064, but still using 5060 (the default) for the remote (proxy) end of the connection.
The way SPA-xxxx adapters work, the "SIP Port" field always represents just your LOCAL (your ATA) port number (NOT the remote proxy port number).
If/when you want to setup the remote proxy (VoIP provider) port number (which usually you want to stay at the default of 5060), you do that (on an SPA-xxxx adapter) by including the number after a colon (:) in the normal "Proxy:" field. If you don't provide a colon in the "Proxy", you will always get the default (remote end) port 5060. However, if you fill in your proxy with "provider:port" (where provider is your VoIP provider, and port is the desired remote port number), you will use that port on the remote end. For example, "Vbuzzer" (a company I used to use a while back) is one of the few VoIP providers that uses an alternate (remote end) proxy. In the case of Vbuzzer, the remote port has to be 80. So when I was using Vbuzzer as my provider, I filled in the proxy field with "vbuzzer.com:80". |
|
DracoFelis
Premium
join:2003-06-15
| reply to joako
said by joako :Not sure about the Cisco but for the Linksys set the following, no need to forward ports. Everything else leave default except for whatever settings are needed for your ITSP. Actually I'm not sure your settings will do much of anything, as you are trying to tell the adapter to process the VIA's, in concert with your STUN and the remote end, but then you are telling the adapter not to use STUN. i.e. I'm not sure you enabled much of anything, with the settings you chose...
And if you did enable STUN with the VIA settings set to "YES", you may experience problems. In my experience, the 4 VIA handling settings, while great in theory, actually cause more incompatibilities than they solve, because they require the remote VoIP company to handle the extra work (and many VoIP companies don't enable those features on their servers). Which is why I prefer the following (alternate) way of setting up adapters, shown below:
NOTE: These STUN settings work most reliably when you are NOT using an "Outbound Proxy" (i.e. the "Outbound Proxy" field, not to be confused with the main "Proxy" field, is empty/blank). However, in most cases, you really don't need the provider's "Outbound proxy" (even if/when they have one) when you have a proper STUN setup on your ATA.
NOTE: Do NOT enable both the initial four "Handle/Insert" VIA settings and the two substitution settings at the same time. They are either/or. I find the two substitution settings to be generally much more reliable, but that's me. In any event, enabling both will almost always cause the two methods to "fight", resulting in your phone calls not working. |
|
christcorp
Premium
join:2001-05-21
Cheyenne, WY
·Bresnan Online
·VOIPo
| reply to DogFace05
said by DogFace05 :said by christcorp :P.S. If you have a 2 LINE VOIP ADAPTER; like a PAP2; then the 1 IP address is running to voip circuits. In that case, you have to make sure the 2 different voip channels are on different ports. But individual adapters don't matter. That's an all too common misconception. The reality is that Linksys/Sipura ATAs will happily run with both lines sharing the same port number. Try it and you'll see. I used to make the same assumption until low and behold, one day by accident I ended up with both lines of my adapter set to the same port, and to my surprise it was accepted just fine. As it turns out, the lines are identified and differentiated by their associated user id, so they do not each need to have their own distinct port number. I agree that it will accept it fine. I have, through experience, been talking on 1 line when a call came in on the other line; so the caller said; and it wouldn't ring on my end. I experimented with it and it was a problem when 1 line was in use and the other tried to receive a call. Not saying that YMMV; just that it happened consistently on mine. Later... Mike.... |
|
joako
Premium
join:2000-09-07
/dev/null
·AT&T U-Verse
| reply to DogFace05
said by DogFace05 :said by christcorp :P.S. If you have a 2 LINE VOIP ADAPTER; like a PAP2; then the 1 IP address is running to voip circuits. In that case, you have to make sure the 2 different voip channels are on different ports. But individual adapters don't matter. That's an all too common misconception. The reality is that Linksys/Sipura ATAs will happily run with both lines sharing the same port number. Try it and you'll see. I used to make the same assumption until low and behold, one day by accident I ended up with both lines of my adapter set to the same port, and to my surprise it was accepted just fine. As it turns out, the lines are identified and differentiated by their associated user id, so they do not each need to have their own distinct port number. Assuming the ATA is correctly configured AND the NAT router plays well with SIP you are totally correct. But if you have problems first thing I might suspect is the router.
--
09:F9:11:02:9D:74:E3:5B:D8:41:56:C5:63:56:88:C0 |
|
