[gh2]

I am new fan of the VoIP world. Can someone help me undesrtand how to configure two Sipura SPA3000 to work in a P2P setting? Is it always required to use a SIP service provider? I understand the Sipura has the Gateway (1-4) functionality in Line-1. Can this method be used by specifying the receiving IP-address as a gateway?

Thanks

G

[jcgalvezv]

Hello gh2.

To configure two Sipuras 3000 to talk to each other you need them to be in the same net (LAN) or to have a public IP address (In the Internet). Each one must have an IP address known by the other.

You don't need to use a SIP service provider to talk from one Sipura 3000 to the other.

You can use anyone of those 4 gateways to configure the IP address of the other end to talk between the 2 sipuras 3000.

Juan C.

[DracoFelis]

Originally Posted by jcgalvezv

To configure two Sipuras 3000 to talk to each other you need them to be in the same net (LAN) or to have a public IP address (In the Internet). Each one must have an IP address known by the other.

I'm not sure that this is true. I think you can also do a P2P VoIP call, by setting up all of the following: 1) Having the Sipuras setup with STUN, 2) Having your (dynamic) public/internet address on a "static DNS" entry using a dynamic DNS service (such as the free NO-IP.COM), and 3) forwarding the SIP ports you are using from your router to your Sipura. I have not been able to completly test this theory (because I currently don't know anyone else with an SPA3000 to help me run some tests against), but the tests I have been able to do suggest it should work.

This would (in theory at least), let you have both Sipuras be behind different NAT routers, and also be on "dynamic IP" broadband (like a lot of us with DSL and cable modems have), and still be completely P2P voice. And that also means you could (again in theory at least) turn on the Sipura to Sipura encription option, to keep your voice conversations "private".

NOTE: I've been mega busy at work lately, so I don't know what time I'll have free "to play with VoIP". However, I am curious about the details of this, so if anyone wants to help test this theory with me, send me a PM (private message via this board). Please include a good phone number to call you at (either a normal USA phone number, a FWD number, or any SIP URI), who I should ask for one someone answers, and a good time to call (and be sure to indicate which time zone you are in, or I will asssume that "a good time to call" means MY local timezone, which isn't necessarily yours)! I could then give you a call, and we could run some tests on using P2P Sipura calling over the general internet. (with my Sipura on my end of the test, and your Sipura on your end of the test).

[gh2]

Thanks for your responses!!

I was actually able to setup and test the 2 Sipuras storing the remote SPA3000's IP address in the gateway-1 field ("gw1"on Line-1 parameters), No authentication required and using a simple dial plan on line-1:

(<8,>55<:UserID@gw1:5060>)

5060 is the port ID on the remote unit.


I still need to test this set up with one unit outside my LAN with a public IP adress.

Regards,

G

[DracoFelis]

Originally Posted by gh2

Thanks for your responses!!

I was actually able to setup and test the 2 Sipuras storing the remote SPA3000's IP address in the gateway-1 field ("gw1"on Line-1 parameters), No authentication required and using a simple dial plan on line-1:

(<8,>55<:UserID@gw1:5060>)

Why waste a GWx field for that, when you can put the IP (or DNS name) directly into your dial plan where you currently have "gw1"? Or even better, use one of your 8 "speed dial" slots (they are on the "User" tab), and avoid cluttering up your dial plan with P2P SIP entries!

NOTE: If you do use the "speed dials", just put "userid@SipuraAddress:SIP_port" into the speed dial, and you should be able to call the speed dial by simply entering "number #" (as in "2#" to use Speed dial 2).

IMHO you only need to use the 4 "gateway" fields if/when you are calling someone that needs an authentication account (which you don't need for P2P Sipura calling).

NOTE: I have gotten P2P calls working between my SPA3000 and my old SPA-2000, when both are behind my home NAT router. And you are pretty much correct about the syntax that you use (except you don't need to waste a "gateway" slot for this, when the sytax works without the gateways in both the dial plan itself, and the "speed dials"). I just haven't had the opportunity yet, to verify if the P2P still works if/when both Sipuras are behind different NAT routers on the general internet. I suspect it will work (with proper STUN setup, and SIP port forwarding at the routers), but I haven't had someone to test the setup against yet.

If you get this to work, please post the details here (for others to use). Alternatively, if you want someone elsewhere on the net to test this theory about, send me a PM with a number I can call you at, and we'll test this setup togeather.

[mamabile]

It does work with proper STUN setup, and SIP port forwarding at the routers. as mentioned by DracoFelis.

These are my setting and work ok for me...

- set your Sipura Box as Static IP address. Otherwise, you cannot do SIP port forwarding to Sipura box from router.

- setup your STUN setting.

- setup SIP port forwarding to Sipura 3000 box. I send these ports to Sipura 3000.
129
5000 to 6000 [I guess you only need 5060/5061]
69

I am not sure if I need all of the above ports...

In [Line 1] you need to have this set.
Ans Call Without Reg: yes

In [User 1] you add the speed dial.
e.g
xxxxxx@carl.dyndns.org:5060 where xxxxx is the UserId of Line 1 of Sipura 3000 box that you are trying to ring.

This has been tested over the internet to another Sipura 3000 box.
Both parties rang successfully.
Both parties talked successfully.

I assume you could have the xxxxx@carl.dyndns.org:5060 as part of dialplan .. but not tested.

[mberlant]

STUN service on top of manually forwarded ports is usually superfluous, but you seem to have backed into a very useful application. In your scenario, though, the STUN service is useful because it plays a part in the negotiation of the RTP port to be used for a particular conversation.

As far as what SIP ports you need to map through the routers, you only need the particular listening port you have configured for whatever ATA channel (Line 1, Line 2, PSTN Line, etc.) you wish to use as point-to-point without a service provider. For Sipura devices Line 1 is usually configured to listen on Port 5060 and Line 2 or PSTN Line are usually configured for 5061, but in a private configuration there is no standard. Bottom line is that ports 69 and 129 are not required, but you might pipe port 80 (or whatever port you change it to) through your router for remote management of your SPA.

[mamabile]

thanks ... I have tested it with just port forwarding 5060 to Sipura 3000 ip-address and all Ok.