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scerruti
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PostPosted: Sat Feb 05, 2005 10:35 pm    Post subject: SIP call to ATA using IP Reply with quote Back to top

I have a VT1000 behind a router. I am using NAT.

I would like to make a SIP call from my internal LAN directly to my ATA without going through Vonage.

Is this possible?
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sajer
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PostPosted: Sun Feb 06, 2005 11:46 am    Post subject: Reply with quote Back to top

Yes, it's possible, but will be more or less work depending on what you want to do.

You said only that you wanted to place a call to your ATA from another device on your network. That is pretty easily actually. If you have another SIP phone on your network, you could simply set its sip proxy to be your ATA. Assuming there are no nat issues between the two of them, you can simply dial your Vonage number from that other phone and it should ring your ATA.

However, if you want the reverse as well -- to call your other devices on the LAN from your ATA without going through Vonage -- then that is more difficult. The reason is that your ATA is configured to send every call to Vonage. People that need to do this type of thing typically use an open source product called Asterisk, which is a software-based SIP PBX. You would run an Asterisk server that registers with Vonage on behalf of all your LAN phones, then your ATA would talk to Asterisk -- if you dial an extension on your LAN it is completed directly, if you dial any other number asterisk sends it through Vonage.

Running asterisk requires a Linux box and the accompanying know-how to configure and run asterisk. Also support from Vonage in terms of giving you the phone authentication key so you can configure it into asterisk.
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scerruti
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PostPosted: Sun Feb 06, 2005 4:31 pm    Post subject: Reply with quote Back to top

Yes, I just want to call the ATA from inside my network, nothing fancy.

I have tried what you have suggested with a couple of soft phones but so far have been unsuccessful.

I set up an FWD account. I get a softphone working with that account and then I attempt to contact the ATA by mucking with the SIP proxy settings.

The result is that I am usually either sending <my number>@fwd.pulver.com and not getting a response (408 timeout), or I am managing to unregister myself all together and the soft phone won't let me even place a call.

Have you got any suggestions on what else I could try? I am about to break down and start writing code in an attempt to talk to the ATA. I was just hoping to prove it could be done first.
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sajer
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PostPosted: Sun Feb 06, 2005 4:41 pm    Post subject: Reply with quote Back to top

Take fwd out of the picture. Your ATA won't respond to an INVITE with anything other than the sip address that it registered with. So if the ip address of the ATA on your network is 192.168.1.100, you would want to be sending the <my number>@192.168.1.100. Try the following:

1. Configure your softphone (xten, correct?) to NOT register (Register: never)

2. Remove sip realm or anything else that refers to fwd.

2. Configure the sip proxy as the ip address of your ata.

Then try again. If this doesn't work, then your ATA is probably registering with an address something than what I guessed above. In that case you'll need to get an ethereal trace of the sip REGISTER message and see what the phone is putting in the Contact header, so you can send the call with that address.
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scerruti
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PostPosted: Sun Feb 06, 2005 11:02 pm    Post subject: Reply with quote Back to top

Still couldn't make this work. Going to sleep on it and try it again with ethereal in the loop once more.

Ethereal told me that the REGISTER contact used my private IP address on port 5061. The INVITE from a call I placed from PSTN through Vonage had my public IP and I assume the dynamic port assignment NAT had handed out.

At this point I am trying to figure out if I have to route my calls through my public IP and if I will have to use my public IP and the NAT port on the invite. Any thoughts?
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sajer
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PostPosted: Sun Feb 06, 2005 11:09 pm    Post subject: Reply with quote Back to top

scerruti wrote:
Ethereal told me that the REGISTER contact used my private IP address on port 5061. The INVITE from a call I placed from PSTN through Vonage had my public IP and I assume the dynamic port assignment NAT had handed out.

At this point I am trying to figure out if I have to route my calls through my public IP and if I will have to use my public IP and the NAT port on the invite. Any thoughts?


I don't think you should have to use the public ip address and port. I would think you should be able to send the INVITE to the private ip address and port, as shown in the contact header of the register. If the TA is using sip port 5061, then you would need to include that in the sip proxy you configure in the xten (e.g., x.x.x.x:5061), did you try that ?

Also, the xten has the ability to generate a sip trace of its own. You can find that under the Advanced system settings, diagnostic page. It might be useful to verify the INVITE you are sending out. If you want to post the REGISTER that the ATA sends, and the INVITE that the xten sends, I might be able to help more
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scerruti
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PostPosted: Mon Feb 07, 2005 12:02 pm    Post subject: Reply with quote Back to top

First of all, I appreciate the assistance you have provided. I haven't worked with SIP before and I have been taking shortcuts on my research just to get an idea whether this type of connection is possible.

I ran the tests again to capture the packets. The ATA is simply not responding to my INVITE requests.

The request line I am generating is identical to the request line Vonage is generating. It is being delivered to the correct IP/port for the ATA. I had intended to capture packets with an invite to a sip address using the internal IP of the ATA but I forgot. I have tested it without any success.

Configuration Notes:
In order to capture packets, I connect my vt1000 to my laptop and share my wireless connection.

That results in

Router (24.131.13.118 -> 192.168.2.1) wireless Laptop (192.168.2.2 -> 192.168.0.1) cat5 VT1000 (ATA) 192.168.0.134


Here are the packets captured by ethereal;

This is the register;

Code:
No.     Time        Source                Destination           Protocol Info
     92 31.734660   192.168.0.134         216.115.25.171        SIP      Request: REGISTER sip:bspgroup1.bsp.vonage.net:5061
 
Frame 92 (669 bytes on wire, 669 bytes captured)
Ethernet II, Src: 00:0c:e5:c8:60:a0, Dst: 00:40:d0:2a:8e:5a
Internet Protocol, Src Addr: 192.168.0.134 (192.168.0.134), Dst Addr: 216.115.25.171 (216.115.25.171)
User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5061 (5061)
Session Initiation Protocol
    Request-Line: REGISTER sip:bspgroup1.bsp.vonage.net:5061 SIP/2.0
    Message Header
        From: "804-746-3268"<sip:18047463268@bspgroup1.bsp.vonage.net:5061;user=phone>;tag=c0a80086-13c5-42079126-650a-5d8f
        To: <sip:18047463268@bspgroup1.bsp.vonage.net:5061;user=phone>
        Call-ID: 9452c59c-16955-1107792166-1552-128603335201925800000000-0@192.168.0.134
        CSeq: 1 REGISTER
        Via: SIP/2.0/UDP 192.168.0.134:5061;branch=z9hG4bK-42079126-650a-181c
        User-Agent: <Motorola VT1000 mac: 000CE5C860A0 sw:VT20_01.2.d4 ln:0 cfg:1107503338972/1002156384>
        Max-Forwards: 70
        Contact: <sip:18047463268@192.168.0.134:5061;transport=UDP;user=phone>
        Expires: 20
        Content-Length: 0


This is the invite generated when I rang my Vonage phone with my cell.

Code:
No.     Time        Source                Destination           Protocol Info
     98 49.767228   216.115.25.171        192.168.0.134         SIP/SDP  Request: INVITE sip:18047463268@24.131.13.118:61193, with session description
 
Frame 98 (1012 bytes on wire, 1012 bytes captured)
Ethernet II, Src: 00:40:d0:2a:8e:5a, Dst: 00:0c:e5:c8:60:a0
Internet Protocol, Src Addr: 216.115.25.171 (216.115.25.171), Dst Addr: 192.168.0.134 (192.168.0.134)
User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5061 (5061)
Session Initiation Protocol
    Request-Line: INVITE sip:18047463268@24.131.13.118:61193 SIP/2.0
    Message Header
        Via: SIP/2.0/UDP 216.115.25.171:5061
        Via: SIP/2.0/UDP 216.115.25.231:5060
        Via: SIP/2.0/UDP 208.51.18.11:5060;branch=z9hG4bKA7
        Record-Route: <sip:18047463268@216.115.25.171:5061>
        Record-Route: <sip:18047463268@216.115.25.231:5060>
        From: "WIRELESS CALLER" <sip:18042441342@208.51.18.11>;tag=783034341
        To: <sip:18047463268@atlas-east.vonage.net>
        Call-ID: 95AE360D-785811D9-85EFF76F-65E8B735@208.51.18.11
        CSeq: 101 INVITE
        Contact: <sip:18042441342@208.51.18.11:5060>;rtpupdated=-
        Max-Forwards: 13
        Content-Type: application/sdp
        Content-Length:   358
    Message body
 
     98 49.767228   216.115.25.171        192.168.0.134         SIP/SDP  Request: INVITE 
     99 49.780861   192.168.0.134         216.115.25.171        SIP      Status: 100 Trying
    100 49.791031   192.168.0.134         216.115.25.171        SIP      Status: 180 Ringing
    101 52.582725   216.115.25.171        192.168.0.134         SIP      Request: CANCEL 
    102 52.592729   192.168.0.134         216.115.25.171        SIP      Status: 200 OK
    103 52.595822   192.168.0.134         216.115.25.171        SIP      Status: 487 Request Terminated


Invite generated by softphone

Code:
No.     Time        Source                Destination           Protocol Info
      3 0.342423    192.168.0.1           192.168.0.134         SIP/SDP  Request: INVITE sip:18047463268@24.131.13.118:61193, with session description
 
Frame 3 (682 bytes on wire, 682 bytes captured)
Ethernet II, Src: 00:40:d0:2a:8e:5a, Dst: 00:0c:e5:c8:60:a0
Internet Protocol, Src Addr: 192.168.0.1 (192.168.0.1), Dst Addr: 192.168.0.134 (192.168.0.134)
User Datagram Protocol, Src Port: 5062 (5062), Dst Port: 5061 (5061)
Session Initiation Protocol
    Request-Line: INVITE sip:18047463268@24.131.13.118:61193 SIP/2.0
    Message Header
        Call-ID: 424ee63b4b44dcda60213e110e914e7b@192.168.0.1
        CSeq: 1 INVITE
        From: "Stephen P. Cerruti" <sip:18047463268@192.168.2.2:5062;transport=udp>;tag=12570890
        To: <sip:18047463268@24.131.13.118:61193>
        Via: SIP/2.0/UDP 192.168.0.1:5062;branch=z9hG4bK2cde7d03174dbe44962889964efaa30d
        Max-Forwards: 70
        Contact: "Stephen P. Cerruti" <sip:24.131.13.118:5062;transport=udp>
        Content-Type: application/sdp
        Content-Length: 160
    Message body
 
      1 0.000000    192.168.0.134         216.115.25.171        SIP      Request: REGISTER 
      2 0.031547    216.115.25.171        192.168.0.134         SIP      Status: 200 OK     
      3 0.342423    192.168.0.1           192.168.0.134         SIP/SDP  Request: INVITE 
      4 0.798574    192.168.0.1           192.168.0.134         SIP/SDP  Request: INVITE 
      5 1.809729    192.168.0.1           192.168.0.134         SIP/SDP  Request: INVITE 
      6 3.802264    192.168.0.1           192.168.0.134         SIP/SDP  Request: INVITE 
      7 7.808070    192.168.0.1           192.168.0.134         SIP/SDP  Request: INVITE 
      8 11.813838   192.168.0.1           192.168.0.134         SIP/SDP  Request: INVITE 
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sajer
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PostPosted: Mon Feb 07, 2005 12:18 pm    Post subject: Reply with quote Back to top

I don't think you want to send the INVITE to the nat'ed public address and port. Depending on the type of firewall, the router is only routing packets according to that mapping for the specific remote Vonage server. So your packets got discarded by the router/firewall and never reached the ATA.

So I think you want to send the INVITE to the private address and port advertised in the REGISTER. In other words, the INVITE you send should be:

INVITE sip:18047463268@192.168.0.134:5061

However, I think you are saying you tried that as well and it didn't work ?

Now, this assumes that the PC you are sending from has connectivity to the ATA, and I see that the PC you are sending from is not on the 192.168.0.0 network -- from the INVITE it sends it looks it has an ip address of 24.131.13.118. I'm guessing that either the PC is on a different network than the ATA, or it is on the same network but you have nat support turned on in the xten configuration. If the latter, turn off NAT support in the xten configuration. Or is the xten truly on a different network?
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scerruti
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Posts: 1424
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PostPosted: Mon Feb 07, 2005 1:04 pm    Post subject: Reply with quote Back to top

sajer wrote:
I don't think you want to send the INVITE to the nat'ed public address and port.


Actually the invite is not being routed through the router, but since the ATA is directly attached to the laptop the packets are being delivered directly to the ATA. You are correct, my attempts to send the packets via the public address, even with port forwarding on, were fruitless.

sajer wrote:
So I think you want to send the INVITE to the private address and port advertised in the REGISTER. In other words, the INVITE you send should be:

INVITE sip:18047463268@192.168.0.134:5061

However, I think you are saying you tried that as well and it didn't work ?


Yes, but I didn't catch the packets so I am unsure if the ATA responded to those, however I was unable to establish a call. Basically I am able to control the proxy and the domain in the SIP address independently so I have been able to try all of the permutations. I am only able to route packets to the ATA when the SIP proxy is set to the internal address of the ATA.

sajer wrote:
If the [the PC is on a different network], turn off NAT support in the xten configuration. Or is the xten truly on a different network?


I actually gave up on xten and am using some demonstration code (sip-communicator) becuase it provides more control and diagnostic information. I got annoyed with xten (X-Lite) because when I have misconfigured the SIP Proxy is doesn't tell you why, it just pops up the configuration sheet.
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