Hi Nabeel,

This time the SIP trace looks ok - the message is sent to a public IP and not a private one (as shown in the prev capture). Also the usrloc data looks ok, telling that the NAT traversal works ok.
As you have Mobile Data, some operators do filter SIP, so not sure what say.

Do you see the sip messages going back and forward during the registration process ?

An issue may be the fact the default script does non-SIP pinging (which is unidirectional), so the NAT may close. Add the followings:
1) modparam("nathelper", "sipping_bflag", "SIP_PING_FLAG")
modparam("nathelper", "sipping_from", "sip:[email protected]")

2) setbflag(SIP_PING_FLAG); before doing save()

You should see OpenSIPS doing keep alive with OPTIONS requests.
Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 06.08.2015 14:59, Nabeel wrote:

My OpenSIPS runs on a public IP. The callee was connected to Wi-Fi in my first test earlier, but in the second test the callee was connected to a public IP (public mobile network). In both cases, the same '404 timeout' error occurred on call attempt. The SIP trace for the second case is at this link:

http://pastebin.com/jGxRQ34q

Regarding private IP, you said it's impossible to route from public IP to private IP. Although at the IP level this may be true, even if the user is on Wi-Fi, the whole point of NAT traversal is that the user's public IP is discovered and the call can get connected, is that not right? I'm fact, using a TURN server and a different SIP proxy, I was able to connect these same devices under the same networks, so I know this should be possible. I feel something is not configured correctly in OpenSIPS / rtpproxy.

I did "opensipsctl ul show" and the results seem normal; please check it:

http://pastebin.com/n1BbTuMK

Perhaps the NAT processing just needs a bit more time; in thar case what are the config options to increase the request timeout for UDP? I have seen the 'tcp_send_timeout' and 'tcp_connect_timeout' options for TCP, but please let me know if there are similar options for UDP.

On 6 Aug 2015 12:08, "Bogdan-Andrei Iancu" <[email protected] <mailto:[email protected]>> wrote:

    Nabeel,

    I suppose you OpenSIPS seats on a public IP, right ? The callee
    looks to have a private IP. And, at IP level, it is impossible to
    route from a public IP to a private one.

    I see your script has NAT traversal support. My question is - did
    the callee properly registered via this script ? can you do an
    "opensipsctl ul show" to see the callee's registration ?

    Regards,

    Bogdan-Andrei Iancu
    OpenSIPS Founder and Developer
    http://www.opensips-solutions.com

    On 06.08.2015 07:14, Nabeel wrote:
    Hi,

    Yes, the destination IP is 192.168.0.19:60912
    <http://192.168.0.19:60912/> and both phones are registered to
    OpenSIPS.  In this case, the callee is connected to Wi-Fi (hence
    192.xx IP address) and the caller is connected to a mobile network.

    The opensips.cfg I am using was generated from 'make menuconfig',
    except with the addition of "alias=domain.com
    <http://domain.com>". I have attached my config file at this link:

    http://pastebin.com/0QRyC938



    On 6 August 2015 at 05:00, SamyGo <[email protected]
    <mailto:[email protected]>> wrote:

        Hi Nabeel,
        Quick question; what is this destination ip?
        192.168.0.19:60912 <http://192.168.0.19:60912> ? -
        Destination User Agent Registered on OpenSIPS?
        Can you share the opensips.cfg code snippet for this call ?

        On Wed, Aug 5, 2015 at 11:55 PM, Nabeel
        <[email protected] <mailto:[email protected]>> wrote:

            Hi,

            I am using the residential script generated by 'make
            menuconfig', with UDP and NAT support enabled.  I added
            "alias=domain.com <http://domain.com>" to the config
            because otherwise the UA did not register with my domain
            ([email protected] <mailto:[email protected]>). When
            I attempt to make a call, I see '408 Request Timeout' in
            the sip trace and the call does not connect.  Please
            check the log/trace below and advise how to fix this.

            SIP trace:

            http://pastebin.com/u5h9qGNr

            OpenSIPS log:

            http://pastebin.com/B8PUCKh0

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