[OpenSIPS-Users] 408 Request Timeout with UDP

[Bogdan-Andrei Iancu]

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:pinger at xx.xx.xx.xx")

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" <bogdan at opensips.org 
> <mailto:bogdan at opensips.org>> 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 <govoiper at gmail.com
>>     <mailto:govoiper at gmail.com>> 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
>>         <nabeelshikder at gmail.com <mailto:nabeelshikder at gmail.com>> 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
>>             (username at domain.com <mailto:username at domain.com>). 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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