[OpenSIPS-Users] 408 Request Timeout with UDP

Nabeel nabeelshikder at gmail.com
Thu Aug 6 13:59:46 CEST 2015

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:


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:


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> 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 Developerhttp://www.opensips-solutions.com
> On 06.08.2015 07:14, Nabeel wrote:
> Hi,
> Yes, the destination IP is 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". 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> wrote:
>> Hi Nabeel,
>> Quick question; what is this destination ip? ? - 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> wrote:
>>> Hi,
>>> I am using the residential script generated by 'make menuconfig', with
>>> UDP and NAT support enabled.  I added "alias=domain.com" to the config
>>> because otherwise the UA did not register with my domain (
>>> 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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