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Thu Mar 26 14:29:42 CET 2015
public IP for callee. Can you do a small test ? Re register your callee
phone and within same second make a call to callee. I'm having doubts about
a firewall closing the connection between the Callee UA and the OpenSIPS.
On Aug 6, 2015 8:00 AM, "Nabeel" <nabeelshikder at gmail.com> 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> 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 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". 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? 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> 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
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>>
>> _______________________________________________
>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable
<div dir=3D"ltr"><p dir=3D"ltr">Im afraid thats not how this works, regardi=
ng your assumptions in last paragraph.=C2=A0</p><p dir=3D"ltr">The NAT rela=
ted parameters get saved in location table as OpenSIPS sets flags and upon =
a call to that user those are used immediately and no timer is used as such=
.=C2=A0</p><p dir=3D"ltr">Now at the time of signalling rtpproxy has no rol=
e to play to work NAT or anything.=C2=A0 RTPproxy will just use IP addresse=
s in SDP as instructed in the script to have a media path.=C2=A0</p><p>From=
the SIP trace you shared it do looks like its sending to the right public =
IP for callee. Can you do a small test ? Re register your callee phone and =
within same second make a call to callee. I'm having doubts about a fir=
ewall closing the connection between the Callee UA and the OpenSIPS.=C2=A0<=
/p><p><br></p>
<div class=3D"gmail_quote">On Aug 6, 2015 8:00 AM, "Nabeel" <<=
a href=3D"mailto:nabeelshikder at gmail.com" target=3D"_blank">nabeelshikder at g=
mail.com</a>> wrote:<br type=3D"attribution"><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex"><p dir=3D"ltr">My OpenSIPS runs on a public IP.=C2=A0 The callee was co=
nnected to Wi-Fi in my first test earlier, but in the second test the calle=
e was connected to a public IP=C2=A0 (public mobile network).=C2=A0 In both=
cases, the same '404 timeout' error occurred on call attempt.=C2=
=A0 The SIP trace for the second case is at this link: </p>
<p dir=3D"ltr"><a href=3D"http://pastebin.com/jGxRQ34q" target=3D"_blank">h=
ttp://pastebin.com/jGxRQ34q</a></p>
<p dir=3D"ltr">Regarding private IP, you said it's impossible to route =
from public IP to private IP.=C2=A0 Although at the IP level this may be tr=
ue, 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 t=
hat not right?=C2=A0 I'm fact,=C2=A0 using a TURN server and a differen=
t SIP proxy, I was able to connect these same devices under the same networ=
ks, so I know this should be possible.=C2=A0 I feel something is not config=
ured correctly in OpenSIPS / rtpproxy.</p>
<p dir=3D"ltr">I did "opensipsctl ul show" and the results seem n=
ormal; please check it:</p>
<p dir=3D"ltr"><a href=3D"http://pastebin.com/n1BbTuMK" target=3D"_blank">h=
ttp://pastebin.com/n1BbTuMK</a></p>
<p dir=3D"ltr">Perhaps the NAT processing just needs a bit more time; in th=
ar case what are the config options to increase the request timeout for UDP=
?=C2=A0 I have seen the 'tcp_send_timeout' and 'tcp_connect_tim=
eout' options for TCP, but please let me know if there are similar opti=
ons for UDP.</p>
<div class=3D"gmail_quote">On 6 Aug 2015 12:08, "Bogdan-Andrei Iancu&q=
uot; <<a href=3D"mailto:bogdan at opensips.org" target=3D"_blank">bogdan at op=
ensips.org</a>> wrote:<br type=3D"attribution"><blockquote class=3D"gmai=
l_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left=
:1ex">
=20
=20
=20
<div text=3D"#000000" bgcolor=3D"#FFFFFF">
<tt>Nabeel,<br>
<br>
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.<br>
<br>
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 =
?<br>
<br>
Regards,<br>
</tt>
<pre cols=3D"72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a href=3D"http://www.opensips-solutions.com" target=3D"_blank">http://www.=
opensips-solutions.com</a></pre>
<div>On 06.08.2015 07:14, Nabeel wrote:<br>
</div>
<blockquote type=3D"cite">
<div dir=3D"ltr">Hi,
<div><br>
</div>
<div>Yes, the destination IP is=C2=A0<span style=3D"font-size:12.80=
00001907349px">=C2=A0</span><span><a href=3D"http://192.168.0.19:60912/" ta=
rget=3D"_blank">192.168.0.19:60912</a>=C2=A0and both phones are
registered to OpenSIPS.=C2=A0 In this case, the callee is
connected to Wi-Fi (hence 192.xx IP address) and the caller
is connected to a mobile network.</span></div>
<div><span><br>
</span></div>
<div><font color=3D"#000000" face=3D"Consolas, Menlo, Monaco, Lucid=
a
Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera
Sans Mono, monospace, serif"><span style=3D"font-size:12px;line=
-height:21px">The opensips.cfg I
am using was generated from 'make menuconfig', except=
with
the addition of "alias=3D<a href=3D"http://domain.com" t=
arget=3D"_blank">domain.com</a>". I have
attached my config file at this link:</span></font></div>
<div><font color=3D"#000000" face=3D"Consolas, Menlo, Monaco, Lucid=
a
Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera
Sans Mono, monospace, serif"><span style=3D"font-size:12px;line=
-height:21px"><br>
</span></font></div>
<div><font color=3D"#000000" face=3D"Consolas, Menlo, Monaco, Lucid=
a
Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera
Sans Mono, monospace, serif"><span style=3D"font-size:12px;line=
-height:21px"><a href=3D"http://pastebin.com/0QRyC938" target=3D"_blank">ht=
tp://pastebin.com/0QRyC938</a></span><br>
</font></div>
<div><font color=3D"#000000" face=3D"Consolas, Menlo, Monaco, Lucid=
a
Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera
Sans Mono, monospace, serif"><span style=3D"font-size:12px;line=
-height:21px"><br>
</span></font></div>
<div><font color=3D"#000000" face=3D"Consolas, Menlo, Monaco, Lucid=
a
Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera
Sans Mono, monospace, serif"><span style=3D"font-size:12px;line=
-height:21px"><br>
</span></font></div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On 6 August 2015 at 05:00, SamyGo <span =
dir=3D"ltr"><<a href=3D"mailto:govoiper at gmail.com" target=3D"_blank">gov=
oiper at gmail.com</a>></span>
wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bord=
er-left:1px #ccc solid;padding-left:1ex">
<div dir=3D"ltr">Hi Nabeel,
<div>Quick question; what is this destination ip?=C2=A0<span>=
<a href=3D"http://192.168.0.19:60912" target=3D"_blank">192.168.0.19:60912<=
/a>
?</span><span>=C2=A0-=C2=A0</span>Destination
User Agent Registered on OpenSIPS?</div>
<div>Can you share the opensips.cfg code snippet for this
call ?</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">
<div>
<div>On Wed, Aug 5, 2015 at 11:55 PM,
Nabeel <span dir=3D"ltr"><<a href=3D"mailto:nabeelsh=
ikder at gmail.com" target=3D"_blank">nabeelshikder at gmail.com</a>></span>
wrote:<br>
</div>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8e=
x;border-left:1px #ccc solid;padding-left:1ex">
<div>
<div>
<div dir=3D"ltr">
<div>Hi,</div>
<div><br>
</div>
<div>I am using the residential script generated
by 'make menuconfig', with UDP and NAT su=
pport
enabled.=C2=A0 I added "alias=3D<a href=3D"h=
ttp://domain.com" target=3D"_blank">domain.com</a>"
to the config because otherwise the UA did not
register with my domain (<a href=3D"mailto:userna=
me at domain.com" target=3D"_blank">username at domain.com</a>).=C2=A0When
I attempt to make a call, I see '408 Request
Timeout' in the sip trace and the call does
not connect.=C2=A0 Please check the log/trace bel=
ow
and advise how to fix this.</div>
<div><br>
</div>
<div>SIP trace:</div>
<div><br>
</div>
<a href=3D"http://pastebin.com/u5h9qGNr" target=3D"=
_blank">http://pastebin.com/u5h9qGNr</a><br>
<div><br>
</div>
<div>OpenSIPS log:=C2=A0</div>
<div><br>
</div>
<div><a href=3D"http://pastebin.com/B8PUCKh0" targe=
t=3D"_blank">http://pastebin.com/B8PUCKh0</a><br>
</div>
</div>
<br>
</div>
</div>
_______________________________________________<br>
Users mailing list<br>
<a href=3D"mailto:Users at lists.opensips.org" target=3D"_bl=
ank">Users at lists.opensips.org</a><br>
<a href=3D"http://lists.opensips.org/cgi-bin/mailman/list=
info/users" rel=3D"noreferrer" target=3D"_blank">http://lists.opensips.org/=
cgi-bin/mailman/listinfo/users</a><br>
<br>
</blockquote>
</div>
<br>
</div>
<br>
_______________________________________________<br>
Users mailing list<br>
<a href=3D"mailto:Users at lists.opensips.org" target=3D"_blank">U=
sers at lists.opensips.org</a><br>
<a href=3D"http://lists.opensips.org/cgi-bin/mailman/listinfo/u=
sers" rel=3D"noreferrer" target=3D"_blank">http://lists.opensips.org/cgi-bi=
n/mailman/listinfo/users</a><br>
<br>
</blockquote>
</div>
<br>
</div>
<br>
<fieldset></fieldset>
<br>
<pre>_______________________________________________
Users mailing list
<a href=3D"mailto:Users at lists.opensips.org" target=3D"_blank">Users at lists.o=
pensips.org</a>
<a href=3D"http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target=
=3D"_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a>
</pre>
</blockquote>
<br>
</div>
</blockquote></div>
<br>_______________________________________________<br>
Users mailing list<br>
<a href=3D"mailto:Users at lists.opensips.org" target=3D"_blank">Users at lists.o=
pensips.org</a><br>
<a href=3D"http://lists.opensips.org/cgi-bin/mailman/listinfo/users" rel=3D=
"noreferrer" target=3D"_blank">http://lists.opensips.org/cgi-bin/mailman/li=
stinfo/users</a><br>
<br></blockquote></div>
</div>
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