No subject
Thu Dec 6 16:59:43 CET 2012
shown below.
In the example dialog below at step 7 when the server sends the "200 -
Answering" back to voip2 it includes its local IP address in the
"Record-Route" field. This results in voip2 trying to sending its ACK to the
server's local IP address (step 8) which of course cannot be reached. Voip2
now starts sending RTP's to voip1 (which actually do reach voip1). However,
voip1 does not start sending RTP's to voip2. Instead voip1 just keeps
sending "200 Answering" messages, which I assume is because it has not
received the ACK for its "200 -Answering" message (step 6). I am also
assuming that voip1 does not get the final ACK from the server because the
server never gets the final ACK from voip2 at step 8.
Example dialog:
1. server <-INVITE-- voip2
2. server -100 Trying-> voip2
3. voip1 <-INVITE-- server
4. voip1 -180 Ringing-> server
5. server -200 Ringing-> voip2
6. voip1 -200 Answering-> server
7. server -200 Answering-> voip2
8. nowhere <-ACK-- voip2
9. voip1 <-----RTP------------------------RTP--- voip2
10.voip1 <-----RTP------------------------RTP--- voip2
11.voip1 <-----RTP------------------------RTP--- voip2
12.voip1 -200 Answering-> server
13.voip1 <-----RTP------------------------RTP--- voip2
14.voip1 <-----RTP------------------------RTP--- voip2
15.voip1 <-----RTP------------------------RTP--- voip2
...
Hope you can help.
Regards
Mark Currie
From: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] On Behalf Of Stefano Pisani
Sent: 04 January 2013 09:10
To: users at lists.opensips.org
Subject: Re: [OpenSIPS-Users] NAT issues on client and server
Could you explaint your scenario better?
The server is in a private network?
The cliets are in different private networks?
s
Il 04/01/2013 05:23, Mark Currie ha scritto:
Thanks for the advice Flavio.
Currently I am actually pretty close with my NAT'ed OpenSIPS and NAT'ed
clients. I am assuming this is because I have a simple NAT scenario with
full-cone NATs.
I get as far as making a call, but when I pick-up the call only one side
gets RTP through. I traced the problem to the fact that a local address is
being put in the "Record Route" section of an ACK response.
I would like to try to get this more controllable scenario working first
before I launch into new complexity with RTP proxies etc. I think that I
just need to know how to get the fix_nat_sip() and fix_nat_contact() to
work. Any pointers with that?
Regards
Mark
From: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] On Behalf Of Flavio Goncalves
Sent: 03 January 2013 12:27
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] NAT issues on client and server
Hi Mark,
There is no simple way to traverse NAT. Unless all your routers use a non
symmetric NAT, you will have to use rtpproxy or mediaproxy (you can check
this with a stun client). OpenSIPS behind NAT make things even more
complicated. So I suggest that you follow an example with rtpproxy or media
proxy and also use the OpenSIPS in a valid IP address. The setup you are
trying to do with OpenSIPS behind NAT is possible, but it is even more
complex.
Flavio E. Goncalves
2013/1/3 Mark Currie <mark at ziliant.com>
Hi,
I have a very simple setup for a closed network of users (all NAT'ed) with
one OpenSIPS server (also NAT'ed).
I have managed to solve my first problem with registration by following
previous posts and using fix_nat_register(), but I am still having problems
with NAT issues during a call. I know that I probably need to use
fix_nat_contact() and fix_nat_sip() but I can't figure out how to use these
properly through the documentation.
I have tried to search for examples of opensips.cfg that suit my scenario
but all the ones I found are complicated with proxies etc. Can someone point
me to simple example of opensips.cfg that takes care of NAT?
Regards
Mark Currie
_______________________________________________
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
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<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body bgcolor=3Dwhite =
lang=3DEN-ZA link=3Dblue vlink=3Dpurple><div class=3DWordSection1><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Hi Stefano,<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p> </o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Thanks for your interest.<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p> </o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>There is no private network such as a VPN being used. I am testing =
using a virtual cloud server in a CaaS environment which puts it behind =
a NAT firewall (although later the server may have a fixed IP address). =
The clients are currently simple console apps (based on Baresip) =
operating on PC’s behind a NAT/firewall on a LAN. Later these =
clients may operate on a mobile phone, in which case the cellular =
network will be NAT’ed.<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p> </o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Currently, the only mods I have made to the default OpenSIPS (Vers. =
1.6.4.2) config file is “alias=3D<server IP address>” =
and fix_nat_register() related mods. However I actually get quite far =
with these mods as you can see from my dialog =
below.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p> </o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Network architecture:<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Server =3D Ubuntu 10.04 LTS server with OpenSIPS =
1.6.4.2<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p> </o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Courier New";color:#1F497D'>LAN =
clients--NAT------Internet------NAT--OpenSIPS<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p> </o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>From my testing and wireshark traces I have constructed an example =
dialog shown below.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p> </o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>In the example dialog below at step 7 when the server sends the =
“200 – Answering” back to voip2 it includes its local =
IP address in the "Record-Route" field. This results in voip2 =
trying to sending its ACK to the server’s local IP address (step =
8) which of course cannot be reached. Voip2 now starts sending =
RTP’s to voip1 (which actually do reach voip1). However, voip1 =
does not start sending RTP’s to voip2. Instead voip1 just keeps =
sending "200 Answering" messages, which I assume is because it =
has not received the ACK for its “200 –Answering” =
message (step 6). I am also assuming that voip1 does not get the final =
ACK from the server because the server never gets the final ACK from =
voip2 at step 8.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p> </o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Example dialog:<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p> </o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Courier New";color:#1F497D'>1. =
&=
nbsp; server&n=
bsp; <-INVITE-- =
voip2<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Courier New";color:#1F497D'>2. =
&=
nbsp; server&n=
bsp; -100 Trying-> voip2<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Courier =
New";color:#1F497D'>3. voip1 =
<-INVITE-- server<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Courier =
New";color:#1F497D'>4. voip1 -180 Ringing-> =
server<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Courier New";color:#1F497D'>5. =
&=
nbsp; server&n=
bsp; -200 Ringing-> voip2<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Courier =
New";color:#1F497D'>6. voip1 -200 Answering-> =
server<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Courier New";color:#1F497D'>7. =
&=
nbsp; server =
-200 Answering-> voip2<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Courier =
New";color:#1F497D'>8. =
&=
nbsp; nowhere&=
nbsp; <-ACK-- =
voip2<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Courier New";color:#1F497D'>9. =
voip1 <-----RTP------------------------RTP--- =
voip2<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Courier =
New";color:#1F497D'>10.voip1 =
<-----RTP------------------------RTP--- voip2<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Courier =
New";color:#1F497D'>11.voip1 =
<-----RTP------------------------RTP--- voip2<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Courier =
New";color:#1F497D'>12.voip1 -200 Answering-> =
server<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Courier =
New";color:#1F497D'>13.voip1 =
<-----RTP------------------------RTP--- voip2<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Courier =
New";color:#1F497D'>14.voip1 =
<-----RTP------------------------RTP--- voip2<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Courier =
New";color:#1F497D'>15.voip1 =
<-----RTP------------------------RTP--- voip2<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Courier =
New";color:#1F497D'>...<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Courier =
New";color:#1F497D'><o:p> </o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Hope you can help.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Regards<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Mark Currie</span><span =
style=3D'font-size:11.0pt;font-family:"Courier =
New";color:#1F497D'><o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p> </o:p></span></p><div =
style=3D'border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm =
4.0pt'><div><div style=3D'border:none;border-top:solid #B5C4DF =
1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=3DMsoNormal><b><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowt=
ext'>From:</span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowt=
ext'> users-bounces at lists.opensips.org =
[mailto:users-bounces at lists.opensips.org] <b>On Behalf Of </b>Stefano =
Pisani<br><b>Sent:</b> 04 January 2013 09:10<br><b>To:</b> =
users at lists.opensips.org<br><b>Subject:</b> Re: [OpenSIPS-Users] NAT =
issues on client and server<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p> </o:p></p><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'>Could you explaint your scenario =
better?<br>The server is in a private network?<br>The cliets are in =
different private networks?<br><br>s<o:p></o:p></p><div><p =
class=3DMsoNormal>Il 04/01/2013 05:23, Mark Currie ha =
scritto:<o:p></o:p></p></div><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Thanks for the advice Flavio.</span><o:p></o:p></p><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'> </span><o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Currently I am actually pretty close with my NAT’ed OpenSIPS =
and NAT’ed clients. I am assuming this is because I have a simple =
NAT scenario with full-cone NATs.</span><o:p></o:p></p><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'> </span><o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>I get as far as making a call, but when I pick-up the call only one =
side gets RTP through. I traced the problem to the fact that a local =
address is being put in the “Record Route” section of an ACK =
response.</span><o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'> </span><o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>I would like to try to get this more controllable scenario working =
first before I launch into new complexity with RTP proxies etc. I think =
that I just need to know how to get the fix_nat_sip() and =
fix_nat_contact() to work. Any pointers with =
that?</span><o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'> </span><o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Regards</span><o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Mark</span><o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'> </span><o:p></o:p></p><div =
style=3D'border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm =
4.0pt'><div><div style=3D'border:none;border-top:solid #B5C4DF =
1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=3DMsoNormal><b><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> <a =
href=3D"mailto:users-bounces at lists.opensips.org">users-bounces at lists.open=
sips.org</a> [<a =
href=3D"mailto:users-bounces at lists.opensips.org">mailto:users-bounces at lis=
ts.opensips.org</a>] <b>On Behalf Of </b>Flavio =
Goncalves<br><b>Sent:</b> 03 January 2013 12:27<br><b>To:</b> OpenSIPS =
users mailling list<br><b>Subject:</b> Re: [OpenSIPS-Users] NAT issues =
on client and server</span><o:p></o:p></p></div></div><p =
class=3DMsoNormal> <o:p></o:p></p><p class=3DMsoNormal>Hi =
Mark, <o:p></o:p></p><div><p =
class=3DMsoNormal> <o:p></o:p></p></div><div><p =
class=3DMsoNormal>There is no simple way to traverse NAT. Unless all =
your routers use a non symmetric NAT, you will have to use rtpproxy or =
mediaproxy (you can check this with a stun client). OpenSIPS behind NAT =
make things even more complicated. So I suggest that you follow an =
example with rtpproxy or media proxy and also use the OpenSIPS in a =
valid IP address. The setup you are trying to do with OpenSIPS behind =
NAT is possible, but it is even more =
complex. <o:p></o:p></p></div><div><p class=3DMsoNormal><br =
clear=3Dall><o:p></o:p></p><div><div><p class=3DMsoNormal>Flavio E. =
Goncalves<o:p></o:p></p></div><div><p =
class=3DMsoNormal> <o:p></o:p></p></div></div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'> <o:p></o:p></p><div><p =
class=3DMsoNormal>2013/1/3 Mark Currie <<a =
href=3D"mailto:mark at ziliant.com" =
target=3D"_blank">mark at ziliant.com</a>><o:p></o:p></p><p =
class=3DMsoNormal>Hi,<br><br>I have a very simple setup for a closed =
network of users (all NAT'ed) with<br>one OpenSIPS server (also =
NAT'ed).<br><br>I have managed to solve my first problem with =
registration by following<br>previous posts and using =
fix_nat_register(), but I am still having problems<br>with NAT issues =
during a call. I know that I probably need to use<br>fix_nat_contact() =
and fix_nat_sip() but I can't figure out how to use these<br>properly =
through the documentation.<br><br>I have tried to search for examples of =
opensips.cfg that suit my scenario<br>but all the ones I found are =
complicated with proxies etc. Can someone point<br>me to simple example =
of opensips.cfg that takes care of NAT?<br><br>Regards<br>Mark =
Currie<br><br><br>_______________________________________________<br>User=
s mailing list<br><a =
href=3D"mailto:Users at lists.opensips.org">Users at lists.opensips.org</a><br>=
<a href=3D"http://lists.opensips.org/cgi-bin/mailman/listinfo/users" =
target=3D"_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/user=
s</a><o:p></o:p></p></div><p =
class=3DMsoNormal> <o:p></o:p></p></div></div><p =
class=3DMsoNormal><br><br><br><o:p></o:p></p><pre>_______________________=
________________________<o:p></o:p></pre><pre>Users mailing =
list<o:p></o:p></pre><pre><a =
href=3D"mailto:Users at lists.opensips.org">Users at lists.opensips.org</a><o:p=
></o:p></pre><pre><a =
href=3D"http://lists.opensips.org/cgi-bin/mailman/listinfo/users">http://=
lists.opensips.org/cgi-bin/mailman/listinfo/users</a><o:p></o:p></pre></b=
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