subject:"Re\: \[OpenSIPS\-Users\] NAT problem"

Re: [OpenSIPS-Users] NAT problem

2015-06-30 Thread xiaofeng

Hi,

Thanks for the reply.

Manually insert header field Record-Route by
insert_hf(Record-Route: sip:106.x.x.x\r\n);

And, change IP in SDP

if (has_body(application/sdp)) {
replace_all(IN IP4 [0-9]\.[0-9]\.[0-9]\.[0-9], 106.x.x.x);
}

Seems work with WIFI behind one level NAT, not test 3G/4G.



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Re: [OpenSIPS-Users] NAT problem

2015-06-26 Thread Ionut Muntean


Nice ngrep invocation. ;)

On 26/06/2015 19:26, Terrance Devor wrote:

ngrep -d eth0 -qt -W byline portrange 5060



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Re: [OpenSIPS-Users] NAT problem

2015-06-26 Thread Terrance Devor

Attach SIP signalling pls.
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Re: [OpenSIPS-Users] NAT problem

2015-06-26 Thread Terrance Devor

ngrep -d eth0 -qt -W byline portrange 5060

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Re: [OpenSIPS-Users] NAT problem on 5060

2014-11-21 Thread amirehsan

in last days i change opensips port to 5090 

now 

i want change 5090 port to default port 5060 

i try it , so zoiper ip phone can not translate voice and X_light eyebeam
worked ok 

please help me 



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Re: [OpenSIPS-Users] Nat Problem

2010-05-06 Thread Bogdan-Andrei Iancu

Hi Ahmed,

check the following things:

1) you do fix_nated_register() before save(location)

2) the received_avp param has the same value in registrar and nathelper 
module

3) you configured the nat_bflag param in usrloc module and you are 
setting it before save(location)

Regards,
Bogdan

Ahmed Munir wrote:
 Hi,

 I've configured OpenSIPs using Nathelper module and rtpproxy. the 
 problem I'm facing is when I try to register my softphone, it got 
 registered but as I issue the command opensipsctl ul show, in contact 
 header the IP is private not public. The configuration of OpenSIPs is 
 listed down below;




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Re: [OpenSIPS-Users] NAT Problem using Nat helper

2010-05-03 Thread Ahmed Munir

Hi,

Thanks for replying. Can you please check my configuration of OpenSIPs what
I posted on my previous digest i.e.Users Digest, Vol 21, Issue 146.

Please point out in which section do I required to add force_rtp_proxy(),
because I already configured Nat on it. kindly advise me soon.

On Fri, Apr 30, 2010 at 11:35 AM, users-requ...@lists.opensips.org wrote:

 Send Users mailing list submissions to
users@lists.opensips.org

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 When replying, please edit your Subject line so it is more specific
 than Re: Contents of Users digest...


 Today's Topics:

   1. Re: NAT Problem using Nat helper (Laszlo)


 --

 Message: 1
 Date: Fri, 30 Apr 2010 08:35:00 +0200
 From: Laszlo las...@voipfreak.net
 Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
 To: OpenSIPS users mailling list users@lists.opensips.org
 Message-ID:
r2za950b2031004292335re52872d5tf60c9830c8c21...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1

 Hi Ahmed,

 As you can see, the other party gets local ip in SDP

 c=IN IP4 192.168.0.168.

 You can try to play with flags:
 http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html#id229028

 -Laszlo



 2010/4/30 Ahmed Munir ahmedmunir...@gmail.com

 
 
  Hi.
 
  Thanks for your reply, the traces are metioned below;
 
  U 203.215.176.22:55134 - 11.22.33.44:5060
  .
  .
  ..
 
  U 81.201.82.45:5060 - 11.22.33.44:5060
  INVITE sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44 
 sip%3a1234...@11.22.33.44 sip%253a1234...@11.22.33.44 SIP/2.0.
  Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
  CSeq: 102 INVITE.
  From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 sip%3a4572727...@voxbone.com sip%253a4572727...@voxbone.com
  ;tag=43772.
  To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44 
 sip%3a1234...@11.22.33.44 sip%253a1234...@11.22.33.44.
  Via: SIP/2.0/UDP 81.201.82.45:5060
  ;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
  Max-Forwards: 69.
  Content-Type: application/sdp.
  Contact: sip:4572727...@81.201.82.45:5060;transport=udp.
  User-Agent: Vox Callcontrol.
  Content-Length: 210.
  .
  v=0.
  o=root 13293 13293 IN IP4 81.201.82.146.
  s=session.
  c=IN IP4 81.201.82.146.
  t=0 0.
  m=audio 11458 RTP/AVP 8 0.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:0 PCMU/8000.
  a=silenceSupp:off - - - -.
  a=ptime:20.
  a=sendrecv.
 
 
  U 11.22.33.44:5060 - 81.201.82.45:5060
  SIP/2.0 100 Giving a try.
  Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
  CSeq: 102 INVITE.
  From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 sip%3a4572727...@voxbone.com sip%253a4572727...@voxbone.com
  ;tag=43772.
  To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44 
 sip%3a1234...@11.22.33.44 sip%253a1234...@11.22.33.44.
  Via: SIP/2.0/UDP 81.201.82.45:5060
  ;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0;rport=5060.
  Server: OpenSIPS (1.6.1-notls (i386/linux)).
  Content-Length: 0.
  .
 
 
  U 11.22.33.44:5060 - 203.215.176.22:55134
  INVITE sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26 SIP/2.0.
  Record-Route: sip:11.22.33.44;lr=on.
  Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
  CSeq: 102 INVITE.
  From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 sip%3a4572727...@voxbone.com sip%253a4572727...@voxbone.com
  ;tag=43772.
  To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44 
 sip%3a1234...@11.22.33.44 sip%253a1234...@11.22.33.44.
  Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK1c7c.78b70285.0.
  Via: SIP/2.0/UDP 81.201.82.45:5060
 
 ;rport=5060;received=81.201.82.45;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
  Max-Forwards: 68.
  Content-Type: application/sdp.
  Contact: sip:4572727...@81.201.82.45:5060;transport=udp.
  User-Agent: Vox Callcontrol.
  Content-Length: 210.
  P-hint: usrloc applied.
  .
  v=0.
  o=root 13293 13293 IN IP4 81.201.82.146.
  s=session.
  c=IN IP4 81.201.82.146.
  t=0 0.
  m=audio 11458 RTP/AVP 8 0.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:0 PCMU/8000.
  a=silenceSupp:off - - - -.
  a=ptime:20.
  a=sendrecv.
 
 
  U 203.215.176.22:55134 - 11.22.33.44:5060
  SIP/2.0 180 Ringing.
  Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK1c7c.78b70285.0.
  Via: SIP/2.0/UDP 81.201.82.45:5060
 
 ;rport=5060;received=81.201.82.45;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
  Record-Route: sip:11.22.33.44;lr.
  Contact: sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26.
  To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44 
 sip%3a1234...@11.22.33.44 sip%253a1234...@11.22.33.44;tag=611cee1e.
  From: 4572727220sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 sip%3a4572727

Re: [OpenSIPS-Users] NAT Problem using Nat helper

2010-05-03 Thread Bogdan-Andrei Iancu

Hi Ahmed,

as a hint, probably you do not handle correctly the case when only the 
callee is nated (caller is public) - for such cases, to see if rtpproxy 
is needed, after the lookup(location) the nat_bflag will will 
automatically set if the callee location is nated - you can use that 
flag to detect the nated callee and to do the nat fixups - force rtpp 
and fix the 200 ok from the callee (SDP and contact).

Regards,
Bogdan

Ahmed Munir wrote:
 Hi,

 Thanks for replying. Can you please check my configuration of OpenSIPs 
 what I posted on my previous digest i.e.Users Digest, Vol 21, Issue 146.

 Please point out in which section do I required to add 
 force_rtp_proxy(), because I already configured Nat on it. kindly 
 advise me soon.

 On Fri, Apr 30, 2010 at 11:35 AM, users-requ...@lists.opensips.org 
 mailto:users-requ...@lists.opensips.org wrote:

 Send Users mailing list submissions to
users@lists.opensips.org mailto:users@lists.opensips.org

 To subscribe or unsubscribe via the World Wide Web, visit
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
 or, via email, send a message with subject or body 'help' to
users-requ...@lists.opensips.org
 mailto:users-requ...@lists.opensips.org

 You can reach the person managing the list at
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 mailto:users-ow...@lists.opensips.org

 When replying, please edit your Subject line so it is more specific
 than Re: Contents of Users digest...


 Today's Topics:

   1. Re: NAT Problem using Nat helper (Laszlo)


 --

 Message: 1
 Date: Fri, 30 Apr 2010 08:35:00 +0200
 From: Laszlo las...@voipfreak.net mailto:las...@voipfreak.net
 Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
 To: OpenSIPS users mailling list users@lists.opensips.org
 mailto:users@lists.opensips.org
 Message-ID:
  
  r2za950b2031004292335re52872d5tf60c9830c8c21...@mail.gmail.com
 mailto:r2za950b2031004292335re52872d5tf60c9830c8c21...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1

 Hi Ahmed,

 As you can see, the other party gets local ip in SDP

 c=IN IP4 192.168.0.168.

 You can try to play with flags:
 http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html#id229028

 -Laszlo



 2010/4/30 Ahmed Munir ahmedmunir...@gmail.com
 mailto:ahmedmunir...@gmail.com

 
 
  Hi.
 
  Thanks for your reply, the traces are metioned below;
 
  U 203.215.176.22:55134 http://203.215.176.22:55134 -
 11.22.33.44:5060 http://11.22.33.44:5060
  .
  .
  ..
 
  U 81.201.82.45:5060 http://81.201.82.45:5060 -
 11.22.33.44:5060 http://11.22.33.44:5060
  INVITE sip:1234...@11.22.33.44
 mailto:sip%3a1234...@11.22.33.44 sip%3a1234...@11.22.33.44
 mailto:sip%253a1234...@11.22.33.44 SIP/2.0.
  Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45
 mailto:nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
  CSeq: 102 INVITE.
  From: 4572727220 sip:4572727...@voxbone.com
 mailto:sip%3a4572727...@voxbone.comsip%3a4572727...@voxbone.com
 mailto:sip%253a4572727...@voxbone.com
  ;tag=43772.
  To: sip:1234...@11.22.33.44 mailto:sip%3a1234...@11.22.33.44
 sip%3a1234...@11.22.33.44 mailto:sip%253a1234...@11.22.33.44.
  Via: SIP/2.0/UDP 81.201.82.45:5060 http://81.201.82.45:5060
  ;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
  Max-Forwards: 69.
  Content-Type: application/sdp.
  Contact: sip:4572727...@81.201.82.45:5060;transport=udp.
  User-Agent: Vox Callcontrol.
  Content-Length: 210.
  .
  v=0.
  o=root 13293 13293 IN IP4 81.201.82.146.
  s=session.
  c=IN IP4 81.201.82.146.
  t=0 0.
  m=audio 11458 RTP/AVP 8 0.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:0 PCMU/8000.
  a=silenceSupp:off - - - -.
  a=ptime:20.
  a=sendrecv.
 
 
  U 11.22.33.44:5060 http://11.22.33.44:5060 -
 81.201.82.45:5060 http://81.201.82.45:5060
  SIP/2.0 100 Giving a try.
  Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45
 mailto:nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
  CSeq: 102 INVITE.
  From: 4572727220 sip:4572727...@voxbone.com
 mailto:sip%3a4572727...@voxbone.comsip%3a4572727...@voxbone.com
 mailto:sip%253a4572727...@voxbone.com
  ;tag=43772.
  To: sip:1234...@11.22.33.44 mailto:sip%3a1234...@11.22.33.44
 sip%3a1234...@11.22.33.44 mailto:sip%253a1234...@11.22.33.44.
  Via: SIP/2.0/UDP 81.201.82.45:5060 http://81.201.82.45:5060
  ;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0;rport=5060.
  Server: OpenSIPS (1.6.1-notls (i386/linux)).
  Content-Length: 0.
  .
 
 
  U 11.22.33.44:5060 http://11.22.33.44:5060 -
 203.215.176.22:55134 http://203.215.176.22:55134
  INVITE sip:4

Re: [OpenSIPS-Users] NAT Problem using Nat helper

2010-05-03 Thread Ahmed Munir

Hi,

Thanks for supporting me, really appreciated your help.

 Date: Mon, 03 May 2010 12:39:55 +0300
 From: Bogdan-Andrei Iancu bog...@voice-system.ro
 Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
 To: OpenSIPS users mailling list users@lists.opensips.org
 Message-ID: 4bde99eb.9090...@voice-system.ro
 Content-Type: text/plain; charset=ISO-8859-1; format=flowed

 Hi Ahmed,

 as a hint, probably you do not handle correctly the case when only the
 callee is nated (caller is public) - for such cases, to see if rtpproxy
 is needed, after the lookup(location) the nat_bflag will will
 automatically set if the callee location is nated - you can use that
 flag to detect the nated callee and to do the nat fixups - force rtpp
 and fix the 200 ok from the callee (SDP and contact).

 Regards,
 Bogdan

 Ahmed Munir wrote:
  Hi,

  Thanks for replying. Can you please check my configuration of OpenSIPs
  what I posted on my previous digest i.e.Users Digest, Vol 21, Issue 146.

  Please point out in which section do I required to add
  force_rtp_proxy(), because I already configured Nat on it. kindly
  advise me soon.

  On Fri, Apr 30, 2010 at 11:35 AM, users-requ...@lists.opensips.org
  mailto:users-requ...@lists.opensips.org wrote:

  Send Users mailing list submissions to
 users@lists.opensips.org mailto:users@lists.opensips.org

  To subscribe or unsubscribe via the World Wide Web, visit
 http://lists.opensips.org/cgi-bin/mailman/listinfo/users
  or, via email, send a message with subject or body 'help' to
 users-requ...@lists.opensips.org
  mailto:users-requ...@lists.opensips.org

  You can reach the person managing the list at
 users-ow...@lists.opensips.org
  mailto:users-ow...@lists.opensips.org

  When replying, please edit your Subject line so it is more specific
  than Re: Contents of Users digest...

  Today's Topics:

1. Re: NAT Problem using Nat helper (Laszlo)

 --

  Message: 1
  Date: Fri, 30 Apr 2010 08:35:00 +0200
  From: Laszlo las...@voipfreak.net mailto:las...@voipfreak.net
  Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
  To: OpenSIPS users mailling list users@lists.opensips.org
  mailto:users@lists.opensips.org
  Message-ID:

   r2za950b2031004292335re52872d5tf60c9830c8c21...@mail.gmail.com
  mailto:
 r2za950b2031004292335re52872d5tf60c9830c8c21...@mail.gmail.com
  Content-Type: text/plain; charset=iso-8859-1

  Hi Ahmed,

  As you can see, the other party gets local ip in SDP

  c=IN IP4 192.168.0.168.

  You can try to play with flags:

 http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html#id229028

  -Laszlo

 --
 Bogdan-Andrei Iancu
 www.voice-system.ro

 --

 ___
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 End of Users Digest, Vol 22, Issue 13
 *

-- 
Regards,

Ahmed Munir
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Re: [OpenSIPS-Users] NAT Problem using Nat helper

2010-04-30 Thread Ahmed Munir

-Length: 130.
.
v=0.
o=- 2 2 IN IP4 192.168.0.168.
s=CounterPath X-Lite 3.0.
c=IN IP4 192.168.0.168.
t=0 0.
m=audio 1876 RTP/AVP 8 0.
a=sendrecv.


U 81.201.82.45:5060 - 11.22.33.44:5060
ACK sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26;nat=yes
SIP/2.0.
Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
CSeq: 102 ACK.
From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
;tag=43772.
To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
Via: SIP/2.0/UDP 81.201.82.45:5060
;branch=z9hG4bKb10281cafd7849f7565fbfda2b5951fd.
Max-Forwards: 69.
Contact: sip:4572727...@81.201.82.45:5060;transport=udp.
Route: sip:11.22.33.44;lr.
User-Agent: Vox Callcontrol.
Content-Length: 0.
.


U 11.22.33.44:5060 - 203.215.176.22:55134
ACK sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26 SIP/2.0.
Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
CSeq: 102 ACK.
From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
;tag=43772.
To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK1c7c.78b70285.2.
Via: SIP/2.0/UDP 81.201.82.45:5060
;rport=5060;received=81.201.82.45;branch=z9hG4bKb10281cafd7849f7565fbfda2b5951fd.
Max-Forwards: 68.
Contact: sip:4572727...@81.201.82.45:5060;transport=udp.
User-Agent: Vox Callcontrol.
Content-Length: 0.
.


U 11.22.33.44:5060 - 203.215.176.22:55134


U 203.215.176.22:55134 - 11.22.33.44:5060
.
.
..

U 203.215.176.22:55134 - 11.22.33.44:5060
BYE sip:4572727...@81.201.82.45:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.168:55134
;branch=z9hG4bK-d8754z-cf10f40220676576-1---d8754z-;rport.
Max-Forwards: 70.
Route: sip:11.22.33.44;lr.
Contact: sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26.
To: 4572727220sip:4572727...@voxbone.com sip%3a4572727...@voxbone.com
;tag=43772.
From: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
CSeq: 2 BYE.
User-Agent: X-Lite release 1104o stamp 56125.
Reason: SIP;description=User Hung Up.
Content-Length: 0.
.



U 11.22.33.44:5060 - 81.201.82.45:5060
BYE sip:4572727...@81.201.82.45:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK529a.32cc52a.0.
Via: SIP/2.0/UDP 192.168.0.168:55134
;received=203.215.176.22;branch=z9hG4bK-d8754z-cf10f40220676576-1---d8754z-;rport=55134.
Max-Forwards: 69.
Contact: sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26;nat=yes.
To: 4572727220sip:4572727...@voxbone.com sip%3a4572727...@voxbone.com
;tag=43772.
From: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
CSeq: 2 BYE.
User-Agent: X-Lite release 1104o stamp 56125.
Reason: SIP;description=User Hung Up.
Content-Length: 0.
.


U 81.201.82.45:5060 - 11.22.33.44:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK529a.32cc52a.0,SIP/2.0/UDP
192.168.0.168:55134
;received=203.215.176.22;branch=z9hG4bK-d8754z-cf10f40220676576-1---d8754z-;rport=55134.
To: 4572727220 sip:4572727...@voxbone.com sip%3a4572727...@voxbone.com
;tag=43772.
From: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
CSeq: 2 BYE.
Content-Length: 0.
.


U 11.22.33.44:5060 - 203.215.176.22:55134
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.168:55134
;received=203.215.176.22;branch=z9hG4bK-d8754z-cf10f40220676576-1---d8754z-;rport=55134.
To: 4572727220 sip:4572727...@voxbone.com sip%3a4572727...@voxbone.com
;tag=43772.
From: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
CSeq: 2 BYE.
Content-Length: 0.
.


Date: Thu, 29 Apr 2010 19:34:16 -0300
 From: Antonio Anderson Souza anto...@voicetechnology.com.br
 Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
 To: OpenSIPS users mailling list users@lists.opensips.org
 Message-ID:
s2o285c24cc1004291534m1deec8c4zb6c4ddb003311...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1

 Ahmed,

 Could you send an wireshark trace to the list? It will be easier to check
 what's going wrong.

 Besta regards,

 Antonio Anderson M. Souza
 Voice Technology
 http://www.antonioams.com

 Em 29/04/2010 11:47, Ahmed Munir ahmedmunir...@gmail.comescreveu:


 Hi,

 I've configured OpenSIPs with FreeRadius, the version for OpenSIPs I'm
 using
 is 1.6.1 and FreeRadius verison is update 2 date. When I register 2
 sofphone, they got authenticated and authorized by radius and got
 registered sucessfully. Even I made calls between two softphone
 sucessfully(Can hear one another). The UAS configured on different network
 means hosted with public IP and my softphones are registered other and
 NATed
 network. I mapped a DID on UAS and mapped it on my one of my softphone. The
 problem I'm facing is when call coming from DID and ring my phone the
 caller
 can hear me but I can't hear the caller(one way calling issue). But not
 facing the problem

Re: [OpenSIPS-Users] NAT Problem using Nat helper

2010-04-30 Thread Laszlo

.
 To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
 From: 4572727220sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 ;tag=43772.
 Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
 CSeq: 102 INVITE.
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
 SUBSCRIBE, INFO.
 Content-Type: application/sdp.
 User-Agent: X-Lite release 1104o stamp 56125.
 Content-Length: 130.
 .
 v=0.
 o=- 2 2 IN IP4 192.168.0.168.
 s=CounterPath X-Lite 3.0.
 c=IN IP4 192.168.0.168.
 t=0 0.
 m=audio 1876 RTP/AVP 8 0.
 a=sendrecv.


 U 81.201.82.45:5060 - 11.22.33.44:5060
 ACK sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26;nat=yes
 SIP/2.0.
 Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
 CSeq: 102 ACK.
 From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 ;tag=43772.
 To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
 Via: SIP/2.0/UDP 81.201.82.45:5060
 ;branch=z9hG4bKb10281cafd7849f7565fbfda2b5951fd.
 Max-Forwards: 69.
 Contact: sip:4572727...@81.201.82.45:5060;transport=udp.
 Route: sip:11.22.33.44;lr.
 User-Agent: Vox Callcontrol.
 Content-Length: 0.
 .


 U 11.22.33.44:5060 - 203.215.176.22:55134
 ACK sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26 SIP/2.0.
 Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
 CSeq: 102 ACK.
 From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 ;tag=43772.
 To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
 Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK1c7c.78b70285.2.
 Via: SIP/2.0/UDP 81.201.82.45:5060
 ;rport=5060;received=81.201.82.45;branch=z9hG4bKb10281cafd7849f7565fbfda2b5951fd.
 Max-Forwards: 68.
 Contact: sip:4572727...@81.201.82.45:5060;transport=udp.
 User-Agent: Vox Callcontrol.
 Content-Length: 0.
 .


 U 11.22.33.44:5060 - 203.215.176.22:55134
 

 U 203.215.176.22:55134 - 11.22.33.44:5060
 .
 .
 ..

 U 203.215.176.22:55134 - 11.22.33.44:5060
 BYE sip:4572727...@81.201.82.45:5060;transport=udp SIP/2.0.
 Via: SIP/2.0/UDP 192.168.0.168:55134
 ;branch=z9hG4bK-d8754z-cf10f40220676576-1---d8754z-;rport.
 Max-Forwards: 70.
 Route: sip:11.22.33.44;lr.
 Contact: sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26.
 To: 4572727220sip:4572727...@voxbone.com sip%3a4572727...@voxbone.com
 ;tag=43772.
 From: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
 Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
 CSeq: 2 BYE.
 User-Agent: X-Lite release 1104o stamp 56125.
 Reason: SIP;description=User Hung Up.
 Content-Length: 0.
 .



 U 11.22.33.44:5060 - 81.201.82.45:5060
 BYE sip:4572727...@81.201.82.45:5060;transport=udp SIP/2.0.
 Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK529a.32cc52a.0.
 Via: SIP/2.0/UDP 192.168.0.168:55134
 ;received=203.215.176.22;branch=z9hG4bK-d8754z-cf10f40220676576-1---d8754z-;rport=55134.
 Max-Forwards: 69.
 Contact: sip:4...@203.215.176.22:55134
 ;rinstance=25bfe05618433c26;nat=yes.
 To: 4572727220sip:4572727...@voxbone.com sip%3a4572727...@voxbone.com
 ;tag=43772.
 From: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
 Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
 CSeq: 2 BYE.
 User-Agent: X-Lite release 1104o stamp 56125.
 Reason: SIP;description=User Hung Up.
 Content-Length: 0.
 .


 U 81.201.82.45:5060 - 11.22.33.44:5060
 SIP/2.0 200 OK.
 Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK529a.32cc52a.0,SIP/2.0/UDP
 192.168.0.168:55134
 ;received=203.215.176.22;branch=z9hG4bK-d8754z-cf10f40220676576-1---d8754z-;rport=55134.
 To: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 ;tag=43772.
 From: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
 Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
 CSeq: 2 BYE.
 Content-Length: 0.
 .


 U 11.22.33.44:5060 - 203.215.176.22:55134
 SIP/2.0 200 OK.
 Via: SIP/2.0/UDP 192.168.0.168:55134
 ;received=203.215.176.22;branch=z9hG4bK-d8754z-cf10f40220676576-1---d8754z-;rport=55134.
 To: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 ;tag=43772.
 From: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
 Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
 CSeq: 2 BYE.
 Content-Length: 0.
 .


 Date: Thu, 29 Apr 2010 19:34:16 -0300
 From: Antonio Anderson Souza anto...@voicetechnology.com.br
 Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
 To: OpenSIPS users mailling list users@lists.opensips.org
 Message-ID:
s2o285c24cc1004291534m1deec8c4zb6c4ddb003311...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1


 Ahmed,

 Could you send an wireshark trace to the list? It will be easier to check
 what's going wrong.

 Besta regards,

 Antonio Anderson M. Souza
 Voice Technology
 http://www.antonioams.com

 Em 29/04/2010 11:47, Ahmed Munir ahmedmunir...@gmail.comescreveu:


 Hi,

 I've configured OpenSIPs with FreeRadius, the version for OpenSIPs I'm
 using
 is 1.6.1 and FreeRadius verison is update 2 date. When I register 2
 sofphone, they got authenticated

Re: [OpenSIPS-Users] NAT Problem using Nat helper

2010-04-29 Thread Antonio Anderson Souza

Ahmed,

Could you send an wireshark trace to the list? It will be easier to check
what's going wrong.

Besta regards,

Antonio Anderson M. Souza
Voice Technology
http://www.antonioams.com

Em 29/04/2010 11:47, Ahmed Munir ahmedmunir...@gmail.comescreveu:

Hi,

I've configured OpenSIPs with FreeRadius, the version for OpenSIPs I'm using
is 1.6.1 and FreeRadius verison is update 2 date. When I register 2
sofphone, they got authenticated and authorized by radius and got
registered sucessfully. Even I made calls between two softphone
sucessfully(Can hear one another). The UAS configured on different network
means hosted with public IP and my softphones are registered other and NATed
network. I mapped a DID on UAS and mapped it on my one of my softphone. The
problem I'm facing is when call coming from DID and ring my phone the caller
can hear me but I can't hear the caller(one way calling issue). But not
facing the problem on when calling between to sip clients and also calling
from OpenSIPs to Asterisk. The configuration and cases for OpenSIPs is
listed down below;


UAC-- UAS(OpenSIPs) -
UACtwo way voice is establised
 UAC-- UAS(OpenSIPs) - Asterisk
 UACtwo way voice is establised
PSTN-- UAS(OpenSIPs)
- UAC  one way
voice is establised
(hears the dest voice)(can't hear caller
voice)


#loadmodule auth_diameter.so
loadmodule aaa_radius.so
loadmodule auth_aaa.so
loadmodule permissions.so
loadmodule nathelper.so
#Settings For
Radius-
#modparam(auth_diameter, diameter_client_host, localhost)
modparam(aaa_radius,
radius_config,/usr/etc/radiusclient-ng/radiusclient.conf)
modparam(acc, aaa_url,
radius:/usr/etc/radiusclient-ng/radiusclient.conf)
modparam(acc, aaa_flag, 2)
modparam(acc, aaa_missed_flag, 3)
modparam(acc, aaa_extra, User-Name=$Au; \
Calling-Station-Id=$from; \
Called-Station-Id=$to; \
Sip-Translated-Request-URI=$ruri; \
Sip-RPid=$avp(s:rpid); \
Source-IP=$si; \
Source-Port=$sp; \
Canonical-URI=$avp(s:can_uri); \
Billing-Party=$avp(s:billing_party); \
Divert-Reason=$avp(s:divert_reason); \
X-RTP-Stat=$hdr(X-RTP-Stat); \
Contact=$hdr(contact); \
Event=$hdr(event); \
SIP-Proxy-IP=$avp(s:sip_proxy_ip); \
ENUM-TLD=$avp(s:enum_tld))
modparam(auth_aaa,aaa_url,radius:/usr/etc/radiusclient-ng/radiusclient.conf)
modparam(auth, rpid_prefix, sip:)
modparam(auth, rpid_suffix, @11.22.33.44;screen=yes;privacy=off)
modparam(auth, rpid_avp, $avp(s:rpid))
#modparam(uri,service_type,10)
# - setting module-specific parameters ---
modparam(dispatcher, db_url, mysql://opensips:opensip...@localhost
/opensips)
modparam(permissions, db_url, mysql://opensips:opensip...@localhost
/opensips)
#- setting NAT module parameters -
modparam(nathelper,ping_nated_only,1)
modparam(nathelper, natping_interval, 30)
modparam(nathelper,natping_processes,1)
#modparam(nathelper,rtpproxy_sock,udp:127.0.0.1:7890)
modparam(nathelper,rtpproxy_sock, )
modparam(nathelper,received_avp,$avp(i:42))
#modparam(nathelper, sipping_bflag, 7)
modparam(usrloc, nat_bflag, 6)
### Routing Logic 
# main request routing logic
route{
 if (!mf_process_maxfwd_header(10)) {
  sl_send_reply(483,Too Many Hops);
  exit;
 }

 #NAT detection
 log(# Go to Route 3 for NAT
Detection #);
 route(3);
 if (has_totag()) {
  # sequential request withing a dialog should
  # take the path determined by record-routing
  if (loose_route()) {
   if (is_method(BYE)) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
   } else if (is_method(INVITE)) {
# even if in most of the cases is useless, do RR for
# re-INVITEs alos, as some buggy clients do change route set
# during the dialog.
record_route();
   }
   # route it out to whatever destination was set by loose_route()
   # in $du (destination URI).
   route(1);
  } else {
   if ( is_method(ACK) ) {
if ( t_check_trans() ) {
 # non loose-route, but stateful ACK; must be an ACK after
 # a 487 or e.g. 404 from upstream server
 t_relay();
 exit;
} else {
 # ACK without matching transaction -
 # ignore and discard
 exit;
}
   }
   sl_send_reply(404,Not here);
  }
  exit;
 }
 #initial requests
 # CANCEL processing
 if (is_method(CANCEL))
 {
  if (t_check_trans())
   t_relay();
  exit;
 }
 t_check_trans();
 if (loose_route()) {
  xlog(L_ERR,
  Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]);
  if (!is_method(ACK))
   sl_send_reply(403,Preload Route denied);
  exit;
 }
 # record routing
 if (!is_method(REGISTER|MESSAGE))
  record_route();

Re: [OpenSIPS-Users] NAT problem, no-audio when calling outside network... Please help

2009-08-31 Thread Bogdan-Andrei Iancu

Hi Khan,

You can start with 2 simple checks:

1) be sure your force_rtp_proxy() functions are triggred both for 
request and reply - put some xlog to see if you get there in the script

2) check the messages with SDP (on the outgoing part) if they have the 
rtpproxy indication in SDP

Regards,
Bogdan


Khan wrote:
 Hey everyone,

 I have been trying to work this for a long time, this mailing list is
 my last resort. I have applied NAT traversal using RTP proxy. My
 scenario is as follows:
 UAC1 (behind NAT) --- UAC2 (behind NAT)

 The UAC's get authenticated fine, call establishes but there is no
 voice, neither i hear them nor they hear me. I can't pin point exactly
 where did i go wrong. My script is as follows:

 route{
 ## unrelated script has been stripped!!!
   if (nat_uac_test(3)) {
   if (is_method(REGISTER) || !is_present_hf(Record-Route)) {
   log(LOG:Someone trying to register from private IP, 
 rewriting\n);
   # Rewrite contact with source IP of signalling
   fix_nated_contact();
   if ( is_method(INVITE) ) {
   fix_nated_sdp(1); # Add direction=active to 
 SDP
   };

   force_rport(); # Add rport parameter to topmost Via
   setbflag(6);# Mark as NATed

   # if you want sip nat pinging
   setbflag(8);

   xlog(L_INFO, fixNATed and setbflag 6, 8 - M=$rm RURI=$ru 
 F=$fu
 T=$tu IP=$si ID=$ci\n);
   };
   };

   # sequential requests...
   if (has_totag()) {
   # sequential request withing a dialog should
   # take the path determined by record-routing
   if (loose_route()) {
   xlog(L_INFO, Initial loose-routing - M=$rm RURI=$ru 
 F=$fu T=$tu
 IP=$si \n);

   # mark routing logic in request
   append_hf(P-hint: rr-enforced\r\n);
   if (is_method(BYE)) {
   setflag(1); # do accounting ...
   setflag(3); # ... even if the transaction fails
   xlog(L_INFO, BYE ... unforce RTP - M=$rm RURI=$ru 
 F=$fu T=$tu
 IP=$si ID=$ci\n);
   unforce_rtp_proxy();
   } else if (is_method(INVITE)) {
   # even if in most of the cases is useless, do 
 RR for
   # re-INVITEs alos, as some buggy clients do 
 change route set
   # during the dialog.
   record_route();
   }
   # route it out to whatever destination was set by 
 loose_route()
   # in $du (destination URI).
   route(1);
   } else {
   if ( is_method(ACK) ) {
   if ( t_check_trans() ) {
   # non loose-route, but stateful ACK; 
 must be an ACK after
   # a 487 or e.g. 404 from upstream server
   t_relay();
   exit;
   } else {
   # ACK without matching transaction -
   # ignore and discard
   exit;
   }
   }
   sl_send_reply(404,Not here);
   }
   exit;
   }

   #initial requests
   # CANCEL processing
   if (is_method(CANCEL))
   {
   if (t_check_trans())
   t_relay();
   xlog(L_INFO, CANCEL ... unforce RTP - M=$rm RURI=$ru F=$fu 
 T=$tu
 IP=$si ID=$ci\n);
   unforce_rtp_proxy();
   exit;
   }

   #-- Preventing the UAC problem which sends Option
 ##if(is_method(OPTIONS)){
 ##sl_send_reply(200, OK);
 ##exit;
 ##}

 #-- uncommented followings
 if ((method==OPTIONS|SUBSCRIBE)  from_uri==myself) /*no
 multidomain version*/
 ##if (!(method==OPTIONS)  is_from_local())  /*multidomain 
 version*/
 {
 if (!proxy_authorize(, subscriber)) {
 proxy_challenge(, 0);
 exit;
 }
 if (!check_from()) {
 sl_send_reply(403,Forbidden auth ID);
 exit;
 }

 consume_credentials();
 # caller authenticated
 }

   t_check_trans();
   if (!(method==REGISTER)  from_uri==myself) /*no multidomain 
 version*/
   ##if (!(method==REGISTER)  is_from_local())  /*multidomain version*/
   {
   if

Re: [OpenSIPS-Users] NAT problem

2008-11-28 Thread Juan Backson

Hi Bogdan

Thank you for your help.

The nated client does register to opensips.  It is set to register every
3600 sec, min time  is 20 s and max time is 1800 s.  It is default xLite
setting.

Here is the 200OK I captured from my nated client box:

!'DVVEGTeEd=3SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101
;branch=z9hG4bKbf91.9b9bad57.0;received=233.32.345.5
Via: SIP/2.0/UDP
233.32.345.5:5800;received=233.32.345.5;rport=5800;branch=z9hG4bKNj4y6pUrS49FF
Record-Route: sip:192.168.1.101;lr;ftag=UD1K6e2FpUgNj
Contact: sip:[EMAIL PROTECTED]:33756
To: 1000sip:[EMAIL PROTECTED]:5060;tag=194ddb10
From: 0sip:[EMAIL PROTECTED]:5060;tag=UD1K6e2FpUgNj
Call-ID: MGUzMzZjNGNhNGM3MzY4ZDVjMjg3M2I2OGI2OTc0OWE.
CSeq: 107790129 BYE
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0

Here is the INVITE request:

!'DVVEMKd=*INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:33756
;branch=z9hG4bK-d87543-8e2c20026843651b-1--d87543-;rport
Max-Forwards: 70
Contact: sip:[EMAIL PROTECTED]:33756
To: 0sip:[EMAIL PROTECTED]:5060
From: 1000sip:[EMAIL PROTECTED]:5060;tag=194ddb10
Call-ID: MGUzMzZjNGNhNGM3MzY4ZDVjMjg3M2I2OGI2OTc0OWE.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 423
v=0
o=- 9 2 IN IP4 192.168.1.100
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.100
t=0 0
m=audio 26258 RTP/AVP 107 119 100 106 0 105 98 8 101
a=alt:1 1 : LGfU4oal SL5N8UZJ 192.168.1.100 26258
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv





On Thu, Nov 27, 2008 at 1:53 AM, Bogdan-Andrei Iancu [EMAIL PROTECTED]
 wrote:

 Hi Juan,

 I need to see the request part also to figure out if the flow through the
 NAT is ok or not.

 As a side note - could you check if the device behind the nat is actually
 receiving the 200 OK?. Because a typical reason for a missing ACK is  a
 missing 200 OK.

 Another question - the device placing the call (from behind the nat) is
 registered or not? what is the estimated setup time in this case (time
 between invite and 200 OK) ?

 Regards,
 Bogdan

 Juan Backson wrote:

 Hi,

 I am having problem with configuring opensips to work with NATed clients.
  In my configuration, I am using a B2BUA and Opensips as the sip proxy.
 The problem I am having is that when the B2BUA(233.32.345.5:5800) sends
 out 200 OK, Opensips (192.168.1.101:5060)is able to proxy it to the NATed
 client ( 116.24.163.21:2751 http://116.24.163.21:2751), but the NATed
 client is not sending back any ACK, so the B2BUA hangs up after 30 second.
 Could someone give me any suggestion on what may be wrong in my config?

 Thanks in advance for all the help.


 U 233.32.345.5:5800 - 192.168.1.101:5060 http://192.168.1.101:5060
 SIP/2.0 200 OK.
 Via: SIP/2.0/UDP 192.168.1.101 http://192.168.1.101
 ;branch=z9hG4bK3ab5.9b17c4a1.0;received=233.32.345.5.
 Via: SIP/2.0/UDP 192.168.1.100:26682;received=116.24.163.21 
 http://116.24.163.21
 ;branch=z9hG4bK-d87543-1a09c008b901bc5c-1--d87543-;rport=2751.
 Record-Route: sip:192.168.1.101 
 http://192.168.1.101;lr=on;ftag=b81a6b5e;nat=yes.


 From: 1000 sip:[EMAIL PROTECTED]:5060;tag=b81a6b5e.
 To: 0 sip:[EMAIL PROTECTED]:5060;tag=Sy7K9eUFg61tB.
 Call-ID: ODRiMGUzMGFiZDg2OGU0OGNiYmE0MWY5OWRkMTMxOTA..
 CSeq: 2 INVITE.
 Contact: sip:[EMAIL PROTECTED]:5800;transport=udp.
 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M.
 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
 NOTIFY, REFER, UPDATE, REGISTER, INFO.
 Supported: timer, precondition, path, replaces.
 Allow-Events: talk.
 Session-Expires: 120;refresher=uas.
 Min-SE: 120.
 Content-Type: application/sdp.
 Content-Disposition: session.
 Content-Length: 269.
 .
 v=0.
 o=FreeSWITCH 5494423604621376967 2638962022927722250 IN IP4 233.32.345.5.
 s=FreeSWITCH.
 c=IN IP4 233.32.345.5.
 t=0 0.
 m=audio 10272 RTP/AVP 0 101.
 a=rtpmap:0 PCMU/8000.
 a=rtpmap:101 telephone-event/8000.
 a=fmtp:101 0-16.
 a=silenceSupp:off - - - -.
 a=ptime:20.


 U 192.168.1.101:5060 http://192.168.1.101:5060 - 116.24.163.21:2751 
 http://116.24.163.21:2751
 SIP/2.0 200 OK.
 Via: SIP/2.0/UDP 192.168.1.100:26682;received=116.24.163.21 
 http://116.24.163.21
 ;branch=z9hG4bK-d87543-1a09c008b901bc5c-1--d87543-;rport=2751.
 Record-Route: sip:192.168.1.101 
 http://192.168.1.101;lr=on;ftag=b81a6b5e;nat=yes.


 From: 1000 sip:[EMAIL PROTECTED]:5060;tag=b81a6b5e.
 To: 0 sip:[EMAIL PROTECTED]:5060;tag=Sy7K9eUFg61tB.
 Call-ID: ODRiMGUzMGFiZDg2OGU0OGNiYmE0MWY5OWRkMTMxOTA..
 CSeq: 2 INVITE.
 Contact: sip:[EMAIL PROTECTED]:5800;transport=udp.
 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M.
 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
 NOTIFY, REFER, UPDATE, REGISTER, INFO.
 Supported: timer, precondition, path, replaces.

Re: [OpenSIPS-Users] NAT problem

2008-11-26 Thread Bogdan-Andrei Iancu

Hi Juan,

I need to see the request part also to figure out if the flow through 
the NAT is ok or not.

As a side note - could you check if the device behind the nat is 
actually receiving the 200 OK?. Because a typical reason for a missing 
ACK is  a missing 200 OK.

Another question - the device placing the call (from behind the nat) is 
registered or not? what is the estimated setup time in this case (time 
between invite and 200 OK) ?

Regards,
Bogdan

Juan Backson wrote:
 Hi,

 I am having problem with configuring opensips to work with NATed 
 clients.  In my configuration, I am using a B2BUA and Opensips as the 
 sip proxy. 

 The problem I am having is that when the B2BUA(233.32.345.5:5800) 
 sends out 200 OK, Opensips (192.168.1.101:5060)is able to proxy it to 
 the NATed client ( 116.24.163.21:2751 http://116.24.163.21:2751), 
 but the NATed client is not sending back any ACK, so the B2BUA hangs 
 up after 30 second. 

 Could someone give me any suggestion on what may be wrong in my config?

 Thanks in advance for all the help.


 U 233.32.345.5:5800 - 192.168.1.101:5060 http://192.168.1.101:5060
 SIP/2.0 200 OK.
 Via: SIP/2.0/UDP 192.168.1.101 
 http://192.168.1.101;branch=z9hG4bK3ab5.9b17c4a1.0;received=233.32.345.5.
 Via: SIP/2.0/UDP 192.168.1.100:26682;received=116.24.163.21 
 http://116.24.163.21;branch=z9hG4bK-d87543-1a09c008b901bc5c-1--d87543-;rport=2751.
 Record-Route: sip:192.168.1.101 
 http://192.168.1.101;lr=on;ftag=b81a6b5e;nat=yes.
 From: 1000 sip:[EMAIL PROTECTED]:5060;tag=b81a6b5e.
 To: 0 sip:[EMAIL PROTECTED]:5060;tag=Sy7K9eUFg61tB.
 Call-ID: ODRiMGUzMGFiZDg2OGU0OGNiYmE0MWY5OWRkMTMxOTA..
 CSeq: 2 INVITE.
 Contact: sip:[EMAIL PROTECTED]:5800;transport=udp.
 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M.
 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, 
 NOTIFY, REFER, UPDATE, REGISTER, INFO.
 Supported: timer, precondition, path, replaces.
 Allow-Events: talk.
 Session-Expires: 120;refresher=uas.
 Min-SE: 120.
 Content-Type: application/sdp.
 Content-Disposition: session.
 Content-Length: 269.
 .
 v=0.
 o=FreeSWITCH 5494423604621376967 2638962022927722250 IN IP4 233.32.345.5.
 s=FreeSWITCH.
 c=IN IP4 233.32.345.5.
 t=0 0.
 m=audio 10272 RTP/AVP 0 101.
 a=rtpmap:0 PCMU/8000.
 a=rtpmap:101 telephone-event/8000.
 a=fmtp:101 0-16.
 a=silenceSupp:off - - - -.
 a=ptime:20.


 U 192.168.1.101:5060 http://192.168.1.101:5060 - 116.24.163.21:2751 
 http://116.24.163.21:2751
 SIP/2.0 200 OK.
 Via: SIP/2.0/UDP 192.168.1.100:26682;received=116.24.163.21 
 http://116.24.163.21;branch=z9hG4bK-d87543-1a09c008b901bc5c-1--d87543-;rport=2751.
 Record-Route: sip:192.168.1.101 
 http://192.168.1.101;lr=on;ftag=b81a6b5e;nat=yes.
 From: 1000 sip:[EMAIL PROTECTED]:5060;tag=b81a6b5e.
 To: 0 sip:[EMAIL PROTECTED]:5060;tag=Sy7K9eUFg61tB.
 Call-ID: ODRiMGUzMGFiZDg2OGU0OGNiYmE0MWY5OWRkMTMxOTA..
 CSeq: 2 INVITE.
 Contact: sip:[EMAIL PROTECTED]:5800;transport=udp.
 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M.
 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, 
 NOTIFY, REFER, UPDATE, REGISTER, INFO.
 Supported: timer, precondition, path, replaces.
 Allow-Events: talk.
 Session-Expires: 120;refresher=uas.
 Min-SE: 120.
 Content-Type: application/sdp.
 Content-Disposition: session.
 Content-Length: 269.
 .
 v=0.
 o=FreeSWITCH 5494423604621376967 2638962022927722250 IN IP4 233.32.345.5.
 s=FreeSWITCH.
 c=IN IP4 233.32.345.5.
 t=0 0.
 m=audio 10272 RTP/AVP 0 101.
 a=rtpmap:0 PCMU/8000.
 a=rtpmap:101 telephone-event/8000.
 a=fmtp:101 0-16.
 a=silenceSupp:off - - - -.
 a=ptime:20.


 U 192.168.1.101:5800 http://192.168.1.101:5800 - 233.32.345.5:5060
 BYE sip:[EMAIL PROTECTED]:2751 http://sip:[EMAIL PROTECTED]:2751 
 SIP/2.0.
 Via: SIP/2.0/UDP 233.32.345.5:5800;rport;branch=z9hG4bK01H0jSevQ2Nmc.
 Route: sip:192.168.1.101 
 http://192.168.1.101;lr=on;ftag=b81a6b5e;nat=yes.
 Max-Forwards: 70.
 From: 0 sip:[EMAIL PROTECTED]:5060;tag=Sy7K9eUFg61tB.
 To: 1000 sip:[EMAIL PROTECTED]:5060;tag=b81a6b5e.
 Call-ID: ODRiMGUzMGFiZDg2OGU0OGNiYmE0MWY5OWRkMTMxOTA..
 CSeq: 107702524 BYE.
 Contact: sip:[EMAIL PROTECTED]:5800;transport=udp.
 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M.
 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, 
 NOTIFY, REFER, UPDATE, REGISTER, INFO.
 Supported: timer, precondition, path, replaces.
 Reason: SIP;cause=408;text=ACK Timeout.
 Content-Length: 0.
 .




 #
 # $Id: openser.cfg 1676 2007-02-21 13:16:34Z bogdan_iancu $
 #
 #simple quick-start config script
 #Please refer to the Core CookBook at 
 http://www.openser.org/dokuwiki/doku.php
 #for a explanation of possible statements, functions and parameters.
 #
 # --- global configuration parameters 
 debug=3# debug level (cmd line: -dd)
 fork=no
 log_stderror=yes# (cmd line: -E)
 children=4
 port=5060
 mpath=/usr/local/lib64/opensips/modules/
 loadmodule db_mysql.so
 loadmodule sl.so
 loadmodule tm.so
 loadmodule rr.so

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