Re: [OpenSIPS-Users] NAT problem

[xiaofeng]

Hi,

Thanks for the reply.

Manually insert header field "Record-Route" by
insert_hf("Record-Route: \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

[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

[Terrance Devor]

ngrep -d eth0 -qt -W byline portrange 5060
​
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Re: [OpenSIPS-Users] NAT problem

[Terrance Devor]

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

[xiaofeng]

Hi everyone,

I have a public IP(106.x.x.x), a OpenSIPS(192.168.1.x) and two SIP
UAC(192.168.2.x/192.168.3.x),  the OpenSIPS and UACs are all behind NAT but
not the same NAT.

I configure my route to use the public IP, and forward all packets to my
OpenSIPS.

1. The UACs register to 106.x.x.x, and registered sucessfully.
2. UACs can send/receive 'INVITE' and '200 OK' properly.

What's the problem is:
1. UAC cannot sent/receive 'ACK', as the UAC tries to send 'ACK' request to
the other peer directly. When use nathelper, the UAC tries to send 'ACK' to
192.168.1.x, which still has problems.
2. The RTP cannot send properly, try rtpproxy with no luck. (The rtpproxy
and OpenSIPS run ont the same machine.)

The opensips.cfg is some like the nathelper.cfg which is in the
OpenSIPS/examples/ directory.

Any one can help? Thanks in advance


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*NAT*



 +---+  
   +---+
 |   internet
+-+   public IP   | 
 +---+  
   +---+
/ \ 
   | directly forward to OpenSIPS
  / \   
   ++
/3G/4G/LTE/WiFi   \ 
   |OpenSIPS|
  / \   
   ++
/ \
  / \
 +---+   +---+
 | mobile #1 |  ...  | mobile #N |
 +---+   +---+

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Re: [OpenSIPS-Users] NAT problem on 5060

[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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Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

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[OpenSIPS-Users] NAT problem on 5060

[Amirehsan Sadeghi]

 Dear,
 By the both of  the last versions of opensips 1.8 and 1.10, there  is a
same problem,
 the clients behind of NAT don't have media if the port of opensips is 5060.
 by changing the port to 5090 or any other ports everything is ok.

 Could you please let me where my fault is?


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Re: [OpenSIPS-Users] Nat Problem

[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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[OpenSIPS-Users] Nat Problem

[Ahmed Munir]

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;


loadmodule "dispatcher.so"
loadmodule "avpops.so"
loadmodule "permissions.so"
loadmodule "aaa_radius.so"
loadmodule "auth_aaa.so"
#loadmodule "auth_diameter.so"
loadmodule "nathelper.so"

#Settings For
Radius-
#modparam("auth_diameter", "diameter_client_host", "localhost")
modparam("aaa_radius",
"radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf")
modparam("acc", "aaa_url",
"radius:/usr/local/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/local/etc/radiusclient-ng/radiusclient.conf")
modparam("auth", "rpid_prefix", ";screen=yes;privacy=off")
#modparam("auth", "rpid_suffix", "@203.215.179.54>;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)


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()) {
if (loose_route()) {
if (is_method("BYE")) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction
fails
} else if (is_method("INVITE")) {
record_route();
}
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();


# preloaded route checking
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();

$avp(s:checksrc) = check_source_address(

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

[Ahmed Munir]

Hi,

Thanks for supporting me, really appreciated your help.


> Date: Mon, 03 May 2010 12:39:55 +0300
> From: Bogdan-Andrei Iancu 
> Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
> To: OpenSIPS users mailling list 
> 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,  > <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 mailto:las...@voipfreak.net>>
> > Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
> > To: OpenSIPS users mailling list  > <mailto: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
> >
> >
> >
> >
>
> --
> Bogdan-Andrei Iancu
> www.voice-system.ro
>
>
>
>
> --
>
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>
> End of Users Digest, Vol 22, Issue 13
> *
>



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

[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,  <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 mailto:las...@voipfreak.net>>
> Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
> To: OpenSIPS users mailling list  <mailto:users@lists.opensips.org>>
> Message-ID:
>  
>   <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  <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>  <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"  <mailto:sip%3a4572727...@voxbone.com> <mailto:sip%253a4572727...@voxbone.com>>
> > >;tag=43772.
> > To: mailto: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: .
> > 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"  <mailto:sip%3a4572727...@voxbone.com> <mailto:sip%253a4572727...@v

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

[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,  wrote:

> Send Users mailing list submissions to
>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
>
> You can reach the person managing the list at
>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 
> Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
> To: OpenSIPS users mailling list 
> Message-ID:
>
> 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 
>
> >
> >
> > 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/2.0.
> > Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
> > CSeq: 102 INVITE.
> > From: "4572727220" 
> >
> > >;tag=43772.
> > To:  <
> sip%3a1234...@11.22.33.44 >>.
> > Via: SIP/2.0/UDP 81.201.82.45:5060
> > ;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
> > Max-Forwards: 69.
> > Content-Type: application/sdp.
> > Contact: .
> > 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" 
> >
> > >;tag=43772.
> > To:  <
> sip%3a1234...@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: .
> > Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
> > CSeq: 102 INVITE.
> > From: "4572727220" 
> >
> > >;tag=43772.
> > To:  <
> sip%3a1234...@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: .
> > 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: .
> > Contact: .
> > To:  <
> sip%3a1234...@11.22.33.44 >>;tag=611cee1e

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

[Laszlo]

gt; 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" 
> >;tag=43772.
> To: >;tag=611cee1e.
> Via: SIP/2.0/UDP 81.201.82.45:5060
> ;branch=z9hG4bKb10281cafd7849f7565fbfda2b5951fd.
> Max-Forwards: 69.
> Contact: .
> Route: .
> 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" 
> >;tag=43772.
> To: >;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: .
> 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: .
> Contact: .
> To: "4572727220"
> >;tag=43772.
> From: >;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:  ;rinstance=25bfe05618433c26;nat=yes>.
> To: "4572727220"
> >;tag=43772.
> From: >;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" 
> >;tag=43772.
> From: >;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" 
> >;tag=43772.
> From: >;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 
>> Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
>> To: OpenSIPS users mailling list 
>> Message-ID:
>>
>> 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" escreveu:
>>
>>
>> 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

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

[Ahmed Munir]

.
Max-Forwards: 68.
Contact: .
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: .
Contact: .
To: "4572727220"
>;tag=43772.
From: >;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: .
To: "4572727220"
>;tag=43772.
From: >;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" 
>;tag=43772.
From: >;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" 
>;tag=43772.
From: >;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 
> Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
> To: OpenSIPS users mailling list 
> Message-ID:
>
> 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" escreveu:
>
>
> 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)
>
>
>
>
>
> Kindly help me out with this problem, in which other section Natting is
> required?(or am I missing something in the configuration?)  Please assist
> me
> on it.
> --
> Regards,
>
> Ahmed Munir
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> -- next part --
> An HTML attachment was scrubbed...
> URL:
> http://lists.opensips.org/pipermail/users/attachments/20100429/84192485/attachment.htm
>
> --
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> End of Users Digest, Vol 21, Issue 146
> **
>



-- 
Regards,

Ahmed Munir





-- 
Regards,

Ahmed Munir
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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

[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" escreveu:

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=$avp(s:enum_tld)")
modparam("auth_aaa","aaa_url","radius:/usr/etc/radiusclient-ng/radiusclient.conf")
modparam("auth", "rpid_prefix", ";screen=yes;privacy=off")
modparam("auth", "rpid_avp", "$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;

[OpenSIPS-Users] NAT Problem using Nat helper

[Ahmed Munir]

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=$avp(s:enum_tld)")
modparam("auth_aaa","aaa_url","radius:/usr/etc/radiusclient-ng/radiusclient.conf")
modparam("auth", "rpid_prefix", ";screen=yes;privacy=off")
modparam("auth", "rpid_avp", "$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();

 #$avp(i:27)=check_source_address("0");
 #xlog("Check Source Address from Address TABLE : $(avp(i:27))\n");
 $avp(s:checksrc) = check_source_address("0");
 

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

[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

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

[Khan]

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 (!proxy_authorize("", "subscriber")) {
proxy_challenge("", "0");
exit;
}
if (!check_from()) {
sl_send_reply("403","Forbi

Re: [OpenSIPS-Users] NAT problem

[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: 
Contact: 
To: "1000";tag=194ddb10
From: "0";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: 
To: "0"
From: "1000";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 ), 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 
>> SIP/2.0 200 OK.
>> Via: SIP/2.0/UDP 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: > ;lr=on;ftag=b81a6b5e;nat=yes>.
>>
>>
>> From: "1000" ;tag=b81a6b5e.
>> To: "0" ;tag=Sy7K9eUFg61tB.
>> Call-ID: ODRiMGUzMGFiZDg2OGU0OGNiYmE0MWY5OWRkMTMxOTA..
>> CSeq: 2 INVITE.
>> Contact: .
>> 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  -> 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: > ;lr=on;ftag=b81a6b5e;nat=yes>.
>>
>>
>> From: "1000" ;tag=b81a6b5e.
>> To: "0" ;tag=Sy7K9eUFg61tB.
>> Call-ID: ODRiMGUzMGFiZDg2OGU0OGNiYmE0MWY5OWRkMTMxOTA..
>> CSeq: 2 INVITE.
>> Contact: .
>> 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 26389620229277

Re: [OpenSIPS-Users] NAT problem

[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 ), 
> 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 
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 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 
> ;branch=z9hG4bK-d87543-1a09c008b901bc5c-1--d87543-;rport=2751.
> Record-Route:  ;lr=on;ftag=b81a6b5e;nat=yes>.
> From: "1000" ;tag=b81a6b5e.
> To: "0" ;tag=Sy7K9eUFg61tB.
> Call-ID: ODRiMGUzMGFiZDg2OGU0OGNiYmE0MWY5OWRkMTMxOTA..
> CSeq: 2 INVITE.
> Contact: .
> 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  -> 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 
> ;branch=z9hG4bK-d87543-1a09c008b901bc5c-1--d87543-;rport=2751.
> Record-Route:  ;lr=on;ftag=b81a6b5e;nat=yes>.
> From: "1000" ;tag=b81a6b5e.
> To: "0" ;tag=Sy7K9eUFg61tB.
> Call-ID: ODRiMGUzMGFiZDg2OGU0OGNiYmE0MWY5OWRkMTMxOTA..
> CSeq: 2 INVITE.
> Contact: .
> 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  -> 233.32.345.5:5060
> BYE sip:[EMAIL PROTECTED]:2751  
> SIP/2.0.
> Via: SIP/2.0/UDP 233.32.345.5:5800;rport;branch=z9hG4bK01H0jSevQ2Nmc.
> Route:  ;lr=on;ftag=b81a6b5e;nat=yes>.
> Max-Forwards: 70.
> From: "0" ;tag=Sy7K9eUFg61tB.
> To: "1000" ;tag=b81a6b5e.
> Call-ID: ODRiMGUzMGFiZDg2OGU0OGNiYmE0MWY5OWRkMTMxOTA..
> CSeq: 107702524 BYE.
> Contact: .
> 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"
> loadmodule "maxfwd.so"
> loadmodule "usrloc.so"
> loadmodule "registrar.so"
> loadmodule "textops.so"
> loadmodule "mi_fifo.so"
> loadmodule "uri.so"
> loa

[OpenSIPS-Users] NAT problem

[Juan Backson]

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), 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
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 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
;branch=z9hG4bK-d87543-1a09c008b901bc5c-1--d87543-;rport=2751.
Record-Route: .
From: "1000" ;tag=b81a6b5e.
To: "0" ;tag=Sy7K9eUFg61tB.
Call-ID: ODRiMGUzMGFiZDg2OGU0OGNiYmE0MWY5OWRkMTMxOTA..
CSeq: 2 INVITE.
Contact: .
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 -> 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
;branch=z9hG4bK-d87543-1a09c008b901bc5c-1--d87543-;rport=2751.
Record-Route: .
From: "1000" ;tag=b81a6b5e.
To: "0" ;tag=Sy7K9eUFg61tB.
Call-ID: ODRiMGUzMGFiZDg2OGU0OGNiYmE0MWY5OWRkMTMxOTA..
CSeq: 2 INVITE.
Contact: .
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 -> 233.32.345.5:5060
BYE sip:[EMAIL PROTECTED]:2751 SIP/2.0.
Via: SIP/2.0/UDP 233.32.345.5:5800;rport;branch=z9hG4bK01H0jSevQ2Nmc.
Route: .
Max-Forwards: 70.
From: "0" ;tag=Sy7K9eUFg61tB.
To: "1000" ;tag=b81a6b5e.
Call-ID: ODRiMGUzMGFiZDg2OGU0OGNiYmE0MWY5OWRkMTMxOTA..
CSeq: 107702524 BYE.
Contact: .
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"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "mi_fifo.so"
loadmodule "uri.so"
loadmodule "uri_db.so"
loadmodule "domain.so"
loadmodule "xlog.so"
loadmodule "permissions.so"
loadmodule "auth.so"
loadmodule "auth_db.so"
loadmodule "dispatcher.so"
loadmodule "nathelper.so"
loadmodule "mediaproxy.so"









modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("usrloc", "db_mode", 2)



modparam("rr", "enable_full_lr", 1)

modparam("auth_db|usrloc|domain|uri_db|permissions|dispatcher","db_url","mysql://
root:[EMAIL PROTECTED]/app")
modparam("auth_db","calculate_ha1",yes)
modparam("auth_db","password_column","password")
modparam("auth_db","user_column","sip_user")
modparam("auth_db","load_credentials","agent_id")

modparam("uri_db","db_table","agent")
modparam("uri_db","user_column","sip_user")
modparam("uri_db","use_uri_table",0)
modparam("auth_db","use_domain",0)

modparam("permissions", "db_mode", 1)
modparam("permissions", "trusted_table", "server")
modparam("permissions","source_col","server_ip")
modparam("permissions","proto_col","transport")
modparam("permissions","from_col","from_pattern")
modparam("permissions","tag_col","peer_tag")

modparam("dispatcher","table_name","dispatcher")
modparam("dispatche