opensipstack-devel

Re: [OpenSIPStack] ATLSIP 1.0.2 Call Transfer or Conference

[Andre S. <eds...@ya...>]

Call Transfer and Call Conference is on still going development but if you are using astersik or any softswitch you might want to get the dial plan on these features.
   
  In Asterisk dial '#' to transfer and use meetme for conference

Tanmay <ta...@ae...> wrote:
  Hello, 

I have integrated the ATLSIP in .net application but I am not sure how to do the call hold or call transfer or call conference or creating a new line.

any help will be highly appreciated.

thanks

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[OpenSIPStack] ATLSIP crashes

[Yacine A. <yac...@ms...>]

Hello,I'm running a simple softphone using the latest cvs ATLSIP library, e=
verythings works just fine on any computer where Visual studio is installed=
, but the softphone is really instable and crashes each time on any PC with=
out Visual studio installed, i'v compiled ATLSIP release using MT, and i al=
so noticed that Regsvr32 is not able to load the activex on some PCs
_________________________________________________________________
Windows Live Messenger vous offre 30 nouvelles =E9motic=F4nes gratuites, in=
stall=E9es directement dans votre Messenger !
http://www.emoticones-messenger.fr/=

[OpenSIPStack] Error Message on Softphone register

[sobrien <fo...@op...>]

When I attempt to register a soft phone over the network, I get the following Open SIP error, however the soft phone registration packet appears to meet the RFC 3261 spec.  The error is:

2007/07/25 12:29:56.597  Proxy Debug3  *** NO STATIC ROUTE DEFINED *** From: sip:Sean@10.194.124.21 Target: sip:Sean@10.194.124.21
2007/07/25 12:29:56.605  Transport( WRITE ) Debug3  UDP Transport LOOP Detected!!!

How could I go about fixing this?  Thanks.

[OpenSIPStack] ATLSIP 1.0.2 Call Transfer or Conference

[Tanmay <ta...@ae...>]

Hello, 

     I have integrated the ATLSIP  in .net application but I  am not sure how to  do the call hold or call transfer or call conference or creating a new line.

    any help will be highly appreciated.

 thanks

[OpenSIPStack] changing level of microphone during talk

[tomach <to...@dg...>]

Hello!

I try to describe my probelm:
1. I make connection on codec 711.
2. During talk I changed level of microphone several times or even mute it totally.
3. Then I notice that quality of voice is worse then before point 2.

Is it normal? Do You have the same problem? Or is it technical imposible?

Re: [OpenSIPStack] Silence Detection

[tomach <to...@dg...>]

Hello!

About ATLsip
Is it possible to obtain what codec is using during talk? (from the list of audiocodecs0-5)...or the only way is to analyze sip messages coming on SIPMessage event?

BR,
TOm

Re: [OpenSIPStack] STUN support bug?

[Ilian J. C. P. <ip...@so...>]

I'm still cleaning up some issues. Please wait for updates. Probably 
within next week.

Thanks.

- Ilian

Ilian Jeri C. Pinzon wrote:
> Hi,
>
> Now I'm reproducing this problem when using a different server. You're 
> right. The external port is never set.
>
> Thanks for this!
>
> Regards,
> Ilian
>
> H.Kropf wrote:
>   
>> Hi
>>
>> Now in my project STUN-support works correctly
>>
>> For this purpose i has made such changes of a code
>>
>>     

Re: [OpenSIPStack] STUN support bug?

[Ilian J. C. P. <ip...@so...>]

Hi,

Now I'm reproducing this problem when using a different server. You're 
right. The external port is never set.

Thanks for this!

Regards,
Ilian

H.Kropf wrote:
> Hi
>
> Now in my project STUN-support works correctly
>
> For this purpose i has made such changes of a code
>
> *********************************************************
> SIPNATMethods.h
> *********************************************************
> class SIPSTUNClient : public PNatMethod
> {
> ...............
> public:
>   // [+]
>     virtual BOOL GetExternalPort( WORD & externalPort );
> ...............
> protected:
>   // [+]
>     WORD               cachedExternalPort;
> ...............
> };
>
> *********************************************************
> SIPNATMethods.cxx
> *********************************************************
> //-------------------------------------------------------------------------------------
> //-------------------------------------------------------------------------------------
> SIPSTUNClient::SIPSTUNClient(const PString & server,
>                          WORD portBase, WORD portMax,
>                          WORD portPairBase, WORD portPairMax)
> .................
>     cachedExternalPort(0),  // [+]
> .................
> {
> .....................
> .....................
> }
>
> //-------------------------------------------------------------------------------------
> //-------------------------------------------------------------------------------------
> SIPSTUNClient::SIPSTUNClient(const PIPSocket::Address & address, WORD port,
>                          WORD portBase, WORD portMax,
>                          WORD portPairBase, WORD portPairMax)
> .....................
>     cachedExternalPort(0),  // [+]
> .....................
> {
> .....................
> }
>
> //-------------------------------------------------------------------------------------
> //-------------------------------------------------------------------------------------
> BOOL SIPSTUNClient::GetExternalAddress(PIPSocket::Address & externalAddress,
>                                      const PTimeInterval & maxAge)
> {
> .....................
> .....................
>   externalAddress = cachedExternalAddress = mappedAddress->GetIP();
>
>    // [+]
>   cachedExternalPort = (WORD)mappedAddress->port;
> .....................
> }
>
>
> *********************************************************
> SIPTransportManager.h
> *********************************************************
>  
>   class SIPTransportManager : public PObject, public Logger
>   {
> ...............
> public:
>     // [+]
>     BOOL TranslateIPAddress(  PIPSocket::Address & localAddress,   WORD 
> & localPort  );
> ...............
> protexted:
>     // [+]
>     WORD m_TranslationPort; 
> ...............
> };
>
>
> *********************************************************
> SIPTransportManager.cxx
> *********************************************************
>
> //-------------------------------------------------------------------------------------
> //-------------------------------------------------------------------------------------
> SIPSTUNClient::NatTypes SIPTransportManager::SetSTUNServer(
>   const OString & server,
>   WORD udpPortBase,
>   WORD udpPortMax,
>   WORD rtpPortBase,
>   WORD rtpPortMax
> )
> {
> .............................................
>     m_STUNClient = new SIPSTUNClient(server,
>                            udpPortBase, udpPortMax,
>                            rtpPortBase, rtpPortMax );
>
>   SIPSTUNClient::NatTypes type = m_STUNClient->GetNatType();
>
>   if (type != SIPSTUNClient::BlockedNat && type != SIPSTUNClient::OpenNat)
>   {
>     m_STUNClient->GetExternalAddress(m_TranslationAddress);
>
>    // [+]
>     m_STUNClient->GetExternalPort(m_TranslationPort);
>   }
> .............................................
> .............................................
> .............................................
> }
>
> //-------------------------------------------------------------------------------------
> //-------------------------------------------------------------------------------------
> BOOL SIPTransportManager::TranslateIPAddress(
>   PIPSocket::Address & localAddress,
>   WORD & localPort
>  )
> {
> .............................................
> .............................................
>   //if (SIPTransport::IsLocalAddress(remoteAddress))
>   //  return FALSE; // Does not need to be translated
>
>   // Tranlsate it!
>   localAddress = m_TranslationAddress;
>  
>  // [+]
>   localPort = m_TranslationPort;
>
>   return TRUE;
> }
>
>
> *********************************************************
> SoftPhone.cxx
> *********************************************************
>
> BOOL SoftPhoneManager::InitializeSIP()
> {
> .............................................
>   .............................................
> .............................................
> .............................................
>
>   BOOL ok = TRUE;
>
>   if( !m_IsInitialized )
>   {
>
> // [+] begin
>
>   NAT::SIPSTUNClient::NatTypes ntNatType = SetSTUNServer("stun.mysrv.com");
>
>   if(ntNatType == NAT::SIPSTUNClient::ConeNat)
>     GetSoftPhoneInterface()->Event_STUNInit(( true, (char)ntNatType );
>   else
>   {
>     SetSTUNServer("");
>     flip_core::Event_STUNInit( false , (char)ntNatType );
>     GetSoftPhoneInterface()->Event_STUNInit(( false , (char)ntNatType );
>   }
>
> // [+] end
>
>     m_SIPEndPoint->GetProfile().GetTransportProfile().EnableUDP( 
> sipIface, sipPort );
>
> #if ENABLE_TCP_TRANSPORT
>     m_SIPEndPoint->GetProfile().GetTransportProfile().EnableTCP( 
> sipIface, sipPort);
> #endif
>
> .............................................
> .............................................
> .............................................
> .............................................
> }
>
>
>
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Re: [OpenSIPStack] STUN support bug?

[Joegen E. B. <joe...@gm...>]

Hi,

You can use

SIPSTUNClient::NatTypes SIPTransportTools::STUNGetNATType(
  const OString & server
);

To get the NAT type instead of calling SetSTUNServer().   Only call 
SetSTUNSErver() once you have decided whether you want to use STUN or 
not.  Setting the STUN server is a point of no return and will always 
use the stun address through out the lifetime of the application.


H.Kropf wrote:
> Hi
> I could not force STUN-client to work correctly
>
> It is necessary :
> 1. Check NAT presence and Nat type
> 2. If Nat type is ConeNat, in that case it is necessary to enable STUN 
> support
> 3. If Nat type is other, in that case it is necessary to disable STUN 
> support
>
> How to make it?
>
> I have tried to use such method:
>
> in
>
> SoftPhoneManager::InitializeSIP()  function
>
>
> before
>
>     m_SIPEndPoint->GetProfile().GetTransportProfile().EnableUDP( 
> sipIface, sipPort );
>
>
> this piece of code has been inserted
>
> if( SetSTUNServer("stun.mysrv.com") !=  NAT::SIPSTUNClient::ConeNat )   SetSTUNServer("");
>
> and... nothing...
>
> NAT type is SymmetricNat,  (for ConeNat it work correctly) 
>
> The call of function SetSTUNServer("") destroys object SIPTransportManager::m_STUNClient and
> object OpalManager::stun
>
> but OSS for some reason remembers (where? how?) NAT address after SetSTUNServer("stun.mysrv.com") 
> and inserts it into fields of REGISTER requests
>
> Via: SIP/2.0/UDP [NAT IP-address]:5060;....  
> Contact: "100008" <sip:100008@[NAT IP-address]:5060;......
>
>
>
> Tell me please how to solve a problem using only OSS, without other libraries?
>
>
>
>
>
>
>
>
>
>
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>   

Re: [OpenSIPStack] FixedSilence

[Joegen E. B. <joe...@gm...>]

Have you tried working on different threshold levels?   If you 
microphone has noise reduction, you should set silence detection to a 
very low threshold.

Andre Silo wrote:
> Hello,
>    
>   I found a bug on the EnableFixedSilenceDetection(995). When this is enabled voice calls will become one way. Meaning Ican hear whom I called but the called party could not hear me.
>    
>   Can someone help me on this.
>
>        
> ---------------------------------
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>   

[OpenSIPStack] FixedSilence

[Andre S. <eds...@ya...>]

Hello,
   
  I found a bug on the EnableFixedSilenceDetection(995). When this is enabled voice calls will become one way. Meaning Ican hear whom I called but the called party could not hear me.
   
  Can someone help me on this.

       
---------------------------------
Pinpoint customers who are looking for what you sell. 

Re: [OpenSIPStack] STUN support bug?

[H.Kropf <mai...@gl...>]

Hi
I could not force STUN-client to work correctly

It is necessary :
1. Check NAT presence and Nat type
2. If Nat type is ConeNat, in that case it is necessary to enable STUN 
support
3. If Nat type is other, in that case it is necessary to disable STUN 
support

How to make it?

I have tried to use such method:

in

SoftPhoneManager::InitializeSIP()  function


before

    m_SIPEndPoint->GetProfile().GetTransportProfile().EnableUDP( 
sipIface, sipPort );


this piece of code has been inserted

if( SetSTUNServer("stun.mysrv.com") !=  NAT::SIPSTUNClient::ConeNat )   SetSTUNServer("");

and... nothing...

NAT type is SymmetricNat,  (for ConeNat it work correctly) 

The call of function SetSTUNServer("") destroys object SIPTransportManager::m_STUNClient and
object OpalManager::stun

but OSS for some reason remembers (where? how?) NAT address after SetSTUNServer("stun.mysrv.com") 
and inserts it into fields of REGISTER requests

Via: SIP/2.0/UDP [NAT IP-address]:5060;....  
Contact: "100008" <sip:100008@[NAT IP-address]:5060;......



Tell me please how to solve a problem using only OSS, without other libraries?

Re: [OpenSIPStack] STUN support bug?

[tomach <to...@dg...>]

Hi Ilian, 

Still nothing. I checkout whole new project wiht CVS, compile etc...And there is still the same problem...Field contact is with bad local IP address. 

There is only one change I did in ATLSIP it is:

in file ptbuildopts.h:

I removed that: (because otherwise I couldnt compile the code)

/////////////////////////////////////////////////
//
// SAPI speech API (Windows only)
//

#undef P_SAPI 

#if defined(_MSC_VER) && P_SAPI

#pragma include_alias(<sphelper.h>, <@SAPI_DIR@/include/sphelper.h>)
#pragma include_alias(<sapi.h>, <@SAPI_DIR@/include/sapi.h>)
#pragma include_alias(<sapiddk.h>, <@SAPI_DIR@/include/sapiddk.h>)
#pragma include_alias(<SPError.h>, <@SAPI_DIR@/include/SPError.h>)
#pragma include_alias(<SPDebug.h>, <@SAPI_DIR@/include/SPDebug.h>)

#define P_SAPI_LIBRARY "@SAPI_DIR@/Lib/i386/sapi.lib"

#endif

Re: [OpenSIPStack] STUN support bug?

[Ilian J. C. P. <ip...@so...>]

Hi Tom,

Can you get a fresh copy of OssPhone.NET and then run it? Also make sure 
that you have no mods in the OpenSIPStack/ATLSIP projects.

Tell me if you still encounter that problem.

Regards,
Ilian

tomach wrote:
> hmmm its weird that it works in your end. Moving initialize() i tried even before but still were bad header Contact.
>
> Can you compare my stun message with yours? Is it correct? Maybe ATL sip can not parse it or something....
>
> maybe I should force somehow ATLSIP to use stun cos setSTUN is obviously not enough.
> Even it ask STUN for public Ip, it does not use it later at all....
>
> any ideas?
>
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>
>
>   

Re: [OpenSIPStack] STUN support bug?

[Ilian J. C. P. <ip...@so...>]

I still can't reproduce this problem. But I'll check why these are 
needed on their side.

Regards,
Ilian

Joegen E. Baclor wrote:
> Hi,
>
> Thanks for sending in the patches.
>
> Hi Ilian,
>
> Can we validate this patch and come back with a problem analysis?   I'm 
> not particularly sure which part is broken and why modifications are 
> needed to make it work.
>
> Joegen
>
> H.Kropf wrote:
>   
>> Hi
>>
>> Now in my project STUN-support works correctly
>>
>> For this purpose i has made such changes of a code
>>
>> *********************************************************
>> SIPNATMethods.h
>> *********************************************************
>> class SIPSTUNClient : public PNatMethod
>> {
>> ...............
>> public:
>>   // [+]
>>     virtual BOOL GetExternalPort( WORD & externalPort );
>> ...............
>> protected:
>>   // [+]
>>     WORD               cachedExternalPort;
>> ...............
>> };
>>
>> *********************************************************
>> SIPNATMethods.cxx
>> *********************************************************
>> //-------------------------------------------------------------------------------------
>> //-------------------------------------------------------------------------------------
>> SIPSTUNClient::SIPSTUNClient(const PString & server,
>>                          WORD portBase, WORD portMax,
>>                          WORD portPairBase, WORD portPairMax)
>> .................
>>     cachedExternalPort(0),  // [+]
>> .................
>> {
>> .....................
>> .....................
>> }
>>
>> //-------------------------------------------------------------------------------------
>> //-------------------------------------------------------------------------------------
>> SIPSTUNClient::SIPSTUNClient(const PIPSocket::Address & address, WORD port,
>>                          WORD portBase, WORD portMax,
>>                          WORD portPairBase, WORD portPairMax)
>> .....................
>>     cachedExternalPort(0),  // [+]
>> .....................
>> {
>> .....................
>> }
>>
>> //-------------------------------------------------------------------------------------
>> //-------------------------------------------------------------------------------------
>> BOOL SIPSTUNClient::GetExternalAddress(PIPSocket::Address & externalAddress,
>>                                      const PTimeInterval & maxAge)
>> {
>> .....................
>> .....................
>>   externalAddress = cachedExternalAddress = mappedAddress->GetIP();
>>
>>    // [+]
>>   cachedExternalPort = (WORD)mappedAddress->port;
>> .....................
>> }
>>
>>
>> *********************************************************
>> SIPTransportManager.h
>> *********************************************************
>>  
>>   class SIPTransportManager : public PObject, public Logger
>>   {
>> ...............
>> public:
>>     // [+]
>>     BOOL TranslateIPAddress(  PIPSocket::Address & localAddress,   WORD 
>> & localPort  );
>> ...............
>> protexted:
>>     // [+]
>>     WORD m_TranslationPort; 
>> ...............
>> };
>>
>>
>> *********************************************************
>> SIPTransportManager.cxx
>> *********************************************************
>>
>> //-------------------------------------------------------------------------------------
>> //-------------------------------------------------------------------------------------
>> SIPSTUNClient::NatTypes SIPTransportManager::SetSTUNServer(
>>   const OString & server,
>>   WORD udpPortBase,
>>   WORD udpPortMax,
>>   WORD rtpPortBase,
>>   WORD rtpPortMax
>> )
>> {
>> .............................................
>>     m_STUNClient = new SIPSTUNClient(server,
>>                            udpPortBase, udpPortMax,
>>                            rtpPortBase, rtpPortMax );
>>
>>   SIPSTUNClient::NatTypes type = m_STUNClient->GetNatType();
>>
>>   if (type != SIPSTUNClient::BlockedNat && type != SIPSTUNClient::OpenNat)
>>   {
>>     m_STUNClient->GetExternalAddress(m_TranslationAddress);
>>
>>    // [+]
>>     m_STUNClient->GetExternalPort(m_TranslationPort);
>>   }
>> .............................................
>> .............................................
>> .............................................
>> }
>>
>> //-------------------------------------------------------------------------------------
>> //-------------------------------------------------------------------------------------
>> BOOL SIPTransportManager::TranslateIPAddress(
>>   PIPSocket::Address & localAddress,
>>   WORD & localPort
>>  )
>> {
>> .............................................
>> .............................................
>>   //if (SIPTransport::IsLocalAddress(remoteAddress))
>>   //  return FALSE; // Does not need to be translated
>>
>>   // Tranlsate it!
>>   localAddress = m_TranslationAddress;
>>  
>>  // [+]
>>   localPort = m_TranslationPort;
>>
>>   return TRUE;
>> }
>>
>>
>> *********************************************************
>> SoftPhone.cxx
>> *********************************************************
>>
>> BOOL SoftPhoneManager::InitializeSIP()
>> {
>> .............................................
>>   .............................................
>> .............................................
>> .............................................
>>
>>   BOOL ok = TRUE;
>>
>>   if( !m_IsInitialized )
>>   {
>>
>> // [+] begin
>>
>>   NAT::SIPSTUNClient::NatTypes ntNatType = SetSTUNServer("stun.mysrv.com");
>>
>>   if(ntNatType == NAT::SIPSTUNClient::ConeNat)
>>     GetSoftPhoneInterface()->Event_STUNInit(( true, (char)ntNatType );
>>   else
>>   {
>>     SetSTUNServer("");
>>     flip_core::Event_STUNInit( false , (char)ntNatType );
>>     GetSoftPhoneInterface()->Event_STUNInit(( false , (char)ntNatType );
>>   }
>>
>> // [+] end
>>
>>     m_SIPEndPoint->GetProfile().GetTransportProfile().EnableUDP( 
>> sipIface, sipPort );
>>
>> #if ENABLE_TCP_TRANSPORT
>>     m_SIPEndPoint->GetProfile().GetTransportProfile().EnableTCP( 
>> sipIface, sipPort);
>> #endif
>>
>> .............................................
>> .............................................
>> .............................................
>> .............................................
>> }
>>
>>
>>
>> -------------------------------------------------------------------------
>> This SF.net email is sponsored by: Splunk Inc.
>> Still grepping through log files to find problems?  Stop.
>> Now Search log events and configuration files using AJAX and a browser.
>> Download your FREE copy of Splunk now >>  http://get.splunk.com/
>> _______________________________________________
>> opensipstack-devel mailing list
>> ope...@li...
>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>
>>   
>>     
>
>
> -------------------------------------------------------------------------
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> Still grepping through log files to find problems?  Stop.
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>
>
>   

Re: [OpenSIPStack] STUN support bug?

[Joegen E. B. <joe...@gm...>]

Hi,

Thanks for sending in the patches.

Hi Ilian,

Can we validate this patch and come back with a problem analysis?   I'm 
not particularly sure which part is broken and why modifications are 
needed to make it work.

Joegen

H.Kropf wrote:
> Hi
>
> Now in my project STUN-support works correctly
>
> For this purpose i has made such changes of a code
>
> *********************************************************
> SIPNATMethods.h
> *********************************************************
> class SIPSTUNClient : public PNatMethod
> {
> ...............
> public:
>   // [+]
>     virtual BOOL GetExternalPort( WORD & externalPort );
> ...............
> protected:
>   // [+]
>     WORD               cachedExternalPort;
> ...............
> };
>
> *********************************************************
> SIPNATMethods.cxx
> *********************************************************
> //-------------------------------------------------------------------------------------
> //-------------------------------------------------------------------------------------
> SIPSTUNClient::SIPSTUNClient(const PString & server,
>                          WORD portBase, WORD portMax,
>                          WORD portPairBase, WORD portPairMax)
> .................
>     cachedExternalPort(0),  // [+]
> .................
> {
> .....................
> .....................
> }
>
> //-------------------------------------------------------------------------------------
> //-------------------------------------------------------------------------------------
> SIPSTUNClient::SIPSTUNClient(const PIPSocket::Address & address, WORD port,
>                          WORD portBase, WORD portMax,
>                          WORD portPairBase, WORD portPairMax)
> .....................
>     cachedExternalPort(0),  // [+]
> .....................
> {
> .....................
> }
>
> //-------------------------------------------------------------------------------------
> //-------------------------------------------------------------------------------------
> BOOL SIPSTUNClient::GetExternalAddress(PIPSocket::Address & externalAddress,
>                                      const PTimeInterval & maxAge)
> {
> .....................
> .....................
>   externalAddress = cachedExternalAddress = mappedAddress->GetIP();
>
>    // [+]
>   cachedExternalPort = (WORD)mappedAddress->port;
> .....................
> }
>
>
> *********************************************************
> SIPTransportManager.h
> *********************************************************
>  
>   class SIPTransportManager : public PObject, public Logger
>   {
> ...............
> public:
>     // [+]
>     BOOL TranslateIPAddress(  PIPSocket::Address & localAddress,   WORD 
> & localPort  );
> ...............
> protexted:
>     // [+]
>     WORD m_TranslationPort; 
> ...............
> };
>
>
> *********************************************************
> SIPTransportManager.cxx
> *********************************************************
>
> //-------------------------------------------------------------------------------------
> //-------------------------------------------------------------------------------------
> SIPSTUNClient::NatTypes SIPTransportManager::SetSTUNServer(
>   const OString & server,
>   WORD udpPortBase,
>   WORD udpPortMax,
>   WORD rtpPortBase,
>   WORD rtpPortMax
> )
> {
> .............................................
>     m_STUNClient = new SIPSTUNClient(server,
>                            udpPortBase, udpPortMax,
>                            rtpPortBase, rtpPortMax );
>
>   SIPSTUNClient::NatTypes type = m_STUNClient->GetNatType();
>
>   if (type != SIPSTUNClient::BlockedNat && type != SIPSTUNClient::OpenNat)
>   {
>     m_STUNClient->GetExternalAddress(m_TranslationAddress);
>
>    // [+]
>     m_STUNClient->GetExternalPort(m_TranslationPort);
>   }
> .............................................
> .............................................
> .............................................
> }
>
> //-------------------------------------------------------------------------------------
> //-------------------------------------------------------------------------------------
> BOOL SIPTransportManager::TranslateIPAddress(
>   PIPSocket::Address & localAddress,
>   WORD & localPort
>  )
> {
> .............................................
> .............................................
>   //if (SIPTransport::IsLocalAddress(remoteAddress))
>   //  return FALSE; // Does not need to be translated
>
>   // Tranlsate it!
>   localAddress = m_TranslationAddress;
>  
>  // [+]
>   localPort = m_TranslationPort;
>
>   return TRUE;
> }
>
>
> *********************************************************
> SoftPhone.cxx
> *********************************************************
>
> BOOL SoftPhoneManager::InitializeSIP()
> {
> .............................................
>   .............................................
> .............................................
> .............................................
>
>   BOOL ok = TRUE;
>
>   if( !m_IsInitialized )
>   {
>
> // [+] begin
>
>   NAT::SIPSTUNClient::NatTypes ntNatType = SetSTUNServer("stun.mysrv.com");
>
>   if(ntNatType == NAT::SIPSTUNClient::ConeNat)
>     GetSoftPhoneInterface()->Event_STUNInit(( true, (char)ntNatType );
>   else
>   {
>     SetSTUNServer("");
>     flip_core::Event_STUNInit( false , (char)ntNatType );
>     GetSoftPhoneInterface()->Event_STUNInit(( false , (char)ntNatType );
>   }
>
> // [+] end
>
>     m_SIPEndPoint->GetProfile().GetTransportProfile().EnableUDP( 
> sipIface, sipPort );
>
> #if ENABLE_TCP_TRANSPORT
>     m_SIPEndPoint->GetProfile().GetTransportProfile().EnableTCP( 
> sipIface, sipPort);
> #endif
>
> .............................................
> .............................................
> .............................................
> .............................................
> }
>
>
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems?  Stop.
> Now Search log events and configuration files using AJAX and a browser.
> Download your FREE copy of Splunk now >>  http://get.splunk.com/
> _______________________________________________
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> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>   

[OpenSIPStack] STUN support bug?

[H.Kropf <mai...@gl...>]

Hi

Now in my project STUN-support works correctly

For this purpose i has made such changes of a code

*********************************************************
SIPNATMethods.h
*********************************************************
class SIPSTUNClient : public PNatMethod
{
...............
public:
  // [+]
    virtual BOOL GetExternalPort( WORD & externalPort );
...............
protected:
  // [+]
    WORD               cachedExternalPort;
...............
};

*********************************************************
SIPNATMethods.cxx
*********************************************************
//-------------------------------------------------------------------------------------
//-------------------------------------------------------------------------------------
SIPSTUNClient::SIPSTUNClient(const PString & server,
                         WORD portBase, WORD portMax,
                         WORD portPairBase, WORD portPairMax)
.................
    cachedExternalPort(0),  // [+]
.................
{
.....................
.....................
}

//-------------------------------------------------------------------------------------
//-------------------------------------------------------------------------------------
SIPSTUNClient::SIPSTUNClient(const PIPSocket::Address & address, WORD port,
                         WORD portBase, WORD portMax,
                         WORD portPairBase, WORD portPairMax)
.....................
    cachedExternalPort(0),  // [+]
.....................
{
.....................
}

//-------------------------------------------------------------------------------------
//-------------------------------------------------------------------------------------
BOOL SIPSTUNClient::GetExternalAddress(PIPSocket::Address & externalAddress,
                                     const PTimeInterval & maxAge)
{
.....................
.....................
  externalAddress = cachedExternalAddress = mappedAddress->GetIP();

   // [+]
  cachedExternalPort = (WORD)mappedAddress->port;
.....................
}


*********************************************************
SIPTransportManager.h
*********************************************************
 
  class SIPTransportManager : public PObject, public Logger
  {
...............
public:
    // [+]
    BOOL TranslateIPAddress(  PIPSocket::Address & localAddress,   WORD 
& localPort  );
...............
protexted:
    // [+]
    WORD m_TranslationPort; 
...............
};


*********************************************************
SIPTransportManager.cxx
*********************************************************

//-------------------------------------------------------------------------------------
//-------------------------------------------------------------------------------------
SIPSTUNClient::NatTypes SIPTransportManager::SetSTUNServer(
  const OString & server,
  WORD udpPortBase,
  WORD udpPortMax,
  WORD rtpPortBase,
  WORD rtpPortMax
)
{
.............................................
    m_STUNClient = new SIPSTUNClient(server,
                           udpPortBase, udpPortMax,
                           rtpPortBase, rtpPortMax );

  SIPSTUNClient::NatTypes type = m_STUNClient->GetNatType();

  if (type != SIPSTUNClient::BlockedNat && type != SIPSTUNClient::OpenNat)
  {
    m_STUNClient->GetExternalAddress(m_TranslationAddress);

   // [+]
    m_STUNClient->GetExternalPort(m_TranslationPort);
  }
.............................................
.............................................
.............................................
}

//-------------------------------------------------------------------------------------
//-------------------------------------------------------------------------------------
BOOL SIPTransportManager::TranslateIPAddress(
  PIPSocket::Address & localAddress,
  WORD & localPort
 )
{
.............................................
.............................................
  //if (SIPTransport::IsLocalAddress(remoteAddress))
  //  return FALSE; // Does not need to be translated

  // Tranlsate it!
  localAddress = m_TranslationAddress;
 
 // [+]
  localPort = m_TranslationPort;

  return TRUE;
}


*********************************************************
SoftPhone.cxx
*********************************************************

BOOL SoftPhoneManager::InitializeSIP()
{
.............................................
  .............................................
.............................................
.............................................

  BOOL ok = TRUE;

  if( !m_IsInitialized )
  {

// [+] begin

  NAT::SIPSTUNClient::NatTypes ntNatType = SetSTUNServer("stun.mysrv.com");

  if(ntNatType == NAT::SIPSTUNClient::ConeNat)
    GetSoftPhoneInterface()->Event_STUNInit(( true, (char)ntNatType );
  else
  {
    SetSTUNServer("");
    flip_core::Event_STUNInit( false , (char)ntNatType );
    GetSoftPhoneInterface()->Event_STUNInit(( false , (char)ntNatType );
  }

// [+] end

    m_SIPEndPoint->GetProfile().GetTransportProfile().EnableUDP( 
sipIface, sipPort );

#if ENABLE_TCP_TRANSPORT
    m_SIPEndPoint->GetProfile().GetTransportProfile().EnableTCP( 
sipIface, sipPort);
#endif

.............................................
.............................................
.............................................
.............................................
}

Re: [OpenSIPStack] STUN support bug?

[tomach <to...@dg...>]

hmmm its weird that it works in your end. Moving initialize() i tried even before but still were bad header Contact.

Can you compare my stun message with yours? Is it correct? Maybe ATL sip can not parse it or something....

maybe I should force somehow ATLSIP to use stun cos setSTUN is obviously not enough.
Even it ask STUN for public Ip, it does not use it later at all....

any ideas?

Re: [OpenSIPStack] Incoming requests handling in upper registration mode

[Joegen E. B. <joe...@gm...>]

Hi Andrew,

I fully understand your situation.  I think the best solution here is to 
let PortaSIP rewrite the To URI as well.  Is this not a posibility?   As 
you can see OpenSBC tried its best here to act closest to the specs how 
the request would have been routed.  Let's say hypothetically that this 
maybe configurable.    Assume there is a user registered to OpenSBC as 
sip:61...@op....  Imagine what will happen if OpenSBC received 
an INVITE like this:

INVITE sip:61...@fw... SIP/2.0
From: "alice" <sip:al...@wo...>;tag=123
To: "Free World Dial-up Echo Server" <sip:61...@fw...>

This would clearly be a relay at first glance and OpenSBC should forward 
this to the real destination.   However, since 613 might be a registered 
user, in this case sip:61...@op..., the hypothetical behavior 
would be to back off and send the call originally intended for 
sip:61...@fw... to the 613 user of opensbcdomain.com.

At first glance this is doable but would entail more work for OpenSBC 
when in fact the sender could have just written the To URI correctly. 

I am not closing my doors for this case.  If you think there is an 
important real-world reason to let OpenSBC handle this behavior then 
lets start discussing a possible path to take.

Joegen

Andrew Pogrebennyk wrote:
> Dear all,
>
> I am trying to use OpenSBC in b2bua upper registration mode with 
> PortaSIP softswitch. Registration and outgoing calls for UAs behind 
> OpenSBC are OK, requests are forwarded through OpenSBC to PortaSIP. 
> Contact header hijacking works like a charm. But when it comes to 
> incoming calls or, say, NOTIFY from external application server, there 
> is a trouble. Rules that govern distinguishing between local and remote 
> domains do backfire: since incoming requests hit PortaSIP first, an IP 
> in To header does not match the IP address of OpenSBC's listener 
> interface. It contains the IP address of PortaSIP.
> So OpenSBC first checks the IP in Request-URI, sees that it matches the 
> IP address of OpenSBC's listener interface, then checks the To header, 
> sees it does not match AND sends request back to PortaSIP. This leads to 
> a routing loop.
> Can I instruct OpenSBC to overwrite IP address in To header with an AOR 
> it knows for the request target and then send it out? Or I am missing 
> something? Thanks in advance.
>
>   

Re: [OpenSIPStack] Problem with last change in SIPUDPTransport::ReadPacket()

[Joegen E. B. <joe...@gm...>]

You may safely return here.  I've patched CVS accordingly.


Gustavo Curetti wrote:
> Hi Joegen:
> =20
> I have a problem with the last change in SIPUDPTransport::ReadPacket():=

> =20
> if( length =3D=3D 0 )
> {
>  packet.SetSize(0);
>  return TRUE; ----> return FALSE;
> }
> =20
> when SIPUDPTransport::ReadPacket() return false,=20
> SIPTransport::ProcessRead() also return false and
> SIPTransport::ReadThread::Main() finish and no more sip messages are=20
> processed.
> =20
> when I debugged SIPUDPSocketList::ReadFrom(), i found this:
> =20
> BOOL SIPUDPSocketList::ReadFrom()
>  BOOL ok =3D socket.ReadFrom( bytes, len, addr, port ) --> len =3D 2048=
,=20
> addr =3D 127.0.0.1, port =3D 38890
>   if (os_recvfrom(buf, len, 0, sa, &size))=20
>    int recvResult =3D ::recvfrom(os_handle, (char *)buf, len, flags,=20
> from, fromlen);
>    if (!ConvertOSError(recvResult, LastReadError))--> recvResult =3D -1=
,=20
> LastReadError =3D 0
>     BOOL ok =3D ConvertOSError(status, lastError, osError);
>      return PChannel::ConvertOSError(-2, lastError, osError);
>       osError =3D GetLastError(); *(osError =3D 10054 =3D WSAECONNRESET=
)=20
> (Connection reset by peer)
> *      osError |=3D PWIN32ErrorFlag; (osError =3D 1073751878)
>       lastError =3D Miscellaneous; (lastError =3D 12)
>   return lastReadCount > 0;  (lastReadCount =3D 0)
>  LOG( LogError(), "UDP Socket Read Error (" << socket.GetErrorText()=20
> << ")" );=20
>   PString PChannel::GetErrorText(Errors lastError, int osError)=20
> (lastError =3D 12) (osError =3D 1073751881)
>    DWORD err =3D osError & ~PWIN32ErrorFlag; *(err =3D 10057 =3D=20
> WSAENOTCONN) (Socket not connected)
> * /49:28:50.887 ERR: [CID=3D0x0000] UDP Socket Read Error (Socket not=20
> connected)
> / bytesRead =3D 0;
>  return TRUE;  /// return TRUE here.  error in ReadFrom is benign
>
> I don't understand why the errors are differents, but when I searched=20
> for the original error (10054), i found this:
> =20
> "If sending a datagram using the sendto function results in an "ICMP=20
> port unreachable" response and the select function is set for readfds, =

> the program returns 1 and the subsequent call to the recvfrom function =

> does not work with a WSAECONNRESET (10054) error response"
> =20
> Maybe some ICMP validation go wrong with winsock. I attach a log.
> =20
> Thanks for your help.
> =20
> Gustavo
> =20
>
> -----------------------------------------------------------------------=
-
> Env=EDa mensajes de correo electr=F3nico directamente a tu blog con MSN=
=2E=20
> Carga chistes, fotograf=EDas y muchas otras cosas. Es gratis.=20
> <http://clk.atdmt.com/MSN/go/msnnksac0030000001msn/direct/01/?href=3Dht=
tp://www.imagine-msn.com/spaces>

[OpenSIPStack] Problem with last change in SIPUDPTransport::ReadPacket()

[Gustavo C. <cur...@ho...>]

Hi Joegen:
=20
I have a problem with the last change in SIPUDPTransport::ReadPacket():
=20
if( length =3D=3D 0 )
{
 packet.SetSize(0);
 return TRUE; ----> return FALSE;
}
=20
when SIPUDPTransport::ReadPacket() return false, SIPTransport::ProcessRead(=
) also return false and=20
SIPTransport::ReadThread::Main() finish and no more sip messages are proces=
sed.
=20
when I debugged SIPUDPSocketList::ReadFrom(), i found this:
=20
BOOL SIPUDPSocketList::ReadFrom() BOOL ok =3D socket.ReadFrom( bytes, len, =
addr, port ) --> len =3D 2048, addr =3D 127.0.0.1, port =3D 38890  if (os_r=
ecvfrom(buf, len, 0, sa, &size))    int recvResult =3D ::recvfrom(os_handle=
, (char *)buf, len, flags, from, fromlen);   if (!ConvertOSError(recvResult=
, LastReadError))--> recvResult =3D -1, LastReadError =3D 0    BOOL ok =3D =
ConvertOSError(status, lastError, osError);     return PChannel::ConvertOSE=
rror(-2, lastError, osError);      osError =3D GetLastError(); (osError =3D=
 10054 =3D WSAECONNRESET) (Connection reset by peer)      osError |=3D PWIN=
32ErrorFlag; (osError =3D 1073751878)      lastError =3D Miscellaneous; (la=
stError =3D 12)  return lastReadCount > 0;  (lastReadCount =3D 0) LOG( LogE=
rror(), "UDP Socket Read Error (" << socket.GetErrorText() << ")" );   PStr=
ing PChannel::GetErrorText(Errors lastError, int osError) (lastError =3D 12=
) (osError =3D 1073751881)   DWORD err =3D osError & ~PWIN32ErrorFlag; (err=
 =3D 10057 =3D WSAENOTCONN) (Socket not connected) 49:28:50.887 ERR: [CID=
=3D0x0000] UDP Socket Read Error (Socket not connected) bytesRead =3D 0; re=
turn TRUE;  /// return TRUE here.  error in ReadFrom is benign
I don't understand why the errors are differents, but when I searched for t=
he original error (10054), i found this:
=20
"If sending a datagram using the sendto function results in an "ICMP port u=
nreachable" response and the select function is set for readfds, the progra=
m returns 1 and the subsequent call to the recvfrom function does not work =
with a WSAECONNRESET (10054) error response"
=20
Maybe some ICMP validation go wrong with winsock. I attach a log.=20
=20
Thanks for your help.
=20
Gustavo
=20
_________________________________________________________________
Descubre Live.com - tu propia p=E1gina de inicio, personalizada para ver r=
=E1pidamente todo lo que te interesa en un mismo sitio.
http://www.live.com/getstarted=

[OpenSIPStack] Incoming requests handling in upper registration mode

[Andrew P. <and...@po...>]

Dear all,

I am trying to use OpenSBC in b2bua upper registration mode with 
PortaSIP softswitch. Registration and outgoing calls for UAs behind 
OpenSBC are OK, requests are forwarded through OpenSBC to PortaSIP. 
Contact header hijacking works like a charm. But when it comes to 
incoming calls or, say, NOTIFY from external application server, there 
is a trouble. Rules that govern distinguishing between local and remote 
domains do backfire: since incoming requests hit PortaSIP first, an IP 
in To header does not match the IP address of OpenSBC's listener 
interface. It contains the IP address of PortaSIP.
So OpenSBC first checks the IP in Request-URI, sees that it matches the 
IP address of OpenSBC's listener interface, then checks the To header, 
sees it does not match AND sends request back to PortaSIP. This leads to 
a routing loop.
Can I instruct OpenSBC to overwrite IP address in To header with an AOR 
it knows for the request target and then send it out? Or I am missing 
something? Thanks in advance.

-- 
Sincerely,
Andrew Pogrebennyk

Re: [OpenSIPStack] STUN support bug?

[Ilian J. C. P. <ip...@so...>]

Hi Tom,

It works fine in my end. However, you could try moving 
ATLSIP->InitializeSIP() to  void DoLogin() from System::Void Form1_Load. 
I'm assuming you're using the OSSPhone .NET version. This guarantees 
that the transport is initialized *AFTER* setting the STUN address.

But I'm encountering one-way audio (inbound is silent) with calls using 
STUN. Logs says I'm in a Port Restricted NAT. I'll still have to check 
why this is happening.

Regards,
Ilian

Ilian Jeri C. Pinzon wrote:
> Hi Tom,
>
> Thanks for reporting this. I'll have to confirm this first and see 
> what's wrong.
>
> Regards,
> Ilian
>
> tomach wrote:
>   
>> Sorry but I do not know whats wrong. I tried to use STUN server. So in OSSPhone I set everthing as it should be:
>> Registrar, OutboundProxy and STUN server
>>
>> then I tried to register and I got no respond from my registrar server. 
>>
>> Using ethereal I sniffed my eth and I noticed that:
>> 1. Ossphone sends request to stun.
>> 2. Stun answers with this kind of answer. (public address of ossphone is 192.168.2.14 and  local is 192.168.44.46):
>>
>> Simple Traversal of UDP Through NAT
>>     Message Type: Binding Response (0x0101)
>>     Message Length: 0x0044
>>     Message Transaction ID: 0163E14BE5515702A2407C07C23C6073
>>     Attributes
>>         Attribute: MAPPED-ADDRESS
>>             Attribute Type: MAPPED-ADDRESS (0x0001)
>>             Attribute Length: 8
>>             Protocol Family: IPv4 (0x0001)
>>             Port: 50666
>>             IP: 192.168.44.46 (192.168.44.46)
>>         Attribute: SOURCE-ADDRESS
>>             Attribute Type: SOURCE-ADDRESS (0x0004)
>>             Attribute Length: 8
>>             Protocol Family: IPv4 (0x0001)
>>             Port: 3478
>>             IP: 192.168.44.254 (192.168.44.254)
>>         Attribute: CHANGED-ADDRESS
>>             Attribute Type: CHANGED-ADDRESS (0x0005)
>>             Attribute Length: 8
>>             Protocol Family: IPv4 (0x0001)
>>             Port: 3479
>>             IP: 192.168.2.14 (192.168.2.14)
>>         Attribute: XOR_MAPPED_ADDRESS
>>             Attribute Type: XOR_MAPPED_ADDRESS (0x8020)
>>             Attribute Length: 8
>>             Protocol Family: IPv4 (0x0001)
>>             Port (XOR-d): 50313
>>             Port: 50666
>>             IP (XOR-d): 193.203.205.101 (193.203.205.101)
>>             IP: 192.168.44.46 (192.168.44.46)
>>         Attribute: SERVER
>>             Attribute Type: SERVER (0x8022)
>>             Attribute Length: 16
>>             Server version: Vovida.org 0.96
>>
>> 3. Then ossphone sends register message to sipserver. Where Contact and Via headers have incorrect ossphone address. Should be public 192.168.2.14 and there is  local 192.168.44.46.
>> 4. Sip server does not response because in Contact and Via headers is wrong ossphone address.
>>
>> Do you have the same behaviour? why Contact field and via arent changed to public in register message?
>>
>> -------------------------------------------------------------------------
>> This SF.net email is sponsored by: Splunk Inc.
>> Still grepping through log files to find problems?  Stop.
>> Now Search log events and configuration files using AJAX and a browser.
>> Download your FREE copy of Splunk now >>  http://get.splunk.com/
>> _______________________________________________
>> opensipstack-devel mailing list
>> ope...@li...
>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>
>>
>>   
>>     
>
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems?  Stop.
> Now Search log events and configuration files using AJAX and a browser.
> Download your FREE copy of Splunk now >>  http://get.splunk.com/
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
>   

Re: [OpenSIPStack] STUN support bug?

[Ilian J. C. P. <ip...@so...>]

Hi Tom,

Thanks for reporting this. I'll have to confirm this first and see 
what's wrong.

Regards,
Ilian

tomach wrote:
> Sorry but I do not know whats wrong. I tried to use STUN server. So in OSSPhone I set everthing as it should be:
> Registrar, OutboundProxy and STUN server
>
> then I tried to register and I got no respond from my registrar server. 
>
> Using ethereal I sniffed my eth and I noticed that:
> 1. Ossphone sends request to stun.
> 2. Stun answers with this kind of answer. (public address of ossphone is 192.168.2.14 and  local is 192.168.44.46):
>
> Simple Traversal of UDP Through NAT
>     Message Type: Binding Response (0x0101)
>     Message Length: 0x0044
>     Message Transaction ID: 0163E14BE5515702A2407C07C23C6073
>     Attributes
>         Attribute: MAPPED-ADDRESS
>             Attribute Type: MAPPED-ADDRESS (0x0001)
>             Attribute Length: 8
>             Protocol Family: IPv4 (0x0001)
>             Port: 50666
>             IP: 192.168.44.46 (192.168.44.46)
>         Attribute: SOURCE-ADDRESS
>             Attribute Type: SOURCE-ADDRESS (0x0004)
>             Attribute Length: 8
>             Protocol Family: IPv4 (0x0001)
>             Port: 3478
>             IP: 192.168.44.254 (192.168.44.254)
>         Attribute: CHANGED-ADDRESS
>             Attribute Type: CHANGED-ADDRESS (0x0005)
>             Attribute Length: 8
>             Protocol Family: IPv4 (0x0001)
>             Port: 3479
>             IP: 192.168.2.14 (192.168.2.14)
>         Attribute: XOR_MAPPED_ADDRESS
>             Attribute Type: XOR_MAPPED_ADDRESS (0x8020)
>             Attribute Length: 8
>             Protocol Family: IPv4 (0x0001)
>             Port (XOR-d): 50313
>             Port: 50666
>             IP (XOR-d): 193.203.205.101 (193.203.205.101)
>             IP: 192.168.44.46 (192.168.44.46)
>         Attribute: SERVER
>             Attribute Type: SERVER (0x8022)
>             Attribute Length: 16
>             Server version: Vovida.org 0.96
>
> 3. Then ossphone sends register message to sipserver. Where Contact and Via headers have incorrect ossphone address. Should be public 192.168.2.14 and there is  local 192.168.44.46.
> 4. Sip server does not response because in Contact and Via headers is wrong ossphone address.
>
> Do you have the same behaviour? why Contact field and via arent changed to public in register message?
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems?  Stop.
> Now Search log events and configuration files using AJAX and a browser.
> Download your FREE copy of Splunk now >>  http://get.splunk.com/
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
>   

Re: [OpenSIPStack] STUN support bug?

[tomach <to...@dg...>]

Sorry but I do not know whats wrong. I tried to use STUN server. So in OSSPhone I set everthing as it should be:
Registrar, OutboundProxy and STUN server

then I tried to register and I got no respond from my registrar server. 

Using ethereal I sniffed my eth and I noticed that:
1. Ossphone sends request to stun.
2. Stun answers with this kind of answer. (public address of ossphone is 192.168.2.14 and  local is 192.168.44.46):

Simple Traversal of UDP Through NAT
    Message Type: Binding Response (0x0101)
    Message Length: 0x0044
    Message Transaction ID: 0163E14BE5515702A2407C07C23C6073
    Attributes
        Attribute: MAPPED-ADDRESS
            Attribute Type: MAPPED-ADDRESS (0x0001)
            Attribute Length: 8
            Protocol Family: IPv4 (0x0001)
            Port: 50666
            IP: 192.168.44.46 (192.168.44.46)
        Attribute: SOURCE-ADDRESS
            Attribute Type: SOURCE-ADDRESS (0x0004)
            Attribute Length: 8
            Protocol Family: IPv4 (0x0001)
            Port: 3478
            IP: 192.168.44.254 (192.168.44.254)
        Attribute: CHANGED-ADDRESS
            Attribute Type: CHANGED-ADDRESS (0x0005)
            Attribute Length: 8
            Protocol Family: IPv4 (0x0001)
            Port: 3479
            IP: 192.168.2.14 (192.168.2.14)
        Attribute: XOR_MAPPED_ADDRESS
            Attribute Type: XOR_MAPPED_ADDRESS (0x8020)
            Attribute Length: 8
            Protocol Family: IPv4 (0x0001)
            Port (XOR-d): 50313
            Port: 50666
            IP (XOR-d): 193.203.205.101 (193.203.205.101)
            IP: 192.168.44.46 (192.168.44.46)
        Attribute: SERVER
            Attribute Type: SERVER (0x8022)
            Attribute Length: 16
            Server version: Vovida.org 0.96

3. Then ossphone sends register message to sipserver. Where Contact and Via headers have incorrect ossphone address. Should be public 192.168.2.14 and there is  local 192.168.44.46.
4. Sip server does not response because in Contact and Via headers is wrong ossphone address.

Do you have the same behaviour? why Contact field and via arent changed to public in register message?