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Re: [OpenSIPStack] STUN support bug?

From: H.Kropf <mai...@gl...> - 2007-07-24 14:01:27

Ilian Jeri C. Pinzon wrote:

Thanks.
I shall try it.
Also look at it

===================================================
OpalOSSEndPoint.cxx
===================================================

BOOL OpalOSSEndPoint::SendRegister(
  const PString & userName,
  const PString & password,
  const PString & uri,
  const PString & _domain,
  BOOL synchronous
)
{
........  
    WORD contactPort;
........  
........  
    if( !transport->GetListenerAddress( SIPTransport::UDP, 
targetAddress, contactAddress, contactPort ) )
      return FALSE;

    OStringStream contact;
    contact << contactAddress << ":" << contactPort; 

    profile.AppendContact( contact.str() );
........  
........  
........  

}

===================================================
SIPTransportManager.cxx
===================================================
BOOL SIPTransportManager::GetListenerAddress(
  SIPTransport::Protocol proto,
  const PIPSocket::Address & target,
  PIPSocket::Address & listenerAddress,
  WORD & listenerPort
)
{
  ....
  ....
  TranslateIPAddress( listenerAddress, listenerPort );   //// WORD 
listenerPort?  instead of const PIPSocket::Address ?

  return ok;
}

//----------------------------------------------------------------
BOOL SIPTransportManager::TranslateIPAddress(
  PIPSocket::Address & localAddress,
  const PIPSocket::Address & remoteAddress
 )
{
  if (m_TranslationAddress.IsLoopback())
    return FALSE; // Have nothing to translate it to

  if (!m_TranslationAddress.IsValid())
    return FALSE; // Have nothing to translate it to

  if (!SIPTransport::IsLocalAddress(localAddress))
    return FALSE; // Is already translated

  if (SIPTransport::IsLocalAddress(remoteAddress))    //  !!!
    return FALSE; // Does not need to be translated

  // Tranlsate it!
  localAddress = m_TranslationAddress;
  return TRUE;
}
//----------------------------------------------------------------




> Hi,
>
> SetSTUNServer should work when it is called before the transport has
> been set up. Have you tried calling this during initialization?
>
> Regards,
> Ilian
>
> H.Kropf wrote:
>   
>> Hello
>>
>> I try to use OSS SoftPhone client (OSS 1.1.7 CVS) with server OpenSER 
>> and with NAT of type NAT::SIPSTUNClient::ConeNat
>> Calling of function SetSTUNServer connect succefully to my STUN-server.  
>> SoftPhone receive true NAT IP-address and NAT port.
>> But after, when i call SoftPhone::SendRegister() then  OpenSER register 
>> client like "NAT, without STUN"
>>
>> Exemples: (correct and wrong)
>> ===============================
>>
>> REGISTER requests via NAT using STUN:  (aliases: 111.111.111.111 - 
>> instead of local SIP-client IP, 222.222.222.222 - instead of NAT IP, 
>> 333.333.333.333 - instead of SIP-server IP)
>>
>> CORRECT (hardware Zyxel SIP-client)
>>   
>>

Re: [OpenSIPStack] STUN support bug?

From: Ilian J. C. P. <ip...@so...> - 2007-07-24 10:44:13

Will do.

Regards,
Ilian

Joegen E. Baclor wrote:
> Hi Ilian,
>
> Can we change the STUN code so that it asserts when called after the 
> transport has already been initialized?
>
>
> Ilian Jeri C. Pinzon wrote:
>   
>> Hi,
>>
>> SetSTUNServer should work when it is called before the transport has
>> been set up. Have you tried calling this during initialization?
>>
>> Regards,
>> Ilian
>>
>> H.Kropf wrote:
>>   
>>     
>>> Hello
>>>
>>> I try to use OSS SoftPhone client (OSS 1.1.7 CVS) with server OpenSER 
>>> and with NAT of type NAT::SIPSTUNClient::ConeNat
>>> Calling of function SetSTUNServer connect succefully to my STUN-server.  
>>> SoftPhone receive true NAT IP-address and NAT port.
>>> But after, when i call SoftPhone::SendRegister() then  OpenSER register 
>>> client like "NAT, without STUN"
>>>
>>> Exemples: (correct and wrong)
>>> ===============================
>>>
>>> REGISTER requests via NAT using STUN:  (aliases: 111.111.111.111 - 
>>> instead of local SIP-client IP, 222.222.222.222 - instead of NAT IP, 
>>> 333.333.333.333 - instead of SIP-server IP)
>>>
>>> CORRECT (hardware Zyxel SIP-client)
>>>   
>>>     
>>>       
>> <snip>
>>
>>
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>>   
>>     
>
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Re: [OpenSIPStack] STUN support bug?

From: Joegen E. B. <joe...@gm...> - 2007-07-24 10:11:01

Hi Ilian,

Can we change the STUN code so that it asserts when called after the 
transport has already been initialized?


Ilian Jeri C. Pinzon wrote:
> Hi,
>
> SetSTUNServer should work when it is called before the transport has
> been set up. Have you tried calling this during initialization?
>
> Regards,
> Ilian
>
> H.Kropf wrote:
>   
>> Hello
>>
>> I try to use OSS SoftPhone client (OSS 1.1.7 CVS) with server OpenSER 
>> and with NAT of type NAT::SIPSTUNClient::ConeNat
>> Calling of function SetSTUNServer connect succefully to my STUN-server.  
>> SoftPhone receive true NAT IP-address and NAT port.
>> But after, when i call SoftPhone::SendRegister() then  OpenSER register 
>> client like "NAT, without STUN"
>>
>> Exemples: (correct and wrong)
>> ===============================
>>
>> REGISTER requests via NAT using STUN:  (aliases: 111.111.111.111 - 
>> instead of local SIP-client IP, 222.222.222.222 - instead of NAT IP, 
>> 333.333.333.333 - instead of SIP-server IP)
>>
>> CORRECT (hardware Zyxel SIP-client)
>>   
>>     
> <snip>
>
>
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Re: [OpenSIPStack] STUN support bug?

From: Ilian J. C. P. <ip...@so...> - 2007-07-24 09:54:31

Hi,

SetSTUNServer should work when it is called before the transport has
been set up. Have you tried calling this during initialization?

Regards,
Ilian

H.Kropf wrote:
> Hello
>
> I try to use OSS SoftPhone client (OSS 1.1.7 CVS) with server OpenSER 
> and with NAT of type NAT::SIPSTUNClient::ConeNat
> Calling of function SetSTUNServer connect succefully to my STUN-server.  
> SoftPhone receive true NAT IP-address and NAT port.
> But after, when i call SoftPhone::SendRegister() then  OpenSER register 
> client like "NAT, without STUN"
>
> Exemples: (correct and wrong)
> ===============================
>
> REGISTER requests via NAT using STUN:  (aliases: 111.111.111.111 - 
> instead of local SIP-client IP, 222.222.222.222 - instead of NAT IP, 
> 333.333.333.333 - instead of SIP-server IP)
>
> CORRECT (hardware Zyxel SIP-client)
>   
<snip>

Re: [OpenSIPStack] OpenSBC behind NAT/router

From: Joegen E. B. <jb...@so...> - 2007-07-24 09:49:59

openSBc isn't really made for this setup.   OpenSBC is intended to be 
used either as a bridge between multi-homed networks or as a far end NAT 
traversal server.  These two setup requires that OpenSBC itself is not 
the subject of NAT. 

You may try "Static RTP Media Address" in the general parameters 
config.  Set the value to the public address of your NAT router.    As 
for the port number, OpenSBC starts at 30000 onwards for RTP and no, 
there isn't any provision to set the port range.  This however can 
easily be achieved with some code modification.  Let me know if you want 
to stretch it a bit further.

Re: [OpenSIPStack] OpenSBC behind NAT/router

From: Joegen E. B. <joe...@gm...> - 2007-07-24 09:31:27

openSBc isn't really made for this setup.   OpenSBC is intended to be 
used either as a bridge between multi-homed networks or as a far end NAT 
traversal server.  These two setup requires that OpenSBC itself is not 
the subject of NAT. 

You may try "Static RTP Media Address" in the general parameters 
config.  Set the value to the public address of your NAT router.    As 
for the port number, OpenSBC starts at 30000 onwards for RTP and no, 
there isn't any provision to set the port range.  This however can 
easily be achieved with some code modification.  Let me know if you want 
to stretch it a bit further.




Roland Auckenthaler wrote:
> Hi all,
>
> I have a bit of a tricky problem and am not sure if OpenSBC can handle this setup. The network config is as follows
>
> Office:                            Office                                       Home
> SBC (10.17.20.4) <------> Router/NAT <---public internet--> Router/NAT <------> SIPPhone
>
> Now the office NAT has port forwarding rules to send the SIP traffic to the SBC. The SIPPhone can send the invite to the SBC which internally forwards it to a voice service. The SBC sends back the 200 OK and ACK follows. So the SIP sginalling works, but the SDP offer/answer is not quite right.
>
> Obvioulsy the SDP in the invite has the wrong (sipphone-local) ip address. The SBc should be able to correct that. But the SDP answer from the SBC also shows it's internal IP address(10.17.20.4). 
>
> Now the question i have is if it is possible to configure the SBC to use the public ip address in the SDP. Furthermore, i'd like to know if it is possible to configure a port-range for the RTP traffic so that i can put some forwarding onto the NAT/router.
>
> A last question would be, whether i'm able to get this setup to work?
>
> does somebody have some experience or give some advice?
>
> cheers
>
> roland
>
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[OpenSIPStack] OpenSBC behind NAT/router

From: Roland A. <fo...@op...> - 2007-07-24 08:39:53

Hi all,

I have a bit of a tricky problem and am not sure if OpenSBC can handle this setup. The network config is as follows

Office:                            Office                                       Home
SBC (10.17.20.4) <------> Router/NAT <---public internet--> Router/NAT <------> SIPPhone

Now the office NAT has port forwarding rules to send the SIP traffic to the SBC. The SIPPhone can send the invite to the SBC which internally forwards it to a voice service. The SBC sends back the 200 OK and ACK follows. So the SIP sginalling works, but the SDP offer/answer is not quite right.

Obvioulsy the SDP in the invite has the wrong (sipphone-local) ip address. The SBc should be able to correct that. But the SDP answer from the SBC also shows it's internal IP address(10.17.20.4). 

Now the question i have is if it is possible to configure the SBC to use the public ip address in the SDP. Furthermore, i'd like to know if it is possible to configure a port-range for the RTP traffic so that i can put some forwarding onto the NAT/router.

A last question would be, whether i'm able to get this setup to work?

does somebody have some experience or give some advice?

cheers

roland

[OpenSIPStack] STUN support bug?

From: H.Kropf <mai...@gl...> - 2007-07-24 08:35:14

Hello

I try to use OSS SoftPhone client (OSS 1.1.7 CVS) with server OpenSER 
and with NAT of type NAT::SIPSTUNClient::ConeNat
Calling of function SetSTUNServer connect succefully to my STUN-server.  
SoftPhone receive true NAT IP-address and NAT port.
But after, when i call SoftPhone::SendRegister() then  OpenSER register 
client like "NAT, without STUN"

Exemples: (correct and wrong)
===============================

REGISTER requests via NAT using STUN:  (aliases: 111.111.111.111 - 
instead of local SIP-client IP, 222.222.222.222 - instead of NAT IP, 
333.333.333.333 - instead of SIP-server IP)

CORRECT (hardware Zyxel SIP-client)
************************************

REGISTER
  sip:mysrv.com:5060 SIP/2.0
  Via: SIP/2.0/UDP 222.222.222.222:30796;  branch=z9hG4bK332ccb85f16c5cbd
  Max-Forwards: 69
  To: <sip:10...@my...>
  From: <sip:10...@my...>;  tag=xIjMzUDMxID\015\012
  Call-ID: 1464D22DC9741212@222.222.222.222
  CSeq: 2 REGISTER
  Contact: <sip:100010@222.222.222.222:30796>
  Authorization:
    Digest username="zzz",
    realm="mysrv.com",
    nonce="46a5adc49a3608ebdb26cc2664302ba9a7cc47ca",
    uri="sip:mysrv.com:5060",
    response="c5b3f9967cb1ce70e847ce446b7705ec",
    cnonce="0a4f113b",
    qop="auth",
    nc=0000001
  Content-Length: 0
  Expires: 3600

REGISTER
sip:mysrv.com:5060 SIP/2.0
Via: SIP/2.0/UDP 222.222.222.222:30796;branch=z9hG4bK44e19947d5df8012
Max-Forwards: 69
To: <sip:10...@my...>
From: <sip:10...@my...>;tag=xIjMzUDMxID
Call-ID: 1464D22DC9741212@222.222.222.222
CSeq: 1 REGISTER
Contact: <sip:100010@222.222.222.222:30796>
Content-Length: 0
Expires: 3600


***********************
WRONG (OSS SIP-Client)
***********************


REGISTER
  sip:mysrv.com SIP/2.0
  From: 100008 <sip:10...@my...>;tag=396528833cf9181096d6882afa23793b
 To: 100008 <sip:10...@ip...>
Via: SIP/2.0/UDP 
222.222.222.222:31007;iid=3501;branch=z9hG4bK396528833cf9181096d7882afa23793b;uas-addr=333.333.333.333;rport
CSeq: 2 REGISTER
Call-ID: bcdbfa82-3cf9-1810-8d66-882afa23793b
Contact: "100008" <sip:100008@222.222.222.222:5060;transport=udp>    //  
?????????????????   5060?????????????????
User-Agent: OpenSIPStack-1.1.7-1
Expires: 3600
Max-Forwards:  9
Authorization:
  Digest username="olden1",
  realm="mysrv.com",
  nonce="46a5b1b01f9b24a48429cf395d82288132141766",
  uri="sip:mysrv.com",
  response="d48c234555217113d91ec016c92bd612",
  algorithm=MD5
 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
 Content-Length: 0


REGISTER
  sip:mysrv.com SIP/2.0
  From: 100008 <sip:10...@my...>;tag=bcdbfa823cf9181096d5882afa23793b
  To: 100008 <sip:10...@my...>
  Via: SIP/2.0/UDP 
222.222.222.222:31007;iid=3501;branch=z9hG4bKcf4c28833cf9181096d6882afa23793b;uas-addr=333.333.333.333;rport
  CSeq: 2 REGISTER
  Call-ID: bcdbfa82-3cf9-1810-8d66-882afa23793b
  Contact: *
  User-Agent: OpenSIPStack-1.1.7-1
  Expires: 0
  Max-Forwards:  9
  Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
  Content-Length: 0


=========================================================================
 Why Contact  == 100008@222.222.222.222:5060 instead of  
100008@222.222.222.222:31007 ????
and why Contact == *  ?

Re: [OpenSIPStack] Silence Detection

From: Ilian J. C. P. <ip...@so...> - 2007-07-23 10:46:05

Whit Thiele wrote:
> Hey Ilian,
>   
Hi Whit,
> Thanks for exposing these methods in ATLSIP!
>
> Has there been any more work on the adaptive Silence Detection? The
> threshold of 3 is pretty stable, but I have end-users who complain that all
> calls are extremely 'one-sided' meaning that once one side of the call
> begins talking, the other person can't easily interrupt them. Logically that
> makes sense, but the end users I have love to interrupt each other so its
> rather annoying for them to wait till the other person is finished. 
>   
I'm sorry I haven't had time to dig into the adaptive silence detection 
bug further. I'm currently busy with some other aspect of the softphone. 
But I promise I'll look into this closely in the near future.

Btw last time I checked, OpalSilenceDetector::ReceivedPacket(..) was the 
culprit.  You might want to take a look in that function if you have time.

Regards,
Ilian
> Has anyone else deployed ATLSIP to real world end users?
>
> If so, what has your experience been like? I'd love to see what kind of
> settings (Silence, CN etc) others are using in the field.
>
>
> Regards,
> Whit
>
>
>
>
>
> -----Original Message-----
> From: ope...@li...
> [mailto:ope...@li...] On Behalf Of Ilian
> Jeri C. Pinzon
> Sent: Friday, June 29, 2007 6:58 AM
> To: ope...@li...
> Subject: Re: [OpenSIPStack] Problem with OnOutgoingCallConnected
>
> The attachment was not sent. Probably filtered by the server. Anyway, here
> it is:
>
> ===========================================================================
>
> Hi all,
>
> I have exposed the setting of silence detection mode and audio jitter 
> delay in ATLSIP and SoftPhoneInterface.
>
> Here are the methods:
>
> DisableSilenceDetection()
> - Disables silence detection. Disables CNG as well.
>
> EnableFixedSilenceDetection( ULONG threshold )
> - Enables fixed silence detection. Any sound level below the threshold 
> is treated as silence (and CN is generated as a result). Don't use too 
> high threshold values or you'll only hear comfort noise. Try threshold=3 
> as suggested by Whit in another thread.
>
> EnableFixedSilenceDetectionEx( ULONG threshold , ULONG signalDeadband, 
> ULONG silenceDeadband  )
> - An extended version of the previous method. Don't tinker with this 
> unless you know what you're doing. For reference on how signalDeadband 
> and silenceDeadband are used, look in OpalSilenceDetector::ReceivedPacket().
>
> EnableAdaptiveSilenceDetection( ULONG adaptivePeriod )
> - Enables an adaptive silence detection. Supposedly this enables the 
> threshold to *adapt* to the current sound level every adaptivePeriod 
> milliseconds. However, its silence detection doesn't seem to be very 
> effective (at least in my machine). I'll look into this further to see 
> what's wrong. This mode with adaptivePeriod=4800 is the default mode for 
> ATLSIP.
>
> EnableAdaptiveSilenceDetectionEx( ULONG adaptivePeriod, ULONG 
> signalDeadband, ULONG silenceDeadband  )
> - An extended version of the previous method. Don't tinker with this 
> unless you know what you're doing. For reference on how signalDeadband 
> and silenceDeadband are used, look in OpalSilenceDetector::ReceivedPacket().
>
> SetAudioJitterDelay( ULONG minDelay, ULONG maxDelay )
> - Sets audio jitter delay settings.
>
>
> Regards,
> Ilian
>
>
>
> Ilian Jeri C. Pinzon wrote:
>   
>> ATLSIP's default silence detection is still faulty. You can try 
>> setting the silence detection to fixed mode. Somebody said in another 
>> thread that a threshold of 3 will do the trick. You may have to tweak 
>> this value though.
>>
>> Attached is the thread containing the functions to call for setting 
>> the silence detection mode.
>>
>> - Ilian
>>
>> tomach wrote:
>>     
>>> Hello!
>>>
>>> Ok thank you very much! Till now everythign works correct :) Now some 
>>> more advanced test...about quality of voice etc...
>>>
>>> It started to work correctly (From Tag was ok) after I build it as 
>>> debug and reregistered again :) Now everythign works correct:)
>>>
>>> I still have like last question, do you have comfort noise generator? 
>>> Because now its totally silence if somoebdy do not talk at all... and 
>>> it could be uncomcofortable for subscirbers...
>>>
>>> Best Regards,
>>> Tom
>>>
>>>       
>
>
>
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Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

From: Ilian J. C. P. <ip...@so...> - 2007-07-23 05:18:44

Hi Tom,

The logs you redirected to syslog actually came from OpenSIPStack using 
the PTRACE and LOG macros. ATLSIP is merely a wrapper for OpenSIPStack 
after all.

Regards,
Ilian

tomach wrote:
> Thanks Ilian!!
>
> Everythign works correct with syslog, but what about opnesipstack? is it also possisble to direct it to syslog or only ATLSIP can be directed to syslog?
>
> best regards!
>
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Re: [OpenSIPStack] RegSrv32 Error

From: Joegen E. B. <joe...@gm...> - 2007-07-22 14:45:10

It seems you have compiled a Debug version of the control.  MSVCP80D.DLL 
MSVCR80D.DLL are debug versions of the C++ runtime.   If my memory 
serves me well, you will not be able to link with these libraries 
statically in Debug mode.   Make sure you are compiling the Release 
build and that you link statically to the runtime using the "Multi 
threaded" mode in Code Generation Tab of you Project Property sheet.  If 
you still encounter errors, then you need to create an installer for 
your applciaitn and make sure that the depedencies are included with the 
installer package.

Joegen

[OpenSIPStack] Commercial implementations

From: Joegen E. B. <joe...@gm...> - 2007-07-22 14:39:48

Hi Everyone,

I am currently in the process of revamping the OpenSIPStack website.  A 
planning dediate a new page for commercial products using OpenSIPStack, 
ATLSIP and OpenSBC.   If you have such product, please send me an email 
at jo...@op....

Thanks!

Joegen

Re: [OpenSIPStack] RegSrv32 Error

From: Yacine A. <yac...@ms...> - 2007-07-22 11:52:26

Thanks for the reply, i just used depends.exe to determine the missing dlls=
 and i found them:MSVCP80D.DLLMSVCR80D.DLLDWMAPI.DLLBut why i'm getting dep=
endencies with internal Visual studio dlls.Thanks> Date: Sun, 22 Jul 2007 0=
4:16:24 -0700> From: eds...@ya...> To: opensipstack-devel@lists=
.sourceforge.net> Subject: Re: [OpenSIPStack] RegSrv32 Error> > Yacine,>   =
 >   use depends.exe to determine what dll's are missing when registering A=
TLSIP.>   > Yacine Auczone <yac...@ms...> wrote:>   Hello,I have downlo=
aded the latest Atlsip from cvs, the compilation was ok but whene i try to =
load the ATLSIP.dll activex i got this error:LoadLibrary("c:\atlsip\ATLSIP.=
dll") failed - This> application has failed to start because the applicatio=
n configuration> is incorrect. Reinstalling the application may fix this pr=
oblem.I was able to load the dll on the same compilation machine but never =
on others.> _______________________________________________________________=
__> Le blog Messenger de Michel, candidat de la Nouvelle Star : analyse, ne=
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Re: [OpenSIPStack] RegSrv32 Error

From: Andre S. <eds...@ya...> - 2007-07-22 11:23:43

Yacine,
   
  use depends.exe to determine what dll's are missing when registering ATLSIP.
  
Yacine Auczone <yac...@ms...> wrote:
  Hello,I have downloaded the latest Atlsip from cvs, the compilation was ok but whene i try to load the ATLSIP.dll activex i got this error:LoadLibrary("c:\atlsip\ATLSIP.dll") failed - This
application has failed to start because the application configuration
is incorrect. Reinstalling the application may fix this problem.I was able to load the dll on the same compilation machine but never on others.
_________________________________________________________________
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[OpenSIPStack] RegSrv32 Error

From: Yacine A. <yac...@ms...> - 2007-07-22 11:11:26

Hello,I have downloaded the latest Atlsip from cvs, the compilation was ok =
but whene i try to load the ATLSIP.dll activex i got this error:LoadLibrary=
("c:\atlsip\ATLSIP.dll") failed - This=0A=
application has failed to start because the application configuration=0A=
is incorrect. Reinstalling the application may fix this problem.I was able =
to load the dll on the same compilation machine but never on others.
_________________________________________________________________
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coulisses=85 A d=E9couvrir !
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[OpenSIPStack] Silence Detection

From: Whit T. <de...@wh...> - 2007-07-21 18:55:22

Hey Ilian,

Thanks for exposing these methods in ATLSIP!

Has there been any more work on the adaptive Silence Detection? The
threshold of 3 is pretty stable, but I have end-users who complain that all
calls are extremely 'one-sided' meaning that once one side of the call
begins talking, the other person can't easily interrupt them. Logically that
makes sense, but the end users I have love to interrupt each other so its
rather annoying for them to wait till the other person is finished. 

Has anyone else deployed ATLSIP to real world end users?

If so, what has your experience been like? I'd love to see what kind of
settings (Silence, CN etc) others are using in the field.


Regards,
Whit





-----Original Message-----
From: ope...@li...
[mailto:ope...@li...] On Behalf Of Ilian
Jeri C. Pinzon
Sent: Friday, June 29, 2007 6:58 AM
To: ope...@li...
Subject: Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

The attachment was not sent. Probably filtered by the server. Anyway, here
it is:

===========================================================================

Hi all,

I have exposed the setting of silence detection mode and audio jitter 
delay in ATLSIP and SoftPhoneInterface.

Here are the methods:

DisableSilenceDetection()
- Disables silence detection. Disables CNG as well.

EnableFixedSilenceDetection( ULONG threshold )
- Enables fixed silence detection. Any sound level below the threshold 
is treated as silence (and CN is generated as a result). Don't use too 
high threshold values or you'll only hear comfort noise. Try threshold=3 
as suggested by Whit in another thread.

EnableFixedSilenceDetectionEx( ULONG threshold , ULONG signalDeadband, 
ULONG silenceDeadband  )
- An extended version of the previous method. Don't tinker with this 
unless you know what you're doing. For reference on how signalDeadband 
and silenceDeadband are used, look in OpalSilenceDetector::ReceivedPacket().

EnableAdaptiveSilenceDetection( ULONG adaptivePeriod )
- Enables an adaptive silence detection. Supposedly this enables the 
threshold to *adapt* to the current sound level every adaptivePeriod 
milliseconds. However, its silence detection doesn't seem to be very 
effective (at least in my machine). I'll look into this further to see 
what's wrong. This mode with adaptivePeriod=4800 is the default mode for 
ATLSIP.

EnableAdaptiveSilenceDetectionEx( ULONG adaptivePeriod, ULONG 
signalDeadband, ULONG silenceDeadband  )
- An extended version of the previous method. Don't tinker with this 
unless you know what you're doing. For reference on how signalDeadband 
and silenceDeadband are used, look in OpalSilenceDetector::ReceivedPacket().

SetAudioJitterDelay( ULONG minDelay, ULONG maxDelay )
- Sets audio jitter delay settings.


Regards,
Ilian



Ilian Jeri C. Pinzon wrote:
> ATLSIP's default silence detection is still faulty. You can try 
> setting the silence detection to fixed mode. Somebody said in another 
> thread that a threshold of 3 will do the trick. You may have to tweak 
> this value though.
>
> Attached is the thread containing the functions to call for setting 
> the silence detection mode.
>
> - Ilian
>
> tomach wrote:
>> Hello!
>>
>> Ok thank you very much! Till now everythign works correct :) Now some 
>> more advanced test...about quality of voice etc...
>>
>> It started to work correctly (From Tag was ok) after I build it as 
>> debug and reregistered again :) Now everythign works correct:)
>>
>> I still have like last question, do you have comfort noise generator? 
>> Because now its totally silence if somoebdy do not talk at all... and 
>> it could be uncomcofortable for subscirbers...
>>
>> Best Regards,
>> Tom
>>

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

From: tomach <to...@dg...> - 2007-07-19 12:50:34

Thanks Ilian!!

Everythign works correct with syslog, but what about opnesipstack? is it also possisble to direct it to syslog or only ATLSIP can be directed to syslog?

best regards!

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

From: Joegen E. B. <joe...@gm...> - 2007-07-19 11:58:06

I believe this behavior is just appropriate.  There might be other 
applications that might want to known if multiple provisional responses 
are received.   If you don't need the event, then just ignore it on your 
application.

Also, the subject in this thread is no longer the subject of this mail.  
Please change your subject to be more descriptive of the post content.

tomach wrote:
> Hello Ilian!
>
> I have found next problem. 
> Event ringing in ATLsip comes to often. It sends its twice during connection:
> 1.when sipserver sends message "180 Rinnging", atlsip fires event ringing,
> 2. when sipserver sends message "183 sessoin progress", atlsip firesevent ringing agaiin...
>
> I belive that it shoudl fire ringing event only once??
>
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Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

From: tomach <to...@dg...> - 2007-07-19 11:05:50

Hello Ilian!

I have found next problem. 
Event ringing in ATLsip comes to often. It sends its twice during connection:
1.when sipserver sends message "180 Rinnging", atlsip fires event ringing,
2. when sipserver sends message "183 sessoin progress", atlsip firesevent ringing agaiin...

I belive that it shoudl fire ringing event only once??

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

From: Ilian J. C. P. <ip...@so...> - 2007-07-17 08:17:59

Hi,

tomach wrote:
> Or how to send logs to syslog? Have You consider such a solution? 
For syslog you can do this:

    Logger::SetDefaultLevel( 5 );

    static unsigned int options =
      LogDateAndTime |
      LogThread |
      LogAppendToFile;

    Logger::SetDefaultLogStream( new LoggingSystemLogStream(
      "ATLSIP", PIPSocket::Address("127.0.0.1"), 
PIPSocket::Address("127.0.0.1"), 514, 0, options ) );
> How do You cope with big log files finallly if You do not send them to syslog? I belive that after several days of working logs files can be really huge?
>   
Typically *for* ATLSIP, we really not need to store all the logs. If a 
bug/problem is found, you would want the logs only for that scenario. So 
storing the logs per application runtime shouldn't hurt. Of course, 
there is no problem if you prefer to keep logs for a longer period. 
You'll just have to implement your own housekeeping.

Regards,
Ilian

> -------------------------------------------------------------------------
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Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

From: tomach <to...@dg...> - 2007-07-17 07:31:30

Or how to send logs to syslog? Have You consider such a solution? How do You cope with big log files finallly if You do not send them to syslog? I belive that after several days of working logs files can be really huge?

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

From: tomach <to...@dg...> - 2007-07-17 07:06:56

Thanks a lot. To be honest I am not very good in C++ and COM objects...I hoped more that You tell me some easy way how to send all strings logs as an event to ATLSIP, that in C# I could catch it(log) and send it to my own log system....Because all my log systems are in C#...

I mean I would like to have a log system similar to sipMessage event is it possible with not small pice of work and it will not influence on stableness?

Re: [OpenSIPStack] OpenSBC Version 1.1.4

From: Joegen E. B. <joe...@gm...> - 2007-07-17 01:46:13

I don't see any reason why it wouldn't. 


Andre Silo wrote:
> Will my curent ATLSIP work with this new version?
>
> */"Joegen E. Baclor" <joe...@gm...>/* wrote:
>
>     Hi Everyone,
>
>     First, I apologize for the volume of unanswered e-mails in the
>     mailing
>     list. I have been busy with the July 15th Russell release.
>
>     Version 1.1.4 of OpenSBC is now available in CVS. This version is
>     going
>     to be tagged as the "Russell" release within the next few days as
>     soon
>     as our internal Q.A. has given the go signal. For the meantime,
>     the CVS
>     is again open for minor bug fixes. Feel free to test the new version
>     and notify me rightaway if you encounter any trouble.
>
>     Among the things you would notice in this release is the use of a new
>     STL based String class which is 5 times faster than the PWLIB PString
>     class. We have seen that we have regained 16% of CPU load simply
>     because of this change. You will see more of the container classes
>     being migrated to their STL couter-parts in the future. There is also
>     a new section in the HTTP admin that displays numerous resource
>     counters such as volume of transactions, calls, connections,
>     registrations etc.
>
>     The download section in opensipstack will be updated within the week.
>     We also plan to reactivate the source forge download pages. User
>     documentations will be published together with the downloads.
>
>     Cheers!
>
>     Joegen
>
>     -------------------------------------------------------------------------
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Re: [OpenSIPStack] OpenSIPStack: Problem

From: Joegen E. B. <joe...@gm...> - 2007-07-17 01:45:32

 

OpenSBC - Session Border Controller based on OpenSIPStack
ATLSIP - ActiveX Softphone Interface
OSPhone - Softphone reference implementation

Please see http://www.opensipstack.org website for details how to obtain 
the source code.  Your best bet currently is to download it from CVS.

Joegen


Arif V. Kadiwal wrote:
> Hello,
>
> I am wondering after the compilation of Opensipstack solution what i
> can do further, your documentation didn't specify how to run this
> solution(Opensipstack) after build and compilation. I need to know how
> to run the opensipstack and I want to see if you have any running
> application based on opensipstack or developed with opensipstack.
>
> Again need to know how to run the solution OpenSIPstack after
> compilation and build and also want to look at any application based
> on OpenSIPStack.
>
>
> Arif
> Research Assistant
> Electrical and Computer Engineering
> Concordia University
> Montreal, Canada
>
>
> On 7/8/07, Joegen E. Baclor <joe...@gm...> wrote:
>> */Windows.h missing/* is a complile error commonly reported in the
>> mailing list. This may happen in Visual Stdio 2005 Express edition of
>> visual C++. Windows.h goes with the Platform SDK and is available via a
>> separate installation here
>> http://msdn.microsoft.com/vstudio/express/visualc/usingpsdk/.
>>
>> Arif V. Kadiwal wrote:
>> > Dear Developers/Staff,
>> >
>> > I am using OpenSIPStack 1.1.5, according to your web documentation
>> > under section Compiling In Windows I was following steps to compile
>> > OpenSIPStack solution in VS 8. During the compilation time it give me
>> > error in file contain.h saying it is unable to find file windows.h the
>> > detail error is given below I got lots of error with this file
>> > (windows.h is not found)
>> >
>> > c:\documents and settings\student6\desktop\open source
>> > 
>> sipstack\opensipstack-1.1.5\opensipstack-1.1.5-rc1\include\ptlib\msos/ptlib/contain.h(290) 
>>
>> > : fatal error C1083: Cannot open include file: 'windows.h': No such
>> > file or directory
>> > pxmlrpc.cxx
>> >
>> >
>> > from the above it seems there is no windows.h file in OpenSIPStakc
>> > folder hierarchy and I also looked thoroughly and i didnt find
>> > windows.h file in OpenSIPStack folder tree. Can please help me to
>> > rectify this problem. I would appreciate your quick soon in this
>> > regard.
>> >
>> >
>> > Arif
>> >
>> > Research Assistant
>> > Electrical and Computer Engineering
>> > Concordia Unviersity
>> > Montreal, Canada
>> > www.concordia.ca
>> >
>> > 
>> ------------------------------------------------------------------------- 
>>
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Re: [OpenSIPStack] FW: FW: OpenSBC as Forking Proxy

From: Joegen E. B. <joe...@gm...> - 2007-07-17 01:42:12

Hi Gustavo,

Can send a diff versus current CVS HEAD? 

Joegen


Gustavo Curetti wrote:
> Hi Joegen:
>  
> I made changes in the code to avoid that the invite continue being 
> sent when sesion is being cancelled.
> When a cancel is received, after cancelling the Invite Server 
> Transaction, I terminate the Invite Client Transaction:
>  
> /BOOL ProxySession::ValidateRequest(SIPMessage & request)/
> /{/
> /..../
> /  }else if( request.IsCancel() )
>   {
>     
> m_SessionManager.GetUserAgent().GetStack().CancelInviteServerTransaction(m_Request);
>     
> m_SessionManager.GetUserAgent().GetStack().TerminateInviteClientTransaction(m_CurrentUACRequest);
>   }/
> /}/
> // 
> /BOOL SIPTransactionManager::TerminateInviteClientTransaction(const 
> SIPMessage & invite)
> {
>   if( m_IsTerminating )
>     return FALSE;/
> // 
> /  PAssert( invite.IsInvite(), PLogicError );/
> // 
> /  PWaitAndSignal lock( m_TransactionPoolMutex );/
> // 
> /  TransactionId id = invite.GetTransactionId();/
> // 
> /  if( id.AsString().IsEmpty() )
>   {
>     LOG_IF_DEBUG( LogInfo(), "TerminateInviteClientTransaction() - 
> Unable to match transaction because transaction-id is empty" );
>     return FALSE;
>   }/
> // 
> /  id.SetStateMachine( "ICT" );/
> // 
> /  SIPTransaction * transaction = m_TransactionPool.GetAt( 
> id.AsString() );
>   if( transaction == NULL )
>   {
>     return FALSE;
>   }
>  
>   transaction->Terminate();/
> // 
> /  return TRUE;
> }/
>
>  
> And in ProxySession::OnDNSFailOver(), the m_CurrentUACRequest is 
> actualized:
>  
> /void ProxySession::OnDNSFailOver(SIPDNSFailOver & failOverEvent)
> {
>  SIPMessage msg = failOverEvent.GetMessage();/
> // 
> / if( msg.IsInvite() )
>  {
>   m_CurrentUACRequest = msg.AsString();
>   SIPURI targetURI;
>   msg.GetRequestURI(targetURI);
>   SetDialogPeerAddress( targetURI.AsString() );
>  }
> }/
>
> First, I tried:
>
> /m_CurrentUACRequest = msg;/
>
> But the m_CurrentUACRequest was not actualized, because the
> m_InternalId was the same for both messages and:
>  
> /SIPMessage & SIPMessage:://operator=(const// SIPMessage & msg)/
> /{/
> / if//( msg.m_InternalId == m_InternalId )/
> /  return *this;/
> /.../
> /}/
>  
> Then i used the AsString():
>  
> /m_CurrentUACRequest = msg.AsString();/
>  
> I know this is not the best solution, then if you could suggest a 
> better one please let me know. Maybe in
> SIPTransaction::DoDNSFailover(), changing :
>  
> /SIPMessage request = m_OpeningRequest;/
>  
> I attach the sip flows before and after the changes in the code, the 
> second sip flow is more normal.
>  
> Thanks for your help.
>  
> Gustavo
>  
>  
> ------------------------------------------------------------------------
>
> > Date: Wed, 4 Jul 2007 15:06:09 +0800
> > To: cur...@gm...
> > From: joe...@gm...
> > CC: ope...@li...
> > Subject: Re: [OpenSIPStack] FW: FW: OpenSBC as Forking Proxy
> >
> > Gustavo Curetti wrote:
> > > Hi Joegen:
> > >
> > > I don't have a call sesion, i have a proxy session. The cancel is 
> sent
> > > to the m_DialogPeerAddress for a proxy session. I have added the
> > > following code:
> > >
> > > void ProxySession::OnDNSFailOver(
> > > SIPDNSFailOver & failOverEvent
> > > )
> > > {
> > > SIPMessage msg = failOverEvent.GetMessage();
> > > if( msg.IsInvite() )
> > > {
> > > SIPURI targetURI;
> > > msg.GetRequestURI(targetURI);
> > > SetDialogPeerAddress( targetURI.AsString() );
> > > }
> > > }
> >
> > I have patched CVS accordingly.
> >
> >
> > >
> > > Then the cancel is sent to the actual target of the ICT transaction,
> > > but if the actual target is not answering, the ICT continue trying
> > > with the actual target and with the followings. The ICT transaction
> > > must stop trying. Could you give some directions, please?
> >
> > I think this is a compliant behavior. Don't you want the UA to know
> > that the UAS is actually down? I have checked 3261 again and paragraph
> > indicates that a proxy MAY send a response to the CANCEL and generate a
> > new transaction towards the UAS. This is not a MUST and is left to the
> > implementors to decide on. I am open to discuss this further if you
> > think it is a better approach to handle extra state in the proxy for
> > cancelling transaction. My main reason for doing this is to make proxy
> > transactions light and I would favor to let it stay that way.
> >
> > 9.2 Server Behavior
> >
> > The CANCEL method requests that the TU at the server side cancel a
> > pending transaction. The TU determines the transaction to be
> > cancelled by taking the CANCEL request, and then assuming that the
> > request method is anything but CANCEL or ACK and applying the
> > transaction matching procedures of Section 17.2.3. The matching
> > transaction is the one to be cancelled.
> >
> > The processing of a CANCEL request at a server depends on the type of
> > server. A stateless proxy will forward it, a stateful proxy might
> > respond to it and generate some CANCEL requests of its own, and a UAS
> > will respond to it. See Section 16.10 for proxy treatment of CANCEL.
> >
> >
> >
> >
> > >
> > > Thanks for your help.
> > > Gustavo.
> > >
> > > 
> ------------------------------------------------------------------------
> > >
> > > > Date: Tue, 26 Jun 2007 15:23:39 +0800
> > > > From: jb...@so...
> > > > To: jb...@so...; 
> ope...@li...
> > > > CC: cur...@gm...
> > > > Subject: Re: [OpenSIPStack] FW: FW: OpenSBC as Forking Proxy
> > > >
> > > > Hi Gustavo,
> > > >
> > > > I have committed a fix of this. It's very simple actually. I just
> > > > propagated a SIPDNSFailOver stack event to the call session and made
> > > > sure that m_CurrentUACInvite is replaced. Can you check if this is a
> > > > safe change? Thanks.
> > > >
> > > > Joegen
> > > >
> > > >
> > > > Joegen E. Baclor wrote:
> > > > > Hmmmn thats TRUE. Lemme dig further. I'll let you know once a 
> fix is
> > > > > available.
> > > > >
> > > > >
> > > > > Gustavo Curetti wrote:
> > > > >
> > > > >> Joegen:
> > > > >>
> > > > >> The fix works fine. But i have the following problem: When i 
> hang up
> > > > >> and send a Cancel to OpenSBC, the OpenSBC respond with
> > > > >> Code487_RequestCancelled but the Cancel is not been send to the
> > > actual
> > > > >> target, the Cancel is just routed again.
> > > > >>
> > > > >> Thanks for your help.
> > > > >>
> > > > >> Gustavo
> > > > >>
> > > > >>
> > > 
> ------------------------------------------------------------------------
> > > > >>
> > > > >>
> > > > >>> Date: Wed, 13 Jun 2007 22:15:14 +0800
> > > > >>> To: cur...@gm...;
> > > ope...@li...
> > > > >>> Subject: Re: [OpenSIPStack] FW: FW: OpenSBC as Forking Proxy
> > > > >>> From: joe...@gm...
> > > > >>>
> > > > >>> I've just checked in a fix for this in CVS.
> > > > >>>
> > > > >>> Gustavo Curetti wrote:
> > > > >>>
> > > > >>>> Joegen,
> > > > >>>>
> > > > >>>> I tried the DoDNSFailover's code but i have the following 
> behavior:
> > > > >>>>
> > > > >>>> When DoDNSFailover() is called for the first time
> > > m_FailOverAttempts
> > > > >>>> is increased from 0 to 1.
> > > > >>>>
> > > > >>>> Then the via is changed (+ "-") and with this change when
> > > > >>>> FindTransactionAndAddEvent() is called a new transaction is
> > > created.
> > > > >>>> In SIPTransaction::SIPTransaction() m_FailOverAttempts is
> > > initialized
> > > > >>>> to 0. Then when DoDNSFailover() is called again, the same 
> target
> > > > >>>>
> > > > >> is used.
> > > > >>
> > > > >>>> Gustavo
> > > > >>>>
> > > > >>>>
> > > > >>>>
> > > > >>
> > > 
> ------------------------------------------------------------------------
> > > > >>
> > > > >>>> From: cur...@ho...
> > > > >>>> To: jb...@so...
> > > > >>>> Subject: RE: [OpenSIPStack] FW: OpenSBC as Forking Proxy
> > > > >>>> Date: Wed, 30 May 2007 14:43:32 +0200
> > > > >>>>
> > > > >>>> Joegen,
> > > > >>>>
> > > > >>>> Thanks a lot. I will try to change the behavior.
> > > > >>>>
> > > > >>>> Gustavo
> > > > >>>>
> > > > >>>>
> > > > >>>>
> > > > >>>>
> > > > >>
> > > 
> ------------------------------------------------------------------------
> > > > >>
> > > > >>>>> Date: Fri, 18 May 2007 08:15:20 +0800
> > > > >>>>> From: jb...@so...
> > > > >>>>> To: cur...@gm...;
> > > > >>>>>
> > > > >>>> ope...@li...
> > > > >>>>
> > > > >>>>> Subject: Re: [OpenSIPStack] FW: OpenSBC as Forking Proxy
> > > > >>>>>
> > > > >>>>> Gustavo,
> > > > >>>>>
> > > > >>>>> Sorry, I forgot to get back to you. If you have the latest 
> CVS
> > > head
> > > > >>>>> code, check out BOOL SIPTransaction::DoDNSFailover() in
> > > > >>>>> SIPTransaction.cxx. I have committed this a few days ago to
> > > > >>>>> demonstrate fail-over by forking using DNS/SRV records. New
> > > > >>>>> transactions are created by calling
> > > > >>>>>
> > > > >>>> FindTransactionAndAddEvent(). You
> > > > >>>>
> > > > >>>>> can just change its behavior a bit and get the fail-over 
> routes
> > > > >>>>> somewhere instead of DNS/SRV queries.
> > > > >>>>>
> > > > >>>>> Joegen
> > > > >>>>>
> > > > >>>>>
> > > > >>>>> Gustavo Curetti wrote:
> > > > >>>>>
> > > > >>>>>> Joegen:
> > > > >>>>>>
> > > > >>>>>> I don't understand how to create a new client transaction 
> when
> > > > >>>>>>
> > > > >>>> the
> > > > >>>>
> > > > >>>>>> first invite fail in the FSM layer. Could you give some
> > > > >>>>>>
> > > > >>>> directions,
> > > > >>>>
> > > > >>>>>> please?
> > > > >>>>>>
> > > > >>>>>> Thanks for your help.
> > > > >>>>>>
> > > > >>>>>> Gustavo.
> > > > >>>>>>
> > > > >>>>>>
> > > > >>>>>>
> > > > >>>>>>
> > > > >>>>>>
> > > > >>
> > > 
> ------------------------------------------------------------------------
> > > > >>
> > > > >>>>>> From: cur...@ho...
> > > > >>>>>> To: jb...@so...
> > > > >>>>>> Subject: RE: [OpenSIPStack] OpenSBC as Forking Proxy
> > > > >>>>>> Date: Mon, 14 May 2007 16:18:57 +0200
> > > > >>>>>>
> > > > >>>>>> Joegen:
> > > > >>>>>>
> > > > >>>>>> Thanks for your help. Do you suggest to do the serial 
> forking in
> > > > >>>>>> FSM layer with a custom header?.Must each new try create 
> a new
> > > > >>>>>> client transaction?
> > > > >>>>>>
> > > > >>>>>> Thanks
> > > > >>>>>> Gustavo
> > > > >>>>>>
> > > > >>>>>>
> > > > >>>>>>
> > > > >>
> > > 
> ------------------------------------------------------------------------
> > > > >>
> > > > >>>>>>> Date: Thu, 10 May 2007 13:28:29 +0800
> > > > >>>>>>> From: jb...@so...
> > > > >>>>>>> To: cur...@gm...
> > > > >>>>>>> Subject: Re: [OpenSIPStack] OpenSBC as Forking Proxy
> > > > >>>>>>>
> > > > >>>>>>> Gustavo,
> > > > >>>>>>>
> > > > >>>>>>> This will be a bit tricky. It is not as simple as 
> spawning an
> > > > >>>>>>>
> > > > >>>>>> outbound
> > > > >>>>>>
> > > > >>>>>>> invite. There should be a clean mechanism to clone
> > > > >>>>>>>
> > > > >>>> transactions and
> > > > >>>>
> > > > >>>>>>> this is not present in the FSM currently. Forking should 
> be done
> > > > >>>>>>>
> > > > >>>>>> in the
> > > > >>>>>>
> > > > >>>>>>> FSM layer, not in the UACore layer. I will see what I 
> can do to
> > > > >>>>>>>
> > > > >>>>>> help
> > > > >>>>>>
> > > > >>>>>>> you. I will let you know when I have something you can use
> > > > >>>>>>>
> > > > >>>>>> cleanly to
> > > > >>>>>>
> > > > >>>>>>> fork your calls. Perhaps over the weekend, but that isn't a
> > > > >>>>>>>
> > > > >>>> promise.
> > > > >>>>
> > > > >>>>>>> Joegen
> > > > >>>>>>>
> > > > >>>>>>> Gustavo Curetti wrote:
> > > > >>>>>>>
> > > > >>>>>>>> Joegen:
> > > > >>>>>>>>
> > > > >>>>>>>> What I want to do is a very simple sequential search. 
> When one
> > > > >>>>>>>> destination don't answer or reject the call I want the 
> OpenSBC
> > > > >>>>>>>>
> > > > >>>>>> try
> > > > >>>>>>
> > > > >>>>>>>> another.
> > > > >>>>>>>>
> > > > >>>>>>>> I made the following changes in the code for timer B
> > > > >>>>>>>>
> > > > >>>>>> expiration just
> > > > >>>>>>
> > > > >>>>>>>> for do some tests:
> > > > >>>>>>>>
> > > > >>>>>>>> void ProxySessionManager::OnTimerExpire(
> > > > >>>>>>>> SIPTimerExpire & timerEvent,
> > > > >>>>>>>> SIPSession * session
> > > > >>>>>>>> )
> > > > >>>>>>>> {
> > > > >>>>>>>> if( session != NULL )
> > > > >>>>>>>> {
> > > > >>>>>>>> LOG_IF_DEBUG( LogWarning(), "*** TIMER EXPIRATION *** 
> for SIP
> > > > >>>>>>>> Session " << session->GetSessionId() );
> > > > >>>>>>>> if( timerEvent.GetTimer() ==
> > > > >>>>>>>>
> > > > >>>> SIPTransactions::SIPTimerEvent::B)
> > > > >>>>
> > > > >>>>>>>> {
> > > > >>>>>>>> SIPMessage msg = ((ProxySession
> > > > >>>>>>>>
> > > > >>>> *)session)->GetOriginalInvite();
> > > > >>>>
> > > > >>>>>>>> session->EnqueueSessionEvent( new SIPSessionEvent(
> > > > >>>>>>>>
> > > > >>>> *session, 1,
> > > > >>>>
> > > > >>>>>>>> msg ) );
> > > > >>>>>>>> }
> > > > >>>>>>>> session->OnTimerExpire( timerEvent );
> > > > >>>>>>>> }
> > > > >>>>>>>> }
> > > > >>>>>>>>
> > > > >>>>>>>> and
> > > > >>>>>>>>
> > > > >>>>>>>> void ProxySession::OnTimerExpire(
> > > > >>>>>>>> SIPTimerExpire & timerEvent
> > > > >>>>>>>> )
> > > > >>>>>>>> {
> > > > >>>>>>>> GCREF( "SIPSession::OnTimerExpire" );
> > > > >>>>>>>> if( timerEvent.GetTimer() ==
> > > > >>>>>>>>
> > > > >>>> SIPTransactions::SIPTimerEvent::B ||
> > > > >>>>
> > > > >>>>>>>> timerEvent.GetTimer() == 
> SIPTransactions::SIPTimerEvent::F )
> > > > >>>>>>>> {
> > > > >>>>>>>> ///this is an ICT timeout
> > > > >>>>>>>> SIPMessage timeout;
> > > > >>>>>>>> GetCurrentUASRequest().CreateResponse( timeout,
> > > > >>>>>>>> SIPMessage::Code480_TemporarilyNotAvailable );
> > > > >>>>>>>> SendRequest( timeout );
> > > > >>>>>>>> }
> > > > >>>>>>>>
> > > > >>>>>>>> //Destroy();
> > > > >>>>>>>> }
> > > > >>>>>>>>
> > > > >>>>>>>> With these changes and a relay route:
> > > > >>>>>>>>
> > > > >>>>>>>> [sip:*@192.168.0.207:*] sip:192.168.0.1:5060,
> > > > >>>>>>>>
> > > > >>>>>> sip:192.168.0.60:5060
> > > > >>>>>>
> > > > >>>>>>>> the OpenSBC made the second invite successfully. But what i
> > > > >>>>>>>>
> > > > >>>>>> really
> > > > >>>>>>
> > > > >>>>>>>> want is to use some custom headers with a list destination
> > > > >>>>>>>>
> > > > >>>>>> addresses
> > > > >>>>>>
> > > > >>>>>>>> instead of the relay routes and to do the same in case of a
> > > > >>>>>>>>
> > > > >>>>>> reject. Do
> > > > >>>>>>
> > > > >>>>>>>> you have any suggestions?
> > > > >>>>>>>>
> > > > >>>>>>>> Other question: Can i have two active ICT for a session?
> > > > >>>>>>>>
> > > > >>>>>> Because in
> > > > >>>>>>
> > > > >>>>>>>> the case of reject, I must start a new ICT for trying 
> the next
> > > > >>>>>>>> destination but canceling throw the first ICT at the same
> > > > >>>>>>>>
> > > > >>>> time.
> > > > >>>>
> > > > >>>>>>>> Thanks for your help.
> > > > >>>>>>>>
> > > > >>>>>>>> Gustavo
> > > > >>>>>>>>
> > > > >>>>>>>>
> > > > >>>>>>>>
> > > > >>
> > > 
> ------------------------------------------------------------------------
> > > > >>
> > > > >>>>>>>>> Date: Sat, 5 May 2007 13:50:44 +0800
> > > > >>>>>>>>> From: jb...@so...
> > > > >>>>>>>>> To: cur...@gm...;
> > > > >>>>>>>>>
> > > > >>>>>> ope...@li...
> > > > >>>>>>
> > > > >>>>>>>>> Subject: Re: [OpenSIPStack] OpenSBC as Forking Proxy
> > > > >>>>>>>>>
> > > > >>>>>>>>> Gustavo,
> > > > >>>>>>>>>
> > > > >>>>>>>>> Forking is not supported yet in OpenSBC.
> > > > >>>>>>>>>
> > > > >>>>>>>>> Joegen
> > > > >>>>>>>>>
> > > > >>>>>>>>> Gustavo Curetti wrote:
> > > > >>>>>>>>>
> > > > >>>>>>>>>> Hi Joegen:
> > > > >>>>>>>>>>
> > > > >>>>>>>>>> I want to use the OpenSBC as a Forking Proxy. I want
> > > > >>>>>>>>>>
> > > > >>>> that the
> > > > >>>>
> > > > >>>>>>>>>> OpenSBC try the differents Relays Routes one by one. 
> Could
> > > > >>>>>>>>>>
> > > > >>>>>> you give
> > > > >>>>>>
> > > > >>>>>>>>>> some directions, please?
> > > > >>>>>>>>>>
> > > > >>>>>>>>>> Thanks for your help.
> > > > >>>>>>>>>>
> > > > >>>>>>>>>> Gustavo.
> > > > >>>>>>>>>>
> > > > >>>>>>>>>>
> > > > >>>>>>>>>>
> > > > >>
> > > 
> ------------------------------------------------------------------------
> > > > >>
> > > > >>>>>>>>>> Descubre Live.com - tu propia página de inicio,
> > > > >>>>>>>>>>
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> > > > >>>>>>>>>>
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> > > > >>>>>>
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> > > > >>>>>>>>>>
> > > > >>
> > > 
> ------------------------------------------------------------------------
> > > > >>
> > > > >>>>>>>>>>
> > > > >>
> > > 
> -------------------------------------------------------------------------
> > > > >>
> > > > >>>>>>>>>> This SF.net email is sponsored by DB2 Express
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> > > > >>
> > > 
> ------------------------------------------------------------------------
> > > > >>
> > > > >>>>>>>>>> _______________________________________________
> > > > >>>>>>>>>> opensipstack-devel mailing list
> > > > >>>>>>>>>> ope...@li...
> > > > >>>>>>>>>>
> > > > >>>>>>>>>>
> > > > >>>>>> 
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> > > > >>>>>>
> > > > >>>>>>>>>
> > > > >>>>>>>>>
> > > > >>
> > > 
> -------------------------------------------------------------------------
> > > > >>
> > > > >>>>>>>>> This SF.net email is sponsored by DB2 Express
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> > > > >>>>>>>>>
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> > > > >>>>
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> > > > >>>>>>>>> _______________________________________________
> > > > >>>>>>>>> opensipstack-devel mailing list
> > > > >>>>>>>>> ope...@li...
> > > > >>>>>>>>>
> > > > >>>>>>>>>
> > > > >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> > > > >>>>
> > > > >>>>>>>>
> > > > >>>>>>>>
> > > > >>
> > > 
> ------------------------------------------------------------------------
> > > > >>
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> > > 
> ------------------------------------------------------------------------
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> ------------------------------------------------------------------------
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> > > > >>>>>> Se uno de los primeros en probar Windows Live Mail. Windows
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> > > > >>
> > > > >>>>>
> > > > >>>>>
> > > > >>
> > > 
> -------------------------------------------------------------------------
> > > > >>
> > > > >>>>> This SF.net email is sponsored by DB2 Express
> > > > >>>>> Download DB2 Express C - the FREE version of DB2 express 
> and take
> > > > >>>>> control of your XML. No limits. Just data. Click to get it 
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> > > > >>>>> http://sourceforge.net/powerbar/db2/
> > > > >>>>> _______________________________________________
> > > > >>>>> opensipstack-devel mailing list
> > > > >>>>> ope...@li...
> > > > >>>>> 
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> > > > >>>>>
> > > > >>>>
> > > > >>>>
> > > > >>
> > > 
> ------------------------------------------------------------------------
> > > > >>
> > > > >>>> Se uno de los primeros en probar Windows Live Mail. Windows 
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> > > > >>
> > > 
> ------------------------------------------------------------------------
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> > >
> > > > >>
> > > > >>
> > > > >>>>
> > > > >>
> > > 
> ------------------------------------------------------------------------
> > > > >>
> > > > >>>>
> > > > >>
> > > 
> -------------------------------------------------------------------------
> > > > >>
> > > > >>>> This SF.net email is sponsored by DB2 Express
> > > > >>>> Download DB2 Express C - the FREE version of DB2 express 
> and take
> > > > >>>> control of your XML. No limits. Just data. Click to get it now.
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> > > > >>>>
> > > > >>
> > > 
> ------------------------------------------------------------------------
> > > > >>
> > > > >>>> _______________________________________________
> > > > >>>> opensipstack-devel mailing list
> > > > >>>> ope...@li...
> > > > >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> > > > >>>>
> > > > >>>>
> > > > >>> --
> > > > >>> Joegen E. Baclor
> > > > >>> CTO - Solegy LLC
> > > > >>>
> > > > >>> Email: joegen @ solegy.com
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> > > 
> ------------------------------------------------------------------------
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> > > Mail.
> > > > >>
> > > 
> <http://ideas.live.com/programpage.aspx?versionId=5d21c51a-b161-4314-9b0e-4911fb2b2e6d>
> > > > >>
> > > > >
> > > > >
> > > > >
> > > > >
> > > 
> -------------------------------------------------------------------------
> > > > > This SF.net email is sponsored by DB2 Express
> > > > > Download DB2 Express C - the FREE version of DB2 express and take
> > > > > control of your XML. No limits. Just data. Click to get it now.
> > > > > http://sourceforge.net/powerbar/db2/
> > > > > _______________________________________________
> > > > > opensipstack-devel mailing list
> > > > > ope...@li...
> > > > > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> > > > >
> > > > >
> > > >
> > > >
> > > >
> > > 
> -------------------------------------------------------------------------
> > > > This SF.net email is sponsored by DB2 Express
> > > > Download DB2 Express C - the FREE version of DB2 express and take
> > > > control of your XML. No limits. Just data. Click to get it now.
> > > > http://sourceforge.net/powerbar/db2/
> > > > _______________________________________________
> > > > opensipstack-devel mailing list
> > > > ope...@li...
> > > > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> > >
> > >
> > > 
> ------------------------------------------------------------------------
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> >
> >
> > 
> -------------------------------------------------------------------------
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> > Download DB2 Express C - the FREE version of DB2 express and take
> > control of your XML. No limits. Just data. Click to get it now.
> > http://sourceforge.net/powerbar/db2/
> > _______________________________________________
> > opensipstack-devel mailing list
> > ope...@li...
> > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
> ------------------------------------------------------------------------
> Descubre Live.com - tu propia página de inicio, personalizada para ver 
> rápidamente todo lo que te interesa en un mismo sitio. todo en el 
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> ------------------------------------------------------------------------
>
> | 192.168.0.60      | 192.168.0.207     | 192.168.0.58      | 192.168.0.1       | 192.168.0.206     | 192.168.0.38      |
> |         INVITE    |                   |                   |                   |                   |                   |
> |         ------------------>           |                   |                   |                   |                   |
> |         407 Proxy authentication required                 |                   |                   |                   |
> |         <------------------           |                   |                   |                   |                   |
> |         ACK       |                   |                   |                   |                   |                   |
> |         ------------------>           |                   |                   |                   |                   |
> |         INVITE    |                   |                   |                   |                   |                   |
> |         ------------------>           |                   |                   |                   |                   |
> |                   |         INVITE    |                   |                   |                   |                   |
> |                   |         ------------------>           |                   |                   |                   |
> |                   |         100 Trying|                   |                   |                   |                   |
> |                   |         <------------------           |                   |                   |                   |
> |         100 Trying|                   |                   |                   |                   |                   |
> |         <------------------           |                   |                   |                   |                   |
> |                   |                   |         INVITE    |                   |                   |                   |
> |                   |                   |         ------------------>           |                   |                   |
> |                   |                   |         INVITE    |                   |                   |                   |
> |                   |                   |         ------------------>           |                   |                   |
> |                   |                   |         INVITE    |                   |                   |                   |
> |                   |                   |         ------------------>           |                   |                   |
> |                   |                   |         INVITE    |                   |                   |                   |
> |                   |                   |         ------------------>           |                   |                   |
> |                   |                   |         INVITE    |                   |                   |                   |
> |                   |                   |         ------------------>           |                   |                   |
> |                   |                   |         INVITE    |                   |                   |                   |
> |                   |                   |         -------------------------------------->           |                   |
> |                   |                   |         INVITE    |                   |                   |                   |
> |                   |                   |         -------------------------------------->           |                   |
> |                   |                   |         INVITE    |                   |                   |                   |
> |                   |                   |         -------------------------------------->           |                   |
> |                   |                   |         INVITE    |                   |                   |                   |
> |                   |                   |         -------------------------------------->           |                   |
> |         CANCEL    |                   |                   |                   |                   |                   |
> |         ------------------>           |                   |                   |                   |                   |
> |                   |         CANCEL    |                   |                   |                   |                   |
> |                   |         ------------------>           |                   |                   |                   |
> |                   |         487 Request Cancelled         |                   |                   |                   |
> |                   |         <------------------           |                   |                   |                   |
> |                   |                   |         CANCEL    |                   |                   |                   |
> |                   |                   |         -------------------------------------->           |                   |
> |         487 Request Cancelled         |                   |                   |                   |                   |
> |         <------------------           |                   |                   |                   |                   |
> |                   |                   |         CANCEL    |                   |                   |                   |
> |                   |                   |         -------------------------------------->           |                   |
> |         ACK       |                   |                   |                   |                   |                   |
> |         ------------------>           |                   |                   |                   |                   |
> |                   |         ACK       |                   |                   |                   |                   |
> |                   |         ------------------>           |                   |                   |                   |
> |                   |                   |         CANCEL    |                   |                   |                   |
> |                   |                   |         -------------------------------------->           |                   |
> |                   |                   |         CANCEL    |                   |                   |                   |
> |                   |                   |         -------------------------------------->           |                   |
> |                   |                   |         INVITE    |                   |                   |                   |
> |                   |                   |         -------------------------------------->           |                   |
> |                   |                   |         INVITE    |                   |                   |                   |
> |                   |                   |         ---------------------------------------------------------->           |
> |                   |                   |         INVITE    |                   |                   |                   |
> |                   |                   |         ---------------------------------------------------------->           |
> |                   |                   |         CANCEL    |                   |                   |                   |
> |                   |                   |         -------------------------------------->           |                   |
> |                   |                   |         INVITE    |                   |                   |                   |
> |                   |                   |         ---------------------------------------------------------->           |
> |                   |                   |         INVITE    |                   |                   |                   |
> |                   |                   |         ---------------------------------------------------------->           |
> |                   |                   |         CANCEL    |                   |                   |                   |
> |                   |                   |         -------------------------------------->           |                   |
> |                   |                   |         CANCEL    |                   |                   |                   |
> |                   |                   |         -------------------------------------->           |                   |
> |                   |                   |         CANCEL    |                   |                   |                   |
> |                   |                   |         -------------------------------------->           |                   |
> |                   |                   |         CANCEL    |                   |                   |                   |
> |                   |                   |         -------------------------------------->           |                   |
> |                   |                   |         INVITE    |                   |                   |                   |
> |                   |                   |         ---------------------------------------------------------->           |
> |         INVITE    |                   |                   |                   |                   |                   |
> |         <--------------------------------------           |                   |                   |                   |
> |         100 Trying|                   |                   |                   |                   |                   |
> |         -------------------------------------->           |                   |                   |                   |
> |         180 Ringing                   |                   |                   |                   |                   |
> |         -------------------------------------->           |                   |                   |                   |
> |                   |                   |         CANCEL    |                   |                   |                   |
> |                   |                   |         -------------------------------------->           |                   |
> |                   |                   |         CANCEL    |                   |                   |                   |
> |                   |                   |         ---------------------------------------------------------->           |
> |                   |                   |         CANCEL    |                   |                   |                   |
> |                   |                   |         ---------------------------------------------------------->           |
> |                   |                   |         CANCEL    |                   |                   |                   |
> |                   |                   |         ---------------------------------------------------------->           |
> |                   |                   |         CANCEL    |                   |                   |                   |
> |                   |                   |         ---------------------------------------------------------->           |
> |                   |                   |         CANCEL    |                   |                   |                   |
> |                   |                   |         ---------------------------------------------------------->           |
> |         CANCEL    |                   |                   |                   |                   |                   |
> |         <--------------------------------------           |                   |                   |                   |
> |         200 OK    |                   |                   |                   |                   |                   |
> |         -------------------------------------->           |                   |                   |                   |
> |         487 Request Cancelled         |                   |                   |                   |                   |
> |         -------------------------------------->           |                   |                   |                   |
> |                   |         200 OK    |                   |                   |                   |                   |
> |                   |         <------------------           |                   |                   |                   |
> |         200 OK    |                   |                   |                   |                   |                   |
> |         <------------------           |                   |                   |                   |                   |
> |         ACK       |                   |                   |                   |                   |                   |
> |         <--------------------------------------           |                   |                   |                   |
> |                   |         487 Request Cancelled         |                   |                   |                   |
> |                   |         <------------------           |                   |                   |                   |
> |         487 Request Cancelled         |                   |                   |                   |                   |
> |         <------------------           |                   |                   |                   |                   |
> |         ACK       |                   |                   |                   |                   |                   |
> |         ------------------>           |                   |                   |                   |                   |
> |                   |         ACK       |                   |                   |                   |                   |
> |                   |         ------------------>           |                   |                   |                   |
>   
> ------------------------------------------------------------------------
>
> | 192.168.0.60      | 192.168.0.207     | 192.168.0.58      | 192.168.0.1       | 192.168.0.38      |
> |         INVITE    |                   |                   |                   |                   |
> |         ------------------>           |                   |                   |                   |
> |         407 Proxy authentication required                 |                   |                   |
> |         <------------------           |                   |                   |                   |
> |         ACK       |                   |                   |                   |                   |
> |         ------------------>           |                   |                   |                   |
> |         INVITE    |                   |                   |                   |                   |
> |         ------------------>           |                   |                   |                   |
> |                   |         INVITE    |                   |                   |                   |
> |                   |         ------------------>           |                   |                   |
> |                   |         100 Trying|                   |                   |                   |
> |                   |         <------------------           |                   |                   |
> |         100 Trying|                   |                   |                   |                   |
> |         <------------------           |                   |                   |                   |
> |                   |                   |         INVITE    |                   |                   |
> |                   |                   |         ------------------>           |                   |
> |                   |                   |         INVITE    |                   |                   |
> |                   |                   |         ------------------>           |                   |
> |                   |                   |         INVITE    |                   |                   |
> |                   |                   |         ------------------>           |                   |
> |                   |                   |         INVITE    |                   |                   |
> |                   |                   |         ------------------>           |                   |
> |                   |                   |         INVITE    |                   |                   |
> |                   |                   |         ------------------>           |                   |
> |                   |                   |         INVITE    |                   |                   |
> |                   |                   |         -------------------------------------->           |
> |                   |                   |         INVITE    |                   |                   |
> |                   |                   |         -------------------------------------->           |
> |                   |                   |         INVITE    |                   |                   |
> |                   |                   |         -------------------------------------->           |
> |         CANCEL    |                   |                   |                   |                   |
> |         ------------------>           |                   |                   |                   |
> |                   |         CANCEL    |                   |                   |                   |
> |                   |         ------------------>           |                   |                   |
> |                   |         487 Request Cancelled         |                   |                   |
> |                   |         <------------------           |                   |                   |
> |                   |                   |         CANCEL    |                   |                   |
> |                   |                   |         -------------------------------------->           |
> |         487 Request Cancelled         |                   |                   |                   |
> |         <------------------           |                   |                   |                   |
> |         ACK       |                   |                   |                   |                   |
> |         ------------------>           |                   |                   |                   |
> |         ACK       |                   |                   |                   |                   |
> |         <------------------           |                   |                   |                   |
> |                   |                   |         CANCEL    |                   |                   |
> |                   |                   |         -------------------------------------->           |
> |                   |                   |         CANCEL    |                   |                   |
> |                   |                   |         -------------------------------------->           |
> |                   |                   |         CANCEL    |                   |                   |
> |                   |                   |         -------------------------------------->           |
> |                   |                   |         CANCEL    |                   |                   |
> |                   |                   |         -------------------------------------->           |
> |                   |         480 Temporarily not available |                   |                   |
> |                   |         <------------------           |                   |                   |
> |         480 Temporarily not available |                   |                   |                   |
> |         <------------------           |                   |                   |                   |
>

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