opensipstack-devel Mailing List for OpenSIPStack (Page 43)

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Joegen,

     Let me settle down in the forum. Just now I have started evaluating this product and would love to contribute either this unfinished doc stuff or something else.

Thanks,

/RAj

Hi Gaurav,

Well it's a work in progress.  I sure wish we can dedicate more time in 
documentations.   If you have some spare time, maybe you can donate your 
time to create the missing chapters.  I will be glad to publish it and 
recognize your contributions.  ;-)

Joegen

Gaurav Kheterpal wrote:
> The below link works fine for me too.
>
> I guess what Raj may have meant was that though we have entries for Chapters
> 4.3 up to Chapter 7, they are not active links under
> http://www.opensipstack.org/sbc_man_content.html
>
> Moreover, the 'Next' link in Section 4 takes the user back to Section 2.
>
> Regards,
> Gaurav
>
>   
>> -----Original Message-----
>> From: ope...@li...
>> [mailto:ope...@li...] On Behalf Of
>> Joegen E. Baclor
>> Sent: Tuesday, September 11, 2007 6:56 AM
>> To: ope...@op...; opensipstack-
>> de...@li...
>> Subject: Re: [OpenSIPStack] OpneSBC documentaion link failure
>>
>> Hi,
>>
>> I just checked this link and the page displayed correctly.  Anyone else
>> experiencing this?
>>
>> Anyway, the documentation can be downloaded from CVS together with the
>> OpenSBC module.
>>
>> Joegen
>>
>>
>> Rajpal Dangi wrote:
>>     
>>> In docs section at
>>>       
>> [http://www.opensipstack.org/sbc_man_install.html#2.1] ,  all of the
>> contents are NOT displayed. I wonder if this is errro or  under
>> development.
>>     
>>> Regards,
>>>
>>>
>>> /RAj
>>>
>>>
>>>
>>>
>>> ------------------------------------------------------------------------
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>> -
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>>     
>
>
>   

The below link works fine for me too.

I guess what Raj may have meant was that though we have entries for Chapters
4.3 up to Chapter 7, they are not active links under
http://www.opensipstack.org/sbc_man_content.html

Moreover, the 'Next' link in Section 4 takes the user back to Section 2.

Regards,
Gaurav

> -----Original Message-----
> From: ope...@li...
> [mailto:ope...@li...] On Behalf Of
> Joegen E. Baclor
> Sent: Tuesday, September 11, 2007 6:56 AM
> To: ope...@op...; opensipstack-
> de...@li...
> Subject: Re: [OpenSIPStack] OpneSBC documentaion link failure
> 
> Hi,
> 
> I just checked this link and the page displayed correctly.  Anyone else
> experiencing this?
> 
> Anyway, the documentation can be downloaded from CVS together with the
> OpenSBC module.
> 
> Joegen
> 
> 
> Rajpal Dangi wrote:
> > In docs section at
> [http://www.opensipstack.org/sbc_man_install.html#2.1] ,  all of the
> contents are NOT displayed. I wonder if this is errro or  under
> development.
> >
> >
> > Regards,
> >
> >
> > /RAj
> >
> >
> >
> >
> > ------------------------------------------------------------------------
> -
> > This SF.net email is sponsored by: Microsoft
> > Defy all challenges. Microsoft(R) Visual Studio 2005.
> > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/
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> > ope...@li...
> > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> >
> >
> 
> 
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Jack Damn wrote:
> I have a problem where OSBC is running as B2BUpperReg and successfully
> registers a Linksys
> SIP phone (using OSBC as the outbound proxy) to the upstream registrar (a
> B2BUA gateway).
>
> However, in the mangled REGISTER message OSBC sends, it sets the contact
> header to
> its IP address but it's not using port 5060, but the back door port (65080).
>   

That is correct.

> Now what happens is that INVITES from this SIP phone routes through OSBC in
> B2BUA mode
> but OSBC uses source port 5060 to send that invite to the gateway instead of
> using the backdoor
> port as the gatweay has in it active registration tables.  The call is
> therefore denied with a 403 Forbidden.
>   

Hmmn, this behavior is specific to your gateway.  Can you humor me with 
the type and manufacturer?   In upper Registration, egress traffic is 
sent via 5060, while ingress are listening using the backdoor.    This  
approach is due to the fact that calls spawned  by  OpenSBC may spiral 
back to it.  If the call wen through  5060,  call-id collision will 
arise for the B2BUA.    Routing would also be confused because the dial 
string will be the same, thus producing a loop.   Using two different 
managers for ingress and egress takes care of this problem.  There is 
nothing illegal in this approach. 

> Is this a bug in OSBC ?  -or- is there an option I am missing to instruct
> OSBC to use the backdoor
> port in requests to the gateway ?
> Better, I would rather like OSBC to send the REGISTER with the default  port
> 5060 in the contact.
>
>   

No its not a bug in OpenSBC.  The gateway should allow Invites coming 
from other ports.

> Is there documetation available that explains in details the purpose and
> role of the back door thread ?
>
> Calls from the gateway -> OSBC -> SIP phone are working just fine.
>
> Thank you,
> JD
> -------------------------------------------------------------------------
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> ope...@li...
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>
>   

I have a problem where OSBC is running as B2BUpperReg and successfully
registers a Linksys
SIP phone (using OSBC as the outbound proxy) to the upstream registrar (a
B2BUA gateway).

However, in the mangled REGISTER message OSBC sends, it sets the contact
header to
its IP address but it's not using port 5060, but the back door port (65080).

Now what happens is that INVITES from this SIP phone routes through OSBC in
B2BUA mode
but OSBC uses source port 5060 to send that invite to the gateway instead of
using the backdoor
port as the gatweay has in it active registration tables.  The call is
therefore denied with a 403 Forbidden.

Is this a bug in OSBC ?  -or- is there an option I am missing to instruct
OSBC to use the backdoor
port in requests to the gateway ?
Better, I would rather like OSBC to send the REGISTER with the default  port
5060 in the contact.

Is there documetation available that explains in details the purpose and
role of the back door thread ?

Calls from the gateway -> OSBC -> SIP phone are working just fine.

Thank you,
JD

Hi,

I just checked this link and the page displayed correctly.  Anyone else 
experiencing this?   

Anyway, the documentation can be downloaded from CVS together with the 
OpenSBC module.

Joegen

Rajpal Dangi wrote:
> In docs section at [http://www.opensipstack.org/sbc_man_install.html#2.1] ,  all of the contents are NOT displayed. I wonder if this is errro or  under development.
>
>
> Regards,
>
>
> /RAj
>
>
>
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Microsoft
> Defy all challenges. Microsoft(R) Visual Studio 2005.
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> _______________________________________________
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> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>   

In docs section at [http://www.opensipstack.org/sbc_man_install.html#2.1] ,  all of the contents are NOT displayed. I wonder if this is errro or  under development.

Regards,

/RAj

OK. Thanks a lot. I changed some functionality in this funciton and now everythign works great !

Thank you. 

It will be good if you know what exactly your server hates about the 
REGISTER request we send to refresh the bindings.   From what I 
remember, OpenSBC sends exactly the same REGISTER request but with an 
incremented CSeq.  Check out void 
RegisterSession::SendRegistrationRefresh( PTimer &, INT ) method.

tomach wrote:
> <p>
> Hello guys I need help,
> </p>
> <p>
>  
> </p>
> <p>
> I try to describe my problem:
> </p>
> <p>
> My softphone based on atlsip is registering to my sipServer, everything works fine. But after about 30-40 minutes it send registering packet again(its good behaviour) but...my sipserver do not recoginse it because there is small difference before first and second register message send by softphone).
> </p>
> <p>
> First register:
> </p>
> <p>
> Session Initiation Protocol<br />
>     Request-Line: REGISTER sip:sipx.dgt.com.pl SIP/2.0<br />
>         Method: REGISTER<br />
>         Resent Packet: False              &lt;-- this is good here<br />
> <br />
>     Message Header<br />
> <br />
> <br />
> second register:
> </p>
> <p>
> Session Initiation Protocol<br />
>     Request-Line: REGISTER sip:sipx.dgt.com.pl SIP/2.0<br />
>         Method: REGISTER
> </p>
> <p>
>         Resent Packet: True          &lt;-----this should be like previous  (False)
> </p>
> <p>
>         Suspected resend of frame: 1216   &lt;-----here my sipServer do not recognise it totally and response with out of order
> </p>
> <p>
>     Message Header<br />
> </p>
> <p>
>  
> </p>
> <p>
> So after second regiser message my SIPserver response with out of order and softphone gets unregistered...
> </p>
> <p>
>  
> </p>
> <p>
> MY quiestion is how to disable it? this information that the packet is resend? I was trying to locate it in codes but couldnt find it....Please any help is appreciate, because now my softphone only works for half an hour...:(
> </p>
> <p>
>  
> </p>
> <p>
> Best Regards,
> </p>
> <p>
> Tom 
> </p>
>
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>   

I have found only this information:


 * Revision 1.18 2006/12/20 16:24:44 joegenbaclor 
 * Implemented keep-alives for upper reg.


But dont know where I should change the code that all messages that resend information about registration didnt send those two lines:


Resent Packet: True 


Suspected resend of frame: 1216


and insead of it they shoudl send:

Resent Packet: False
</p>
<p>

</p>
<p>
How to do it?
</p>
<p>
 BR,
</p>
<p>
TOM 
</p>
<p>

</p>

<p>
Hello guys I need help,
</p>
<p>

</p>
<p>
I try to describe my problem:
</p>
<p>
My softphone based on atlsip is registering to my sipServer, everything works fine. But after about 30-40 minutes it send registering packet again(its good behaviour) but...my sipserver do not recoginse it because there is small difference before first and second register message send by softphone).
</p>
<p>
First register:
</p>
<p>
Session Initiation Protocol<br />
    Request-Line: REGISTER sip:sipx.dgt.com.pl SIP/2.0<br />
        Method: REGISTER<br />
        Resent Packet: False              &lt;-- this is good here<br />
<br />
    Message Header<br />
<br />
<br />
second register:
</p>
<p>
Session Initiation Protocol<br />
    Request-Line: REGISTER sip:sipx.dgt.com.pl SIP/2.0<br />
        Method: REGISTER
</p>
<p>
        Resent Packet: True          &lt;-----this should be like previous  (False)
</p>
<p>
        Suspected resend of frame: 1216   &lt;-----here my sipServer do not recognise it totally and response with out of order
</p>
<p>
    Message Header<br />
</p>
<p>

So after second regiser message my SIPserver response with out of order and softphone gets unregistered...

MY quiestion is how to disable it? this information that the packet is resend? I was trying to locate it in codes but couldnt find it....Please any help is appreciate, because now my softphone only works for half an hour...:(

</p>
<p>
Best Regards,
</p>
<p>
Tom 
</p>

OK I have solved the problem :).


 The probelm was in win sock. In winxp w32_s.dll are some functions that are not avaliable in win2000 so I had to add two .h files to compilation and now everything work great :)

</p>
<p>

</p>
<p>
Best regards,
</p>
<p>
Tom 
</p>

hmm its weird because in the past it worked correct....now i can not register activex. I used many scenarios to fix it even i didnt change anything since it worked correct....ANywya now i tried in many differnet ways:


1. I used viusal studio under winxp to create instal verison with all dependecices....but on win2k i still couldnt register..


2. I used dependecy walker to check what ATLSIP.dll is missing..it told me that it missing apphelp.dll. Its weird becaus on msdn pages i have found that win2k does not include apphelp.dll. Anyway I copied itformwinxp. It still didnt work. Dll dependecy walker said that apphelp.dll is missing some fuctions from kernel32.dll that were available in winxp....:( 


Did you find anywere apphelp.dll for win2k? how did you fix this dependecy?

</p>
<p>
Best Regards,
</p>
<p>
Tom 
</p>

Tomach,

  You could download the softphone that we made that installs on different windows version at 
https://acplus.jp/accessplus1/install_eng.html

"Ilian Jeri C. Pinzon" <ip...@so...> wrote:
 Haven't tried it on Windows 2K before. Try manually registering ATLSIP by:

1. VS Studio -> Tools -> Activex Control Test Container -> File -> 
Register Control or
2. regsvr32

- Ilian

tomach wrote:
>   
> Hello!
> 

>   
> Has anyone of you have problem with registering activex under windows 2000? Even i compile whole project under windows2000 I still can not register it.....Please guys any ideas? 
> 

>   
> Under winXP i have no problems.. 
> 

>   
> 
> 

>   
> Best Regards,
> 

>   
> Tom 
> 

>
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Tomach,

  Use depends.exe to get the dll's that are missing. I made all versions in Windows and all is working. (Win2k, Win2k Server, XP, Vista)

tomach <to...@dg...> wrote:

Hello!

Has anyone of you have problem with registering activex under windows 2000? Even i compile whole project under windows2000 I still can not register it.....Please guys any ideas? 

Under winXP i have no problems.. 

Best Regards,

Tom 

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

Haven't tried it on Windows 2K before. Try manually registering ATLSIP by:

1. VS Studio -> Tools -> Activex Control Test Container -> File -> 
Register Control or
2. regsvr32

- Ilian

tomach wrote:
> <p>
> Hello!
> </p>
> <p>
> Has anyone of you have problem with registering activex under windows 2000? Even i compile whole project under windows2000 I still can not register it.....Please guys any ideas? 
> </p>
> <p>
> Under winXP i have no  problems.. 
> </p>
> <p>
>  
> </p>
> <p>
> Best Regards,
> </p>
> <p>
> Tom 
> </p>
>
> -------------------------------------------------------------------------
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>
>   

Hello!


Has anyone of you have problem with registering activex under windows 2000? Even i compile whole project under windows2000 I still can not register it.....Please guys any ideas? 


Under winXP i have no problems..

</p>
<p>
Best Regards,
</p>
<p>
Tom 
</p>

Thanks Joegen,
Your inputs will certainly help us.

-Ashish

On 9/5/07, Joegen E. Baclor <joe...@gm...> wrote:
>
> Ashish Khare wrote:
> > But against what you will check the SSRC of the first RTP packet
> recieved
> > from A.
> > Means against what you will check the SSRC of incoming first RTP packet
> from
> > A.
> >
>
> I can see where you are coming from.  The basis of which SSRC to accept
> would be hard if not impossible to determine.   I guess this is a
> limitation of remote media anchoring and implementors must put up other
> mechanisms to prevent such malicious attacks from happening.   If you
> have control over your UA, one possible way is to compute the SSRC from
> the Session ID offered in the SDP and have that as the basis for which
> SSRC you must accept.
>
> > -Ashish
> >
> >
> > On 9/5/07, Joegen E. Baclor <joe...@gm...> wrote:
> >
> >> Ashish Khare wrote:
> >>
> >>> But then in second case
> >>> does it will pose a security threat  ?
> >>> Because anyone can spoof an IP address and sends the RTP from
> different
> >>>
> >> port
> >>
> >>> ?
> >>>
> >>>
> >>>
> >>>
> >> Technically yes.  But this can be augmented by checking the ssrc of the
> >> RTP packet.  OpenSBC can be set to ignore RTP packets if the ssrc has
> >> changed.
> >>
> >>
> >>
> >>> On 9/5/07, Joegen E. Baclor <joe...@gm...> wrote:
> >>>
> >>>
> >>>> Ashish Khare wrote:
> >>>>
> >>>>
> >>>>> Hi Joegen,
> >>>>>
> >>>>> I have two questions for RTP communication in openSBC.
> >>>>>
> >>>>> 1) clients have to use the same RTP ports to send the RTP packets on
> >>>>>
> >>>>>
> >>>> which
> >>>>
> >>>>
> >>>>> they recieve RTP packets. So if the client is recieveing RTP packets
> >>>>>
> >> on
> >>
> >>>> port
> >>>>
> >>>>
> >>>>> 8000, Cient will use 8000 only to send the RTP packets?
> >>>>>
> >>>>>
> >>>>>
> >>>> Most implementations do this, yes.  In fact this behavior is needed
> for
> >>>> OpenSBC NAT traversal to work.
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>> 2) When natted client initiated the call,  for example Client A
> which
> >>>>>
> >> is
> >>
> >>>>> behind a NAT, calls B through openSBC, the first RTP packets need to
> >>>>>
> >> be
> >>
> >>>>> recieved from Client A, at openSBC, in order to send the RTP packets
> >>>>>
> >>>>>
> >>>> from
> >>>>
> >>>>
> >>>>> Client B to Client A. this is because of NAT device, the RTP packets
> >>>>>
> >>>>>
> >>>> from A
> >>>>
> >>>>
> >>>>> comes from different port than what was said in the SDP offfer and
> >>>>>
> >>>>>
> >>>> therefore
> >>>>
> >>>>
> >>>>> RTP packets from B need to be send to this new port.  is my
> >>>>>
> >>>>>
> >>>> understanding
> >>>>
> >>>>
> >>>>> correct. Please clarify.?
> >>>>>
> >>>>> Please reply.
> >>>>>
> >>>>>
> >>>>>
> >>>> Yes, this is exactly how its done in OpenSBC.
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>> Thanks.
> >>>>>
> >>>>> -Ashish
> >>>>>
> >>>>>
> >>>>>
> >>
> -------------------------------------------------------------------------
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Ashish Khare wrote:
> But against what you will check the SSRC of the first RTP packet recieved
> from A.
> Means against what you will check the SSRC of incoming first RTP packet from
> A.
>   

I can see where you are coming from.  The basis of which SSRC to accept 
would be hard if not impossible to determine.   I guess this is a 
limitation of remote media anchoring and implementors must put up other 
mechanisms to prevent such malicious attacks from happening.   If you 
have control over your UA, one possible way is to compute the SSRC from 
the Session ID offered in the SDP and have that as the basis for which 
SSRC you must accept. 

> -Ashish
>
>
> On 9/5/07, Joegen E. Baclor <joe...@gm...> wrote:
>   
>> Ashish Khare wrote:
>>     
>>> But then in second case
>>> does it will pose a security threat  ?
>>> Because anyone can spoof an IP address and sends the RTP from different
>>>       
>> port
>>     
>>> ?
>>>
>>>
>>>
>>>       
>> Technically yes.  But this can be augmented by checking the ssrc of the
>> RTP packet.  OpenSBC can be set to ignore RTP packets if the ssrc has
>> changed.
>>
>>
>>     
>>> On 9/5/07, Joegen E. Baclor <joe...@gm...> wrote:
>>>
>>>       
>>>> Ashish Khare wrote:
>>>>
>>>>         
>>>>> Hi Joegen,
>>>>>
>>>>> I have two questions for RTP communication in openSBC.
>>>>>
>>>>> 1) clients have to use the same RTP ports to send the RTP packets on
>>>>>
>>>>>           
>>>> which
>>>>
>>>>         
>>>>> they recieve RTP packets. So if the client is recieveing RTP packets
>>>>>           
>> on
>>     
>>>> port
>>>>
>>>>         
>>>>> 8000, Cient will use 8000 only to send the RTP packets?
>>>>>
>>>>>
>>>>>           
>>>> Most implementations do this, yes.  In fact this behavior is needed for
>>>> OpenSBC NAT traversal to work.
>>>>
>>>>
>>>>
>>>>         
>>>>> 2) When natted client initiated the call,  for example Client A which
>>>>>           
>> is
>>     
>>>>> behind a NAT, calls B through openSBC, the first RTP packets need to
>>>>>           
>> be
>>     
>>>>> recieved from Client A, at openSBC, in order to send the RTP packets
>>>>>
>>>>>           
>>>> from
>>>>
>>>>         
>>>>> Client B to Client A. this is because of NAT device, the RTP packets
>>>>>
>>>>>           
>>>> from A
>>>>
>>>>         
>>>>> comes from different port than what was said in the SDP offfer and
>>>>>
>>>>>           
>>>> therefore
>>>>
>>>>         
>>>>> RTP packets from B need to be send to this new port.  is my
>>>>>
>>>>>           
>>>> understanding
>>>>
>>>>         
>>>>> correct. Please clarify.?
>>>>>
>>>>> Please reply.
>>>>>
>>>>>
>>>>>           
>>>> Yes, this is exactly how its done in OpenSBC.
>>>>
>>>>
>>>>
>>>>         
>>>>> Thanks.
>>>>>
>>>>> -Ashish
>>>>>
>>>>>
>>>>>           
>> -------------------------------------------------------------------------
>>     
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>   

But against what you will check the SSRC of the first RTP packet recieved
from A.
Means against what you will check the SSRC of incoming first RTP packet from
A.

-Ashish

On 9/5/07, Joegen E. Baclor <joe...@gm...> wrote:
>
> Ashish Khare wrote:
> > But then in second case
> > does it will pose a security threat  ?
> > Because anyone can spoof an IP address and sends the RTP from different
> port
> > ?
> >
> >
> >
>
> Technically yes.  But this can be augmented by checking the ssrc of the
> RTP packet.  OpenSBC can be set to ignore RTP packets if the ssrc has
> changed.
>
>
> > On 9/5/07, Joegen E. Baclor <joe...@gm...> wrote:
> >
> >> Ashish Khare wrote:
> >>
> >>> Hi Joegen,
> >>>
> >>> I have two questions for RTP communication in openSBC.
> >>>
> >>> 1) clients have to use the same RTP ports to send the RTP packets on
> >>>
> >> which
> >>
> >>> they recieve RTP packets. So if the client is recieveing RTP packets
> on
> >>>
> >> port
> >>
> >>> 8000, Cient will use 8000 only to send the RTP packets?
> >>>
> >>>
> >> Most implementations do this, yes.  In fact this behavior is needed for
> >> OpenSBC NAT traversal to work.
> >>
> >>
> >>
> >>> 2) When natted client initiated the call,  for example Client A which
> is
> >>> behind a NAT, calls B through openSBC, the first RTP packets need to
> be
> >>> recieved from Client A, at openSBC, in order to send the RTP packets
> >>>
> >> from
> >>
> >>> Client B to Client A. this is because of NAT device, the RTP packets
> >>>
> >> from A
> >>
> >>> comes from different port than what was said in the SDP offfer and
> >>>
> >> therefore
> >>
> >>> RTP packets from B need to be send to this new port.  is my
> >>>
> >> understanding
> >>
> >>> correct. Please clarify.?
> >>>
> >>> Please reply.
> >>>
> >>>
> >> Yes, this is exactly how its done in OpenSBC.
> >>
> >>
> >>
> >>> Thanks.
> >>>
> >>> -Ashish
> >>>
> >>>
> >>
> -------------------------------------------------------------------------
> >>
> >>> This SF.net email is sponsored by: Splunk Inc.
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> browser.
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> >>> ope...@li...
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> >>>
> >>>
> >>>
> >>>
> >>
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> >>
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> >
> >
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>
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Good day,=20

A patch was made  to handle invite message coming from RTC clients.=20

Invite messages coming from sip client using RTC does not contain any =
branch parameter in its Via header field, in order to process it =
properly opensipstack will set=20
m_Branch to "empty-branch".

Regards,
Julian
  ----- Original Message -----=20
  From: Julian F. Tasis, III=20
  To: ope...@li...=20
  Sent: Wednesday, September 05, 2007 9:53 AM
  Subject: Error on incoming invite message

  Good day,=20

  In CallSession.cxx and OnIncomingSIPMessage

  OString fsm =3D tid.GetStateMachine();

  When a call came from a sip client using RTC the variable fsm is null =
and that call will not be processed.

  For a workaround I just modified the else if part when fsm is equals =
to IST.

  I don't know if my workaround is the right way of handling such =
transaction and is it a must that fsm variable can't be null?

  Note:=20

  This only happens when a sip client is using RTC.

  Regards,
  Julian 

Ashish Khare wrote:
> But then in second case
> does it will pose a security threat  ?
> Because anyone can spoof an IP address and sends the RTP from different port
> ?
>
>
>   

Technically yes.  But this can be augmented by checking the ssrc of the 
RTP packet.  OpenSBC can be set to ignore RTP packets if the ssrc has 
changed.

> On 9/5/07, Joegen E. Baclor <joe...@gm...> wrote:
>   
>> Ashish Khare wrote:
>>     
>>> Hi Joegen,
>>>
>>> I have two questions for RTP communication in openSBC.
>>>
>>> 1) clients have to use the same RTP ports to send the RTP packets on
>>>       
>> which
>>     
>>> they recieve RTP packets. So if the client is recieveing RTP packets on
>>>       
>> port
>>     
>>> 8000, Cient will use 8000 only to send the RTP packets?
>>>
>>>       
>> Most implementations do this, yes.  In fact this behavior is needed for
>> OpenSBC NAT traversal to work.
>>
>>
>>     
>>> 2) When natted client initiated the call,  for example Client A which is
>>> behind a NAT, calls B through openSBC, the first RTP packets need to be
>>> recieved from Client A, at openSBC, in order to send the RTP packets
>>>       
>> from
>>     
>>> Client B to Client A. this is because of NAT device, the RTP packets
>>>       
>> from A
>>     
>>> comes from different port than what was said in the SDP offfer and
>>>       
>> therefore
>>     
>>> RTP packets from B need to be send to this new port.  is my
>>>       
>> understanding
>>     
>>> correct. Please clarify.?
>>>
>>> Please reply.
>>>
>>>       
>> Yes, this is exactly how its done in OpenSBC.
>>
>>
>>     
>>> Thanks.
>>>
>>> -Ashish
>>>
>>>       
>> -------------------------------------------------------------------------
>>     
>>> 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.
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>>> ope...@li...
>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>>
>>>
>>>
>>>       
>> -------------------------------------------------------------------------
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>>
>>     
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>
>
>   

But then in second case
does it will pose a security threat  ?
Because anyone can spoof an IP address and sends the RTP from different port
?

On 9/5/07, Joegen E. Baclor <joe...@gm...> wrote:
>
> Ashish Khare wrote:
> > Hi Joegen,
> >
> > I have two questions for RTP communication in openSBC.
> >
> > 1) clients have to use the same RTP ports to send the RTP packets on
> which
> > they recieve RTP packets. So if the client is recieveing RTP packets on
> port
> > 8000, Cient will use 8000 only to send the RTP packets?
> >
>
> Most implementations do this, yes.  In fact this behavior is needed for
> OpenSBC NAT traversal to work.
>
>
> > 2) When natted client initiated the call,  for example Client A which is
> > behind a NAT, calls B through openSBC, the first RTP packets need to be
> > recieved from Client A, at openSBC, in order to send the RTP packets
> from
> > Client B to Client A. this is because of NAT device, the RTP packets
> from A
> > comes from different port than what was said in the SDP offfer and
> therefore
> > RTP packets from B need to be send to this new port.  is my
> understanding
> > correct. Please clarify.?
> >
> > Please reply.
> >
>
>
> Yes, this is exactly how its done in OpenSBC.
>
>
> > Thanks.
> >
> > -Ashish
> >
> -------------------------------------------------------------------------
> > 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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> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>

Ashish Khare wrote:
> Hi Joegen,
>
> I have two questions for RTP communication in openSBC.
>
> 1) clients have to use the same RTP ports to send the RTP packets on which
> they recieve RTP packets. So if the client is recieveing RTP packets on port
> 8000, Cient will use 8000 only to send the RTP packets?
>   

Most implementations do this, yes.  In fact this behavior is needed for 
OpenSBC NAT traversal to work.

> 2) When natted client initiated the call,  for example Client A which is
> behind a NAT, calls B through openSBC, the first RTP packets need to be
> recieved from Client A, at openSBC, in order to send the RTP packets from
> Client B to Client A. this is because of NAT device, the RTP packets from A
> comes from different port than what was said in the SDP offfer and therefore
> RTP packets from B need to be send to this new port.  is my understanding
> correct. Please clarify.?
>
> Please reply.
>   

Yes, this is exactly how its done in OpenSBC.

> Thanks.
>
> -Ashish
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems?  Stop.
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>
>
>   

  I've been working on an UA project recently. Most =
functionalities=C2=A0an UA should have=C2=A0are added. Now I'm trying to =
add video support. And currently I am able to capture the data from a web =
camera and display it. Although I know video support has been planned, =
I=C2=A0wonder how I can achieve this by myself.=C2=A0Can you=C2=A0tell me =
the knowledge or necessities I should possess?

Regards,
GCC

____________________________________________________________
GET FREE SMILEYS FOR YOUR IM & EMAIL - Learn more at =
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