Re: [OpenSIPStack] Server implementation question

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Thomas Raschbacher wrote:
> Thanks. The 1st one solved my problem.
>
> So .. if I wanted to use a custom authentication class I assume I'd hav=
e to inherit from B2BAuthInterface right? Also how would I then use that?=

> Meaning I still want opensipstack to handle the authentification when i=
t gets a REGISTER (with www-auth,..), but want to get the user-credential=
s from our system instead of storing them in a list like OpenSBC does..=20
>  =20

You are looking at the right place.  For a guy asking for=20
documentations, you're doing quire well.

> To clarify I want opensipstack to handle the sip requests, but need it =
to call methods in our application to get information about users, wether=
 to allow a call or not, ... (also want to always route RTP through the a=
pplication so that the conversation can be recorded if needed.)
>  =20

If you want to not let opensbc handle RTP, you may need to override=20
methods in B2BMediaInterface.   You may simply comment out the entire=20
code blocks of TranslateSDPOffer and TranslateSDPAnswer and just return=20
TRUE in both places.

> /me continues evaluation of opensipstack ;)
>
> -----Original Message-----
> From: ope...@li... [mailto:opensips=
tac...@li...] On Behalf Of joegen@opensipstac=
k.org
> Sent: Freitag, 23. November 2007 11:28
> To: ope...@li...
> Subject: Re: [OpenSIPStack] Server implementation question
>
> Two possible reasons:
>
> 1.  You are passing the address of SIPMessage msg to the SIPTransport
> event which will be destroyed as soon as this function loses scope.
> Try
>
> new SIPTransports::SIPTransportEvent( new SIPMessage(msg),
> SIPTransports::SIPTransportEvent::UDPPacketArrival);
>
> instead.  SIPTransportEvent deletes this pointer in its destructor.
>
> 2.  If this is a debug build, OpenSIPStack might be using PMemoryHeap t=
o
> store NEWly create Objects.  Try putting a "#define new PNEW" in the
> beginning of this file to force you new operator to use the correct
> implementation.
>
>  From what I see, the culprit is item 1.
>
> Thomas Raschbacher wrote:
>  =20
>> Hi.
>>
>> Please have a look at the following code:
>>
>> class OpenSipStack_StackTest
>> {
>> public:
>>         void TransportEventTest(SIPMessage msg);
>>         OpenSipStack_StackTest()
>>         {
>>                 m_Stack =3D new SIP::SIPStack();
>>         }
>> private:
>>         SIP::SIPStack * m_Stack;
>> }
>>
>> void OpenSipStackTest::OpenSipStack_StackTest::TransportEventTest(SIPM=
essage msg)
>> {
>>         SIPTransports::SIPTransportEvent *evt =3D new SIPTransports::S=
IPTransportEvent(&msg, SIPTransports::SIPTransportEvent::UDPPacketArrival=
);
>>
>>         this->m_Stack->OnTransportEvent(evt);
>>         //this->OnTransportEvent(evt); // class was derived from SIP::=
SIPStack before..
>> }
>>
>> This is the program's output (I used a register message from the SIP r=
fc):
>> 23:18:32.858    INF: [CID=3D0x0000] SIP Stack CREATED
>> 23:18:32.943    DBG: [CID=3D0x06e0]
>> 23:18:32.943    DBG: [CID=3D0x06e0]       RECEIVE:  REQ: REGISTER (182=
6)
>> 23:18:32.943    DBG: [CID=3D0x06e0]       FROM: sip:bo...@bi...
>> 23:18:32.943    DBG: [CID=3D0x06e0]       TO: sip:bo...@bi...
>> 23:18:32.943    DBG: [CID=3D0x06e0]       XOR: FALSE
>> 23:18:32.943    DBG: [CID=3D0x06e0]       SRC:  PORT:
>> 23:18:32.943    DBG: [CID=3D0x06e0]
>> 23:18:32.953    DBG: [CID=3D0x06e0]
>> 23:18:32.953    DBG: [CID=3D0x06e0]       *** CREATING TRANSACTION (NI=
ST) ***
>> 23:18:32.953    DBG: [CID=3D0x06e0]       Message: REGISTER sip:regist=
rar.biloxi.c
>> om SIP/2.0
>> 23:18:32.953    DBG: [CID=3D0x06e0]       Call-Id: 843817637684230@998=
sdasdh09
>> 23:18:32.953    DBG: [CID=3D0x06e0]
>> 23:18:32.962    DTL: [CID=3D0x06e0] NIST(83912963)        *** CREATED =
*** - NIST|8
>> 43817637684230@998sdasdh09|z9hG4bKnashds7|REGISTER
>>
>> At that point I get it triggering a breakpoint - heap related
>> HEAP[OpenSipStackTestConsoleApp.exe]: Invalid Address specified to Rtl=
ValidateHeap( 00350000, 0012FC94 )
>> Windows has triggered a breakpoint in OpenSipStackTestConsoleApp.exe.
>>
>> This may be due to a corruption of the heap, and indicates a bug in Op=
enSipStackTestConsoleApp.exe or any of the DLLs it has loaded.
>>
>> The output window may have more diagnostic information
>>
>> (that is all from the output window..)
>>
>> This seems to happen at the end of the OnTransportEvent method (SIPSta=
ck.cxx: 431 ... delete event; )
>>
>> Any ideas?
>>
>> Thanks
>>
>> -----Original Message-----
>> From: ope...@li... [mailto:opensip=
sta...@li...] On Behalf Of joegen@opensipsta=
ck.org
>> Sent: Freitag, 23. November 2007 10:27
>> To: ope...@li...
>> Subject: Re: [OpenSIPStack] Server implementation question
>>
>> SIP::SIPStack is probably the best place for you then.  You just need =
to
>> create your own SIPTransportEvent and feed it to OnTransportEvent()
>> virtual method.   I agree that there is not much to work with in terms=

>> of documentation and I assure you that this is not a permanent
>> condition.   It is just that all the hands working on the project are
>> currently busy with actual coding of functionalities.   For now, you'r=
e
>> best bet is to ask questions and we can answer it to the best of our
>> abilities.   I also have plans of creating a near C API wrapper for th=
e
>> stack to make implementors and integrators life easier.    One thing
>> that might be a bit of a road block for you is that OpenSIPStack desig=
n
>> is not modular and that almost all components like the Transport, FSM
>> and the core are intertwined.  Thus, if you need to use third party
>> replace ments, you will need to subclass the right component.
>>
>>
>> Thomas Raschbacher wrote:
>>
>>    =20
>>> Joegen,
>>>
>>> Well the main thing is that the transport layer is partially existing=
 already and needs to be implemented in some other part of the applicatio=
n I think (also we need TCP and TLS too). Problem is that we need to do t=
he switching and deciding if a call is valid or not,.. ourselves.
>>> Maybe I missed something and this functionality can be achieved other=
wise (our application does have hardware to connect to analogue/digital p=
hones as well as does the switching and more or less the 'media server/pr=
oxy' too. Of course we want to use our switching and the media proxying -=
 since call recording is a required features.)
>>>
>>> If there is an easier way to achieve this please let me know as it is=
 a bit difficult to find those things out as I couldn't find a lot of dev=
eloper documentation / the code isn't commented very much explaining whic=
h class exactly does what.
>>>
>>> Regards,
>>> P.S.: I'm currently playing around with sending Transport events to a=
 derived class of SIP::SIPStack and having that process the event to find=
 out exactly what level of processing this class does already ... It woul=
d really help me if there was some sort of documentation like an architec=
ture diagram or some short description of which class would be best to us=
e if I want some basic replies and timeouts already implemented, as I don=
't know if SIPStack is even the right choice or if I rather should look a=
t SIPFSM, ...
>>>
>>>
>>> -----Original Message-----
>>> From: ope...@li... [mailto:opensi=
pst...@li...] On Behalf Of Joegen E. Baclor=

>>> Sent: Freitag, 23. November 2007 09:39
>>> To: jo...@op...; ope...@li...=

>>> Subject: Re: [OpenSIPStack] Server implementation question
>>>
>>> Thomas,
>>>
>>> BTW, OpenSIPStack default license is MPL 1.1 not GPL/LGPL.   If you a=
re
>>> afraid that the modification below will render your proprietary code
>>> violating the license, do not worry.  The only thing covered by the M=
PL
>>> is the modification done to OpenSIPStack alone and not your entire
>>> application.  Thus, If you indeed deploy your own transport, then the=

>>> code block below is the only thing you are obliged to make open or
>>> contribute back to OpenSIPStack.    Can you share why you want to use=

>>> another transport though?
>>>
>>> Joegen
>>>
>>> jo...@op... wrote:
>>>
>>>
>>>      =20
>>>> Hi Thomas,
>>>>
>>>> You need to create your custom transports as descendant of
>>>> SIPTransport.   You would then need to override the following Virtua=
l
>>>> Method in SIPTransportManager to return an instance fo your custom t=
ransport
>>>>
>>>> SIPTransport * SIPTransportManager::OnCreateTransport(
>>>>   const char * userAgentName,
>>>>   SIPTransport::Protocol protocol
>>>> )
>>>> {
>>>>   if( protocol =3D=3D SIPTransport::UDP )
>>>>     return new SIPUDPTransport( userAgentName, this );
>>>>   else if( protocol =3D=3D SIPTransport::TCP )
>>>>     return new SIPTCPTransport( userAgentName, this );
>>>>
>>>>   return NULL;
>>>> }
>>>>
>>>>
>>>> Hope that helps.
>>>>
>>>> Joegen
>>>>
>>>> Thomas Raschbacher wrote:
>>>>
>>>>
>>>>
>>>>        =20
>>>>> Hi,
>>>>>
>>>>> If I wanted to use a custom transport without changing opensipstack=
code itself (ie it being in a different library and just sending events t=
o the stack when it receives / sends/sent messages) how would I best do t=
hat? (Need to use existing transports in non (L)GPL application, but want=
 to use as much of opensipstack as possible - the parser itself is no pro=
blem of course, but I'd like to use a higher 'level' than just that which=
 also implements timers and responses like TRYING, ...)
>>>>>
>>>>> Regards,
>>>>>
>>>>> Mit freundlichen Gr=FC=DFen
>>>>> Thomas Raschbacher
>>>>> ____________________________________________
>>>>> itCampus Technology GmbH
>>>>> =D6sterreich * Deutschland * Italien
>>>>> Wehlistra=DFe 27b
>>>>> 1200 Wien
>>>>> tho...@it...
>>>>> Tel: +43 (1) 33418557-58
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>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>          =20
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