Re: [OpenSIPStack] sip parser

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

You will have to use and event queue for that.  See the EventQueue class 
(EventQueue.h).

You need to use the EventQueue(BOOL blockIfEmpty) constructor and set 
the blockIfEmpty to TRUE

m_SDPOfferEventQueue = new EventQueue( TRUE );


The in your virtual method you can do something like

void CallSessionManager::EnqueueSDPAnswer( PString * answer )
{

  m_SDPOfferEventQueue->Enqueue( answer );

}

BOOL CallSessionManager::OnRequireSDPAnswer(
 const SIPMessage & offer,
 SIPMessage & answer,
 CallSession & session
)
{
 /// report this event to external module
 call_your_external_module( ... );

 PString * eventObject = dynmic_cast<PString *>(m_SDPOfferEventQueue->DequeueEvent());
 answer.SetBody( *eventObject  );
 delete eventObject;

 return TRUE;
}


Be careful about the queue.  It will block forever if an event is not 
received so better make sure that there is an event that will unblock it.

HTH,

Joegen


haripriya alapati wrote:
> I am proceeding as per your suggestion.
> But the thing is my audio and network module is dialog based, asynchronous event driven ones. How can i subclass those modules to this sip parser module.
> Still i am struggling with that.
> Can i have any suggestion regarding this.
>
> Thanks in advance.
>
>
>
> On Thu, 26 Jun 2008 jo...@op... wrote :
>   
>> I am going to assume that you sub-classed the CallSession UACore for your gateway.  If this NOT what you did, then you need to give more info.  If this is what you did, all you need to do is to implement the two blank methods in CallSessionManager.  
>> BOOL CallSessionManager::OnRequireSDPAnswer(
>>  const SIPMessage & offer,
>>  SIPMessage & answer,
>>  CallSession & session
>> )
>> {
>>  /// Media UA should implement this method
>>  return FALSE;
>> }
>>
>> BOOL CallSessionManager::OnRequireSDPOffer(
>>  SIPMessage & _offer,
>>  CallSession & session
>> )
>> {
>>  /// Media UA should implement this method
>>  return FALSE;
>> }
>>
>>
>> Lastly, as you can see, the default implementation of the CallSession does not have RTP. 
>> Joegen
>>
>>
>> haripriya alapati wrote:
>>     
>>>   Hi,
>>>
>>> I have already DSP and NETWORK modules. Only thing is I have to include the SIP module to my project.
>>> For Sip parser I am using opensipstack source code. But I am not getting how to separate the sip parser from the rtp and media modules in the opensipsatck code. Can anyone please help me regarding this.
>>>
>>> Thanks in advance. -------------------------------------------------------------------------
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