Open Source Sip: [OpenSIPStack] sip parser ...

Up to Discussions in OpenSIPStack

3 Replies Last post: Jul 29, 2008 6:26 AM by Guest

Guest

Jun 26, 2008 2:08 AM

[OpenSIPStack] sip parser

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.

Check out the new SourceForge.net Marketplace.
It's the best place to buy or sell services for
just about anything Open Source.
http://sourceforge.net/services/buy/index.php
_______________________________________________
opensipstack-devel mailing list
opensipstack-devel@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

Guest

1. Jun 26, 2008 3:26 AM

in response to: Guest

Re: [OpenSIPStack] sip parser

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:

Guest

2. Jul 29, 2008 6:04 AM

in response to: Guest

Re: [OpenSIPStack] sip parser

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 joegen@opensipstack.org 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.

This SF.Net email is sponsored by the Moblin Your Move Developer's challenge
Build the coolest Linux based applications with Moblin SDK & win great prizes
Grand prize is a trip for two to an Open Source event anywhere in the world
http://moblin-contest.org/redirect.php?banner_id=100&url=/
_______________________________________________
opensipstack-devel mailing list
opensipstack-devel@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

Guest

3. Jul 29, 2008 6:26 AM

in response to: Guest

Re: [OpenSIPStack] sip parser

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(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:

On Thu, 26 Jun 2008 joegen@opensipstack.org 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.

Go to original post

Jun 26, 2008 2:08 AM

[OpenSIPStack] sip parser

Actions

More Like This