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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