Hi,
We are working on the conference server and we
want to hook RTP packets to do the mixing rather than use the create
conference function of sipXtapi. We've read through the messages on the
mailing list and also send some messages before and receive the answer
that there is no way to hook the RTP packets. But I recently find out a
function of sipXtapi:
SIPXTAPI_API SIPX_RESULT sipxCallGetAudioRtpSourceIds(const SIPX_CALL hCall,
unsigned int& iSendSSRC,
unsigned int& iReceiveSSRC) ;
The
description of this function says that it can be used to receive the
SSRC IDs of RTP packets. The SSRC ID is used to identify the RTP/audio
stream. This fuction is
only supported when sipXtapi is bundled with VoiceEngine from GIPS.
Any one can help us answer following questions:
1.
What can this function do to hook RTP packets? Can we use SSRC IDs of
RTP packets to do the mixing (We read on some papers that we should not
use SSRC IDs to do the mixing for some reasons)?
2. What is the VoiceEngine from GISP? How can we use the VoiceEngine with this function?
Thank you very much.
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