I will implement a SIP application and I'm considering using Asterisk for
mixing the media streams (audio). Does anybody know if Asterisk supports or
contains a RTP mixer? If so, how to use it?
Just to be a little more clearer: I will send to Asterisk more than one RTP
stream and they must be mixed. The result must be a single stream to be
forwarded to a SIP phone or to the PSTN.
Thanks,
Leonardo
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