[asterisk-dev] RTP mixer in Asterisk

[Leonardo (listas)]

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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