[asterisk-dev] RTP mixer in Asterisk

[Steven Critchfield]

On Mon, 2006-04-10 at 13:29 +0000, Leonardo (listas) wrote:
> 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.

You mean like a conference. OMG, look at the meetme app. Mark was so
ahead of you by several years.

BTW, this was such a -USER question.
-- 
Steven Critchfield <critch at basesys.com>