Use RTP stream without SIP

Is there a way for asterisk to accept plain RTP traffic (no SIP setup) and automatically direct the stream to a meet me conference.

For example, I want to stream RTP traffic to my asterisk box at ip: The source of the RTP comes from and is streaming to asterisk on port 5062. Can asterisk take the stream, seeing that it is coming from ip: and is arriving on port 5062 and automatically direct that stream into a meetme conference?

You may ask why I don’t simply incorporate the SIP protocol. At this stage, I have the RTP streaming working and I need to incorporate the stream ASAP and I don’t have time to add SIP support. Since RTP is the basis for SIP audio, I figured there must be a way to use the RTP alone.

Thanks! Your help is much appreciated!

sadly, I don’t think this is possible. If you could have a device create a SIP INVITE for the rtp stream asterisk could take it, but if not then I think you’re out of luck until you work in a SIP stack…