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: 192.168.10.5. The source of the RTP comes from 192.168.10.6 and is streaming to asterisk on port 5062. Can asterisk take the stream, seeing that it is coming from ip: 192.168.10.6 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!