>Im doing some research on a project to make a audio/video RTP stream >with a networkcam as video-source (delivers a MotionJPEG stream) and a >microphone as audio source. The stream has to be controlled by SIP >commands. >As far as I understood from the website and the mailing-lists it >should all be possible with the Live555 library's?
Not without some work, unfortunately, because we don't have any SIP server functionality in our libraries. However, if your client can use RTSP (rather than SIP), then you could do this using our existing RTSP server implementation. >And is it possible to do some audio editting (echo canceling, >equalizing) in Live555? Or should i do this outside Live555? We don't have any audio or video codec software, so this would need to be written separately (but perhaps encapsulated as a LIVE555 "FramedFilter"). -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ _______________________________________________ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel
