[Live-devel] SIP and RTP with network cam

[Ross Finlayson]

>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/