[Freeswitch-dev] questions about making a new endpoint

[Seven Du]

Hi, 

I'm working on a mod_rtsp endpoint the idea is to stream sip video to a rtsp client such as VLC and also a sip client could "call" a rtsp camera later. I copied some codes from mod_sofia and it is too long to understand for me so I'd like to get some inputs either highlight me how to do or point out where to read from mod_sofia.

1) VLC "calls" in as a-leg and bridge to a sip client with video(or play_fsv), I made it sort of working now, but

-  How to disable the read_frame and read_video frame io_routine ? I set them to NULL and it fails on read, so I made a fake read function there, kind of ugly.

-  I hard coded to H264 and I can see video on VLC, I'd like to do like sip later negotiation then when b-leg sip client offers SDP and offer to a-leg when DESCRIBE

- looks like switch_io complains when no audio codec, is it possible to do video only calls? or is it possible to set audio and video rtp to a  different path ?

2) Call a rtsp camera would be easy later


3) Maybe a client can call in and record? I don't have such a client.

I would like to share the code later if others interested on this.


Thanks,
Seven.

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