Hello asterisk-dev list,
I am working on a rtp proxy that essentially takes a mp4 video stream and converts it into a sip endpoint. To start I hacked up ekiga to use a text file with a gstreamer pipeline defined as a video and audio source, demuxing the video and audio and feeding it into the sip call. I then modified chan_rtp.c to send both the video and audio streams - which is currently working. I can use gstreamer to receive the udp streams and play back the audio and video. Now I want to get rid of ekiga and make chan_rtp also listen for an audio and video incoming udp stream to feed into the call. I have tried adding the source ports to the channel, but the sockets don't actually get opened and listen. Looking in the other channels I see where sockets are manually opened, but I would rather use the rtp engine. Could someone point me in the direction where a channel defines a rtp address/port using the ast_rtp_engine and opens the listening socket, or some guidance to at least identify the api calls to make that happen? I think I am close, but I am missing something. I have defined the video and audio channels, and can call into the extension and stream the call to my video wall. Do I need to define a separate pair of channels for receiving rtp, and what do I call once the local address is set so that the engine will actually start receiving the rtp data? I tried setting ast_rtp_instance_set_local_address on the channels I am sending on - that doesn't open the actual sockets. Any help would be greatly appreciated. Thank you. Michael
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