[Asterisk-Dev] channel module ast_rtp questions

[Alper Akçan]

hi

>Can you give us a diagram of what you are talking about here? It sounds 
>like the RTP stream is never going through Asterisk, but that there is only 
>one channel being created, which is bizarre...

here is what i am doing;

/* I put
struct ast_rtp *last_one == NULL; on the top of the rtp.c file.
last_one = rtp; just before the return of the ast_rtp_new_with_bindaddr()..
last_one = NULL; on the top of the ast_rtp_destroy();
*/

1. I call asterisk from IP phone (SIP)
2. asterisk sip channel creates a rtp struct for ip phone, and says 
"congradulations, ..." over rtp.
3. I dial the extension of my channel module, and asterisk calls 
module_call() function.
4. I get the ip telephones ip and the rtp port from last_one, and tell the 
device "start rtp connections with this ip:port". so bypassing the asterisk.

and what i want to do is;

1. create a rtp for the device via ast_rtp_new() when astersik calls
module_answer()
2. call ast_rtp_read() when module_read() function called. Get the rtp data
from the device and give it to asterisk.
3. call ast_rtp_write() when module_write() function called. get the rtp
data from asterisk and give it to the device.

best regards

alper.

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