[asterisk-users] How to integrate thirdparty RTP with Asterisk

[garge rama]

Hi,

I'm working on Asterisk and would like to use only Asterisk SIP signalling
for my Voip application.

I have written my own channel driver and want to integrate my own RTP with
Asterisk.



SIP signalling is working fine. But i could not find API's to get RTP Port
and IP address to start

without starting rtp session.



The only way I found to receive/send rtp information is by creating a rtp
session

from channel driver using ast_rtp_new_with_bindaddr( ); which means using
asterisk rtp stack.



This is not what i want.


If anyone has done RTP integration with Asterisk earlier please help me.


Thanks in advance.
Garge.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100708/f457a84f/attachment.htm