[Asterisk-Dev] SIP 183 early media

[Mike Machado]

I am working on a project of integrating asterisk with our pstn gateway.
We are trying to get early media working. Right now a SIP user makes a
call, and asterisk starts a new dialog with the gateway. The gateway
sends a 183 to asterisk and the gateway starts transmitting RTP to
asterisk. Asterisk sends the 183 to the other dialog back to the SIP
user, and the SIP user also starts sending RTP to asterisk. Asterisk
however does not bridge the two RTP stream together until the gateway
sends a 200 to asterisk. Is there any way to get early media support in
asterisk so when it gets a 183 and also starts receiving RTP from both
sides, that it bridges it together? 

When the SIP user gets the 183 as described above, it stops providing
ring tone to the user, so they hear dead air. If asterisk bridged the
RTP together, the RTP coming from the gateway actually contains ringing
tone. I know the simple answer is to let the ATA generate ring for the
SIP user, but we have cases where its not just ring tone in the early
media. We also do not want to start billing until the 200 comes.

Any one have any comments on this?