[asterisk-dev] RTP early media always sent: by design or not?

[Kirill Katsnelson]

The scenario is like this:
* A call is originated via AMI Originate, channel Local/s at local to s at remote
* s at local is executed, places a call to agent via a Queue.
* after it is connected, s at remote is executed, and calls a Dial().

I enable RTP debug to console, I see this right after the Dial sends an 
INVITE and we receive a 183 with an RTP endpoint (64.XXX.XXX.XXX is the 
remote end being dialed):

> Got  RTP packet from    64.XXX.XXX.XXX:19842 (type 00, seq 004261, ts 057920, len 000160)
> Sent RTP packet to      192.168.0.90:16396 (type 00, seq 025975, ts 057920, len 000160)
> Got  RTP packet from    192.168.0.90:16396 (type 00, seq 007229, ts 214375541, len 000160)
> Sent RTP packet to      64.XXX.XXX.XXX:19842 (type 00, seq 001482, ts 214375536, len 000160)

Even if I set prematuremedia=no for the Dial()'ed peer, RTP is still 
being sent.

Is that by design? Why should Asterisk send RTP stream to the remote end 
before receiving an ACK?

  -kkm