Make an audio path under the following call configuration :

SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2

Modifications :
- set bridge type to partial ;
- process media candidates from the remote peer properly.

Now we have Jingle audio, at least between two Asterisk Jingle
clients.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85777 65c4cc65-6c06-0410-ace0-fbb531ad65f3