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(parent: ed690fc) | patch
Make an audio path under the following call configuration :
authorPhilippe Sultan <philippe.sultan@gmail.com>
Tue, 16 Oct 2007 09:47:22 +0000 (09:47 +0000)
committerPhilippe Sultan <philippe.sultan@gmail.com>
Tue, 16 Oct 2007 09:47:22 +0000 (09:47 +0000)
commit37a0b33171dbcc8b4bb05bf3fec9dd018d2bbb2e
tree42f290414f874fa482b96e48673f48ab4714dd1ftree | snapshot
parented690fc3489e97c95344773c178b7e8327a56a08commit | diff
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
channels/chan_jingle.c diff | blob | history
Mirror of the Asterisk Project (https://gerrit.asterisk.org)