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| author | Joshua Colp <jcolp@digium.com> | 2009-04-02 17:20:52 +0000 |
|---|---|---|
| committer | Joshua Colp <jcolp@digium.com> | 2009-04-02 17:20:52 +0000 |
| commit | 63de8343958b91c8836c5e6ddf1c0106b40e9fe6 (patch) | |
| tree | 8a8042738e1c444e5988a648b795c4d2b02febd1 /UPGRADE.txt | |
| parent | 08971ce2056f4e035b4b37324c7f184370cd0ec6 (diff) | |
Merge in the RTP engine API.
This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.
Review: http://reviewboard.digium.com/r/209/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'UPGRADE.txt')
| -rw-r--r-- | UPGRADE.txt | 6 |
1 files changed, 5 insertions, 1 deletions
diff --git a/UPGRADE.txt b/UPGRADE.txt index 62551b0f9..35b0d455a 100644 --- a/UPGRADE.txt +++ b/UPGRADE.txt @@ -20,7 +20,11 @@ From 1.6.2 to 1.6.3: -* Nothing, yet! +* The usage of RTP inside of Asterisk has now become modularized. This means + the Asterisk RTP stack now exists as a loadable module, res_rtp_asterisk. + If you are not using autoload=yes in modules.conf you will need to ensure + it is set to load. If not, then any module which uses RTP (such as chan_sip) + will not be able to send or receive calls. From 1.6.1 to 1.6.2: |