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authorJenkins2 <jenkins2@gerrit.asterisk.org>2017-06-13 09:17:10 -0500
committerGerrit Code Review <gerrit2@gerrit.digium.api>2017-06-13 09:17:10 -0500
commitabe1dd303909416a1b57747bdb162b82fb08f72b (patch)
tree49af9daef02bbda2b535879d7a0c83efb34c583c /CHANGES
parent5d3420a2deb11773062ddff47473669dfcbde10a (diff)
parentd3e951edf5517b9f508a7e1b474176ec2be9e18f (diff)
Merge "pjsip: Extend 'asymmetric_rtp_codec' option to include us changing."
Diffstat (limited to 'CHANGES')
-rw-r--r--CHANGES6
1 files changed, 6 insertions, 0 deletions
diff --git a/CHANGES b/CHANGES
index 829086513..9bfa506d3 100644
--- a/CHANGES
+++ b/CHANGES
@@ -43,6 +43,12 @@ chan_pjsip
function any contact which is considered unreachable due to qualify being
enabled will no longer be called.
+ * The asymmetric_rtp_codec option now also controls whether chan_pjsip will
+ send media as-is without transcoding if the codec has been negotiated in the
+ SDP. If set to "no" then Asterisk will only ever send the preferred codec
+ from the SDP, unless the remote side sends a different codec and we will
+ switch to match.
+
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 14.4.0 to Asterisk 14.5.0 ------------
------------------------------------------------------------------------------