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authorJoshua Colp <jcolp@digium.com>2017-06-13 09:18:18 -0500
committerGerrit Code Review <gerrit2@gerrit.digium.api>2017-06-13 09:18:18 -0500
commitbc51d4324a69a0b8ee4a3be208b91bb2081124ff (patch)
treef98acd14a7d5b0f710e32d9a9743aa02dc36e4e2 /CHANGES
parentd715beae165c5aa7c399441ec897dc9ebd6f4efa (diff)
parent996a4791ff123e80d71d44cb0fd13bb201d197b1 (diff)
Merge "pjsip: Extend 'asymmetric_rtp_codec' option to include us changing." into 13
Diffstat (limited to 'CHANGES')
-rw-r--r--CHANGES6
1 files changed, 6 insertions, 0 deletions
diff --git a/CHANGES b/CHANGES
index a951b2ca3..5cfffda15 100644
--- a/CHANGES
+++ b/CHANGES
@@ -34,6 +34,12 @@ 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 13.15.0 to Asterisk 13.16.0 ----------
------------------------------------------------------------------------------