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authorJoshua Colp <jcolp@digium.com>2017-06-13 09:26:02 -0500
committerGerrit Code Review <gerrit2@gerrit.digium.api>2017-06-13 09:26:02 -0500
commit7994914c882874593eb17fdd8c910ee418f44a1b (patch)
treefc494f06178fd4e922f6afc27966533391e709a4 /CHANGES
parent9bf726d30e4ae5a48a5b315a405de5ce5b8b6ce1 (diff)
parenta34dd63eb0a961a901c7540fb02a8c8cbbf53024 (diff)
Merge "pjsip: Extend 'asymmetric_rtp_codec' option to include us changing." into 14
Diffstat (limited to 'CHANGES')
-rw-r--r--CHANGES6
1 files changed, 6 insertions, 0 deletions
diff --git a/CHANGES b/CHANGES
index 0429de445..d6f5a7500 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 14.4.0 to Asterisk 14.5.0 ------------
------------------------------------------------------------------------------