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authorJoshua Colp <jcolp@digium.com>2012-07-16 12:35:04 +0000
committerJoshua Colp <jcolp@digium.com>2012-07-16 12:35:04 +0000
commite9387375700a1b3e78ce44af0495aefe29c6d2a4 (patch)
treeb46a8344ac5fbb5266e879a8e4409e28a737e327 /CHANGES
parentf9c3585d7312e65bda96755fbb8dab21fe54bcd5 (diff)
Add support for SIP over WebSocket.
This allows SIP traffic to be exchanged over a WebSocket connection which is useful for rtcweb. Review: https://reviewboard.asterisk.org/r/2008 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'CHANGES')
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1 files changed, 2 insertions, 0 deletions
diff --git a/CHANGES b/CHANGES
index db8a93fb8..32fef07e9 100644
--- a/CHANGES
+++ b/CHANGES
@@ -105,6 +105,8 @@ SIP Changes
* Add support for lightweight NAT keepalive. If enabled a blank packet will
be sent to the remote host at a given interval to keep the NAT mapping open.
This can be enabled using the keepalive configuration option.
+ * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
+ as the transport.
Chan_local changes
------------------