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index d5d256ef9..31b521c90 100644
--- a/CHANGES
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@@ -181,6 +181,14 @@ chan_sip
NOTE: This is again separated by an exclamation mark, so the To: header may
not contain one of those.
+ * Session-Timers (RFC 4028) work for TCP (and TLS) transports as well now.
+ Previously Asterisk dropped calls only with UDP transports. However with
+ longer international calls via TCP, the SIP channel might break, because
+ all hops on the Internet route must stay online (have not a single power
+ outage, for example). Therefore with Session-Timers enabled (which are
+ enabled at default), you might see additional dropped calls. Consequently
+ please, consider to go for session-timers=refuse in your sip.conf.
+
chan_pjsip
------------------
* New 'user_eq_phone' endpoint setting. This adds a 'user=phone' parameter