Merge "chan_sip: Enable Session-Timers for SIP over TCP (and TLS)."

1 files changed, 8 insertions, 0 deletions

diff --git a/CHANGES b/CHANGES
index d5d256ef9..31b521c90 100644
--- a/CHANGES
+++ b/CHANGES
@@ -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