From 9be69c163632e485934ce9ca697babc10c6ea896 Mon Sep 17 00:00:00 2001 From: Alexander Traud Date: Tue, 19 Jul 2016 13:16:02 +0200 Subject: chan_sip: Enable Session-Timers for SIP over TCP (and TLS). Asterisk defaults to timers=accept/refresher=uas. In that scenario, only in that scenario, Sessions-Timers (RFC 4028) had no effect via TCP. This change enables Session-Timers for SIP over TCP (and for SIP over TLS). 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 dropped calls. Consequently even with this change, you might be better-off going for session-timers=refuse in your sip.conf. ASTERISK-19968 #close Change-Id: I1cd33453c77c56c8e1394cd60a6f17bb61c1d957 --- CHANGES | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'CHANGES') diff --git a/CHANGES b/CHANGES index 9caa52422..6d7cb4c42 100644 --- a/CHANGES +++ b/CHANGES @@ -177,6 +177,14 @@ 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 -- cgit v1.2.3