I ran in the same issues as John Todd did some while ago:
http://lists.digium.com/pipermail/asterisk-users/2005-November/129541.htmlI use qualify=yes to ping our internal SIP proxies for failover and therefore I have very low delays, e.g.
Name/username Host Dyn Nat ACL Port Status mid2-3 xx.xx.xx.xx 5060 OK (1 ms) which causes Asterisk to use a very small T1 to retransmit SIP requests: Time Protocol Info 4.107899 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED] 4.113318 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED] 4.113339 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED] 4.121283 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED] 4.129283 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED] 4.145284 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED] 4.177281 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED] (this looks like a SIP DOS attack to me)Setting T1 according the SIP qualify delay only makes sense if the delay measurements are done with the final target of a SIP request. If I ping a SIP proxy instead, the ping delay does not say anything about the actual end-to-end SIP signaling path delay.
My recommendation would be to statically set T1 to 500ms according to RFC 3261. If that is not an option I'd set a minimum T1 that is at least 100ms.
-Christian
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