<html><body><div style="color:#000; background-color:#fff; font-family:times new roman, new york, times, serif;font-size:12pt"><div><span>Hi,</span></div><div><span><br></span></div><div><span>The difference is: Version 1.4 don't follow the SIP recommendation and version 1.8 does. <br></span></div><div><span><br></span></div><div><span>By the documment, after receiving a BYE, the dialog shall be destroyed after 32*T1 timer expires. </span></div><div><br></div><div>As I remmember, the recommendation doesn't say anything about SDP and UDP ports. In asterisk, this is only freed when the dialog ends.<br><span></span></div><div> </div><div>Leonardo<br></div><div style="font-family: times new roman,new york,times,serif; font-size: 12pt;"><div style="font-family: times new roman,new york,times,serif; font-size: 12pt;"><font face="Arial" size="2"><hr size="1"><b><span style="font-weight: bold;">De:</span></b> Kristijan Vrban
<vrban.lkml@googlemail.com><br><b><span style="font-weight: bold;">Para:</span></b> Asterisk Developers Mailing List <asterisk-dev@lists.digium.com><br><b><span style="font-weight: bold;">Enviadas:</span></b> Sexta-feira, 16 de Setembro de 2011 11:21<br><b><span style="font-weight: bold;">Assunto:</span></b> [asterisk-dev] After a BYE, chan_sip need 32s to destroy a SIP dialog. bug or feature?<br></font><br>Hello, i am doing stress testing 1.4 vs. 1.8 and the result is, that<br>in a varying<br>tests i have done, that (very very short summary) 1.8 need less CPU<br>then 1.4, also<br>1.8 does not have the 200 calls limit, were you get "ERROR[11264]:<br>utils.c:968 ast_carefulwrite: write() returned error: Broken pipe"<br>great! but when i was doing the stress tests, i run out of free udptl<br>ports because:<br><br>chan_sip in Asterisk 1.8 has the behavior, that i does not destroy the<br>SIP dialog immediately<br>after it get a BYE. So the UDP
Ports for RTP and T.38 stay open, for<br>32s until the "Really destroying SIP dialog<br>'006E36FB-DADE-E011-BB63-0024E820D141@127.0.0.1' Method: BYE" message<br>arrive. So after every SIP call two UDP ports are blocked for 32sec.<br>chan_sip in 1.4 immediately destroy the SIP dialog after a BYE, and<br>freed the used ports.<br><br>Is this a bug or a feature?<br><br>Kristijan<br><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br><br>asterisk-dev mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br><br><br></div></div></div></body></html>