[sr-dev] [tracker] Comment added: Dialog never times-out when ka_timer < 10, even on 481 or 408(fake) response

sip-router bugtracker at sip-router.org
Thu Apr 17 14:47:24 CEST 2014


THIS IS AN AUTOMATED MESSAGE, DO NOT REPLY.

The following task has a new comment added:

FS#417 - Dialog never times-out when ka_timer < 10, even on 481 or 408(fake) response
User who did this - Daniel-Constantin Mierla (miconda)

----------
Detecting interrupted network connections is not for normal operation cases. Beware that in mobile networks are bid delays, you may get false-positive cases if you have lower timeout. If you look at the option of detecting inactivity via RTP, all applications capable of doing that by handing RTP (e.g., Asterisk, Freeswitch) allow 90sec or more.

Now, you can contribute various enhancements on this part, but have in mind that the main role is to be able to handle lots of calls, most of them being ok. You may get 0.1% calls interrupted due to network. If you make a very complex detection system for the 0.1%, then you affect the performances for the other 99.9%. Try to imagine you have 10000active calls, you send keepalives every second, that is 20000 OPTIONS transactions every second. Perhaps many phones will get stuck. Therefore, such contribution must include an option to turn on/off (module parameter).
----------

More information can be found at the following URL:
http://sip-router.org/tracker/index.php?do=details&task_id=417#comment1393

You are receiving this message because you have requested it from the Flyspray bugtracking system.  If you did not expect this message or don't want to receive mails in future, you can change your notification settings at the URL shown above.



More information about the sr-dev mailing list