[asterisk-bugs] [JIRA] (ASTERISK-22745) chan_sip call setup very slow or fails when STUN server not available

[Steve Davies (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-22745?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=227164#comment-227164 ] 

Steve Davies commented on ASTERISK-22745:
-----------------------------------------

Thanks Josh - I really struggle with Gerrit, but will see what I can do. Perhaps I just need more practice :)

Sadly this change is near to code that has been re-factored between 11, 12 and 13 so it won't be a simple cherry-pick either.

I understand trickle ICE, and that Asterisk is not ready for that yet, but I thought a compromise might be to decouple the STUN/TURN setup from the channel+rtp creation code path so a Dial or similar will not block on repeated serialised failed STUN lookup attempts resulting in multiples of 9-second STUN timeouts before placing a call.


> chan_sip call setup very slow or fails when STUN server not available
> ---------------------------------------------------------------------
>
>                 Key: ASTERISK-22745
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22745
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General, Resources/res_rtp_asterisk
>    Affects Versions: 12.0.0-beta1
>         Environment: Ubuntu 10.04
>            Reporter: Michael Walton
>         Attachments: ASTERISK-22745-gtalk-stun.r402438.patch, ASTERISK-22745-sip-stun.r402438.patch
>
>
> Asterisk 12 compiled with chan_pjsip and chan_sip enabled. Call setup to or from chan_sip peer takes 10 seconds or more. To reproduce:
> * Enable icesupport in rtp.conf
> * Use an unreachable STUN server address for stunaddr, or disconnect WAN
> * Disable icesupport in sip.conf for a chan_sip peer that does not require STUN, e.g. local phone
> * Dial to or from phone



--
This message was sent by Atlassian JIRA
(v6.2#6252)