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

[Michael Walton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-22745?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=211470#comment-211470 ] 

Michael Walton commented on ASTERISK-22745:
-------------------------------------------

This patch moves the setup of ICE session from ast_rtp_new to ast_rtp_prop_set (only if setting AST_RTP_PROPERTY_STUN to true). This doesn't significantly change order of execution, since both chan_sip and chan_pjsip call ast_rtp_instance_set_prop directly after successful return from ast_rtp_instance_new. I see chan_gtalk sets AST_RTP_PROPERTY_STUN to 1, so I have attached separately a patch for it that just makes sure the STUN property is set before RTCP property.
                
> 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

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