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

[Joshua Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-22745?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=227161#comment-227161 ] 

Joshua Colp commented on ASTERISK-22745:
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Deferring it is called trickle ICE, which gathers candidates in the background and then trickles them in the negotiation process. Adding support for this requires API changes, chan_sip changes, and pjnath changes. It does not have trickle ICE support. This means that when you are going to send the SDP you need all the candidates.

As for why this hasn't gotten anywhere yet - everything these days goes through gerrit. Until such time as someone takes ownership and takes the patch through the gerrit process then this is waiting.

> 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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