[asterisk-bugs] [JIRA] (ASTERISK-24481) Asterisk PJSIP 16-18 second delay in reply to INVITE

[Anthony Messina (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24481?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=223224#comment-223224 ] 

Anthony Messina commented on ASTERISK-24481:
--------------------------------------------

I'll be darned, Josh.  I tried working with this most of the night with the following in /etc/asterisk/rtp.conf.  I haven't used STUN on my Asterisk server in years and must have left this in there.  After commenting out "icesupport" and "stunaddr", the delay is indeed gone.  I apologize for wasting your time on this Sunday morning.  
{code}
[general]
rtpstart=10000
rtpend=20000
icesupport=true
stunaddr=stun.counterpath.net
{code}


> Asterisk PJSIP 16-18 second delay in reply to INVITE
> ----------------------------------------------------
>
>                 Key: ASTERISK-24481
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24481
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip
>    Affects Versions: SVN
>         Environment: Fedora 20 x86_64, Asterisk 13: branches/13 at 426996
>            Reporter: Anthony Messina
>            Assignee: Anthony Messina
>         Attachments: asterisk-full.log
>
>
> After upgrading to Asterisk 13 release (had been previously using branches/13 at 425991), Inbound and outbound PJSIP calls take 18 seconds to initiate.  I will attach a redacted "full" debug log.  On a wall clock, for a call from a SIP phone endpoint connected directly to Asterisk, it's something like the following:
> {code}
> [00:00] INVITE (from phone, repeated several times due to Asterisk's non-timely reply)
> ...
> ...
> [00:18] Trying (from Asterisk)
> ... Call proceeds normally...
> {code}



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