[asterisk-bugs] [JIRA] (ASTERISK-26689) 183 Session in Progress. Disconnecting channel for lack of RTP activity

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26689?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=234422#comment-234422 ] 

Asterisk Team commented on ASTERISK-26689:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> 183 Session in Progress. Disconnecting channel for lack of RTP activity
> -----------------------------------------------------------------------
>
>                 Key: ASTERISK-26689
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26689
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip
>    Affects Versions: 13.7.0
>            Reporter: Dmitriy Serov
>         Attachments: 183-lack-rtp.txt
>
>
> Asterisk 13.7 (chan_pjsip) & Zyxel Keenetic Plus DECT, firmware v2.06(AAGJ.9)C1
> Outbound call from Zyxel Keenetic (pjsip endpoint) to PSTN (pjsip trunk).
> Call using early media (183 Session in progress) and rtp_timeout=10.
> After 10 seconds: [2016-12-16 13:53:15] NOTICE[6654] res_pjsip_sdp_rtp.c: Disconnecting channel 'PJSIP/xxx-0000027b' for lack of RTP activity in 10 seconds
> SIP dump is attached.
> According to [1] before called user agent send OK or ACK there is one way SDP.
> In sip dump (attached) i didn't find such SIP packets. Whether that call was canceled due to RTP inactivity? 



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