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

[Friendly Automation (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26689?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=258495#comment-258495 ] 

Friendly Automation commented on ASTERISK-26689:
------------------------------------------------

Change 18071 merged by Friendly Automation:
res_pjsip_sdp_rtp: Improve detecting of lack of RTP activity

[https://gerrit.asterisk.org/c/asterisk/+/18071|https://gerrit.asterisk.org/c/asterisk/+/18071]

> res_pjsip_sdp_rtp: 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_sdp_rtp
>    Affects Versions: 13.7.0
>            Reporter: Dmitriy Serov
>            Assignee: Unassigned
>            Severity: Major
>         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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