[asterisk-bugs] [JIRA] (ASTERISK-28774) chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge

[Michael Neuhauser (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-28774?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=249937#comment-249937 ] 

Michael Neuhauser commented on ASTERISK-28774:
----------------------------------------------

The review is up.  I'm not sure about "cherry-picking" (who does it, which branches, at which point in the work flow) - so only branch 16 for now. 

> chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge
> ----------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-28774
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28774
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip, Resources/res_pjsip_sdp_rtp
>    Affects Versions: 16.8.0
>            Reporter: Michael Neuhauser
>            Assignee: Michael Neuhauser
>            Severity: Minor
>
> 2 PJSIP endpoints with identical configuration (codec, etc.) and
> direct_media=yes
> rtp_timeout=10
> rtp_timeout_hold=10
> When those two endpoints are bridged (via simple Dial()) the RTP is flowing directly between them, not through Asterisk. But the code that checks for a RTP timeout is still active and erroneously terminates the connection after same time.
> This happens because the function rtp_check_timeout() in res/res_pjsip_sdp_rtp.c ignores the direct-media state of the endpoint (can be checked via session_media->direct_media_addr).
> I have a small patch that fixes this bug and will add a gerrit code review for it.



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