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

[Joshua C. Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-28774?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=249938#comment-249938 ] 

Joshua C. Colp commented on ASTERISK-28774:
-------------------------------------------

You can do it if you wish, or wait until it is reviewed and then do it. It goes into all supported applicable branches, which would be additionally 13, 17, and master.

> 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: Unassigned
>            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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