[asterisk-bugs] [JIRA] (ASTERISK-26593) bridge_native_rtp: Switching to native rtp disables RTCP reporting

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-26593?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-26593:
------------------------------------

    Assignee: Luke Escude  (was: Unassigned)
      Status: Waiting for Feedback  (was: Triage)

bq. CLI output with SIP debugging attached. I'll post up dial plan configuration next.

Actually, you didn't include the "debug" log channel. Can you attach a new log following these instructions:

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

It should include the logger channels: warning,notice,error,verbose,debug

Important notes:

 * Make sure the log has verbose and debug levels both turned up to *5*
 * Make sure the log includes the SIP packet trace
 * Make sure the log includes the output of "rtcp set debug on"
 * Attach the log in *plain* text format. Do not attach it in rtf or anything fancy. Attach it with a .txt extension for accessibility.
 * Gather the log as the instructions indicate and do not copy output from the Asterisk console. That is, please use the logger output to a file in /var/log/asterisk/

Thanks!

> bridge_native_rtp: Switching to native rtp disables RTCP reporting
> ------------------------------------------------------------------
>
>                 Key: ASTERISK-26593
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26593
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Bridges/bridge_native_rtp
>    Affects Versions: 13.12.2
>         Environment: CentOS x64
>            Reporter: Luke Escude
>            Assignee: Luke Escude
>            Severity: Minor
>         Attachments: console_log.txt, sip_debug.rtf
>
>
> As soon as the call switches from simple_bridge to native_rtp, the server stops receiving RTCP information, and the call will drop because it thinks there's inactivity. We've had to set the rtp timeout to 24 hours to compensate for this.
> Log readout during a phone call:
> [Edit by Rusty - moved debug to console_log.txt]



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