[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:comment-tabpanel&focusedCommentId=233704#comment-233704 ] 

Rusty Newton edited comment on ASTERISK-26593 at 11/14/16 4:42 PM:
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The logs we need include SIP and debug level. Native RTP bridge can actually mean 2 different things. Either a local optimized one, or a remote bridge in which case media and RTCP would not be going through Asterisk. It's unclear which exactly is being done. As well the configuration would allow us to reproduce it, and different configuration influences the results of the bridge and the choice. Finally we reverted a change in Asterisk 13.12.2 which caused rtptimeout to occur when it shouldn't. Thus my question about confirmation of version.


was (Author: jcolp):
The logs we need include SIP and debug level. Native RTP bridge can actually mean 2 different things. Either a local optimized one, or a remote bridge in which case media and RTCP would not be going through Asterisk. It's unclear which exactly is being done. As well the configuration would allow us to reproduce it, and different configuration influences the results of the bridge and the choice. Finally we reverted a change in Asterisk 13.2.2 which caused rtptimeout to occur when it shouldn't. Thus my question about confirmation of version.

> 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: Unassigned
>            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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