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

[Luke Escude (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26593?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233776#comment-233776 ] 

Luke Escude commented on ASTERISK-26593:
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Joshua, it seems as though the RTCP bug has fixed itself after upgrading to 13.12.2. However, the customer is now complaining about occasional one-way audio calls.

The log is showing a whole lot of this feedback:

[Nov 16 21:20:23] DEBUG[8605][C-00001a35] res_rtp_asterisk.c: Unsupported payload type received
[Nov 16 21:20:23] DEBUG[8604][C-00001a34] translate.c: Sample size different 160 vs 240
[Nov 16 21:20:23] DEBUG[8604][C-00001a34] translate.c: Sample size different 160 vs 240
[Nov 16 21:20:23] DEBUG[8605][C-00001a35] res_rtp_asterisk.c: Unsupported payload type received
[Nov 16 21:20:23] DEBUG[8605][C-00001a35] res_rtp_asterisk.c: Unsupported payload type received
[Nov 16 21:20:23] DEBUG[8604][C-00001a34] translate.c: Sample size different 160 vs 240
[Nov 16 21:20:23] DEBUG[8605][C-00001a35] res_rtp_asterisk.c: Unsupported payload type received

The handsets are forcing ilbc:30, and the trunk is forcing uLaw. Is there a correlation between these errors and the one-way audio?

> 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, rtcp_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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