[asterisk-bugs] [JIRA] (ASTERISK-26593) chan_sip: One way audio due to RTP bridging when it shouldn't

[Luke Escude (JIRA)]

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

Luke Escude updated ASTERISK-26593:
-----------------------------------

    Attachment: flowroute-280984.pcap02

What do you make of this capture? iLBC between phone and server, then uLaw from server to Flowroute trunk. One-way audio, callee couldn't hear the caller (ext. 221)

> chan_sip: One way audio due to RTP bridging when it shouldn't
> -------------------------------------------------------------
>
>                 Key: ASTERISK-26593
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26593
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/CodecHandling
>    Affects Versions: 13.12.2
>         Environment: CentOS x64
>            Reporter: Luke Escude
>            Assignee: Unassigned
>            Severity: Minor
>         Attachments: cli_case1.txt, console_log.txt, flowroute-280984.pcap02, ilbc_case1.pcap, messages, 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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