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

[Richard Mudgett (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26593?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=235791#comment-235791 ] 

Richard Mudgett edited comment on ASTERISK-26593 at 3/14/17 11:59 AM:
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I don't think that this bug is related to ASTERISK-26666. This bug is about chan_sip, the other one about pjsip.
I have the same asymmetric codec issue using chan_sip.


was (Author: rvogt):
I don't think that this bug is related to BUG 26666. This bug is about chan_sip, the other one about pjsip.
I have the same asymmetric codec issue using chan_sip.

> 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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