[asterisk-bugs] [JIRA] (ASTERISK-23765) RTP mishandling in chan_unistim

[Tamás Németh (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23765?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=222858#comment-222858 ] 

Tamás Németh commented on ASTERISK-23765:
-----------------------------------------

OK, it works well in 12.5. Do I still have to send the debug data?

> RTP mishandling in chan_unistim
> -------------------------------
>
>                 Key: ASTERISK-23765
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23765
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_unistim
>    Affects Versions: 12.2.0
>         Environment: Linux
>            Reporter: Tamás Németh
>            Assignee: Igor Goncharovsky
>
> In Asterisk 11.x I used my i2001 and i2002 phones with rtp_method=1 but in asterisk 12.x there is no incoming voice on my unistim phones unless the partner is also an unistim phone. So I changed to rtp_method=3, which makes calls mutually audible, but RTP streams are somehow asymmetric: tcpdumping on the asterisk server I can see the RTP stream coming
> from the unistim phone but no RTP stream goes from the server towards the phone! I assume that direction is directmedia or something like that. I tried call forwarding too, but it seems to be unable to handle this asymmetric half-direct, half -indirect RTP connection, and audio gets somehow confused and finally ceases to work.



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