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

[Igor Goncharovsky (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23765?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=222832#comment-222832 ] 

Igor Goncharovsky commented on ASTERISK-23765:
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If your phones works with rtp_method=1 with asterisk 11.x it MUST work with same setings on asterisk 12. Please try latest version, and attach console log with unistim debug enabled and tcpdump of unistim traf while call made.

> 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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