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

[Tamás Németh (JIRA)]

Tamás Németh created ASTERISK-23765:
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             Summary: 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


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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