[asterisk-bugs] [JIRA] (ASTERISK-26143) One way audio when transcoding

[Henning Holtschneider (JIRA)]

Henning Holtschneider created ASTERISK-26143:
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             Summary: One way audio when transcoding
                 Key: ASTERISK-26143
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26143
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Resources/res_rtp_asterisk
    Affects Versions: 13.9.1, 13.7.2
         Environment: Ubuntu 12.04 x86_64, Ubuntu 14.04 x86_64, Yocto 1.5 i686
            Reporter: Henning Holtschneider


This is essentially the same issue as ASTERISK-25197, but that issue has been closed due to inactivity and I am not the original reporter.

I tried both Asterisk 13.7.2 and 13.9.1 on different machines with different Linux environments with the same result:

When making a call with a higher-quality codec to a destination with a lower-quality codec, e.g. G.722 to ALAW, Asterisk tries to set up a native bridge, fails to decode the lower-quality RTP coming from the called party and the line is silent at the caller's end.

Setting up the call with a lower-quality codec to a called party with a higher-quality codec works fine.

I tried with codecs ALAW, G.722 and G.729 all with the same result. I made calls between chan_sip peers and between chan_sip peers and PJSIP endpoints all with the same result.



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