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

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26143?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=231129#comment-231129 ] 

Asterisk Team commented on ASTERISK-26143:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> 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: 13.7.2, 13.9.1
>         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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