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

[Etienne Allovon (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26143?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=236067#comment-236067 ] 

Etienne Allovon edited comment on ASTERISK-26143 at 3/22/17 11:36 AM:
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ASTERISK-26143-extensions_2.conf : with which tests were made
ASTERISK-26143-sip_2.conf : sip.conf with which tests were made
ASTERISK-26143-full-without-t-NOK : 1002 (sip/x2ces1) calls 111009 (sip/e0ujaob5)
ASTERISK-26143-full-with-t-OK : 1002 (sip/x2ces1) calls 111009 (sip/e0ujaob5) - with option t in Dial


was (Author: etienne_pf):
ASTERISK-26143-extensions.conf : with which tests were made
ASTERISK-26143-sip.conf : sip.conf with which tests were made
ASTERISK-26143-full-without-t-NOK : 1002 (sip/x2ces1) calls 111009 (sip/e0ujaob5)
ASTERISK-26143-full-with-t-OK : 1002 (sip/x2ces1) calls 111009 (sip/e0ujaob5) - with option t in Dial

> res_rtp_asterisk: 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, 13.11.0
>         Environment: Ubuntu 12.04 x86_64, Ubuntu 14.04 x86_64, Yocto 1.5 i686
>            Reporter: Henning Holtschneider
>            Assignee: Unassigned
>         Attachments: ASTERISK-26143-extensions_2.conf, ASTERISK-26143-extensions.conf, ASTERISK-26143-full-without-t-NOK, ASTERISK-26143-full-with-t-OK, ASTERISK-26143-sip_2.conf, ASTERISK-26143-sip.conf, call-g711-to-g722-ok.txt, call-g722-to-g711-unsupported-payload.txt
>
>
> 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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