[asterisk-bugs] [JIRA] (ASTERISK-26143) res_rtp_asterisk: One way audio when transcoding
Etienne Allovon (JIRA)
noreply at issues.asterisk.org
Wed Mar 22 11:38:10 CDT 2017
[ https://issues.asterisk.org/jira/browse/ASTERISK-26143?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Etienne Allovon updated ASTERISK-26143:
---------------------------------------
Attachment: ASTERISK-26143-sip_2.conf
ASTERISK-26143-extensions_2.conf
> 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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