[asterisk-bugs] [JIRA] (ASTERISK-27604) chan_sip: turning on srtp results in one way audio in Asterisk 15

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27604?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=242009#comment-242009 ] 

Asterisk Team commented on ASTERISK-27604:
------------------------------------------

Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1].

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

> chan_sip: turning on srtp results in one way audio in Asterisk 15
> -----------------------------------------------------------------
>
>                 Key: ASTERISK-27604
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27604
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/SRTP
>    Affects Versions: 15.1.5, 15.2.0
>         Environment: Debian 9, x86-64
>            Reporter: Martin Dindos
>            Assignee: Martin Dindos
>
> I've set up a test install of Asterisk 15 on my 64 bit x86 machine. All phones were on local network (no NAT). All extensions were registered using the old chan_sip. With media encryption disabled on all extensions everything worked as expected.
> After turning on media encryption (srtp) the calls resulted in one way-audio. The audio worked in the direction 
> (phone 1 - using srtp -> phone 2 srtp disabled) the reverse direction was not working.
> I've downloaded and compiled Asterisk 13 on the same machine with the same configuration. Everything started to work no audio issues regardless of whether srtp was enabled or not.



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