[asterisk-bugs] [JIRA] (ASTERISK-27257) bridge_native_rtp: half-way direct media when using early bridging

Asterisk Team (JIRA) noreply at issues.asterisk.org
Wed Sep 6 10:59:08 CDT 2017


    [ https://issues.asterisk.org/jira/browse/ASTERISK-27257?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=238526#comment-238526 ] 

Asterisk Team commented on ASTERISK-27257:
------------------------------------------

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].

> bridge_native_rtp: half-way direct media when using early bridging
> ------------------------------------------------------------------
>
>                 Key: ASTERISK-27257
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27257
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Bridges/bridge_native_rtp
>    Affects Versions: 14.4.0
>            Reporter: Jean Aunis - Prescom
>
> When early bridging two SIP channels using ARI, SIP endpoint 1 does not receive endpoint 2's RTP port, but instead Asterisk's RTP port. Endpoint 2 properly receives endpoint 1's RTP port.
> This leads to half-way direct media : endpoint 2 sends its RTP stream directly to endpoint 1, but endpoint 1 sends its RTP stream to Asterisk which relays it to endpoint 2.
> To reproduce the problem, please find attached a piece of NodeJS code (for the ARI part), sip.conf and extensions.conf.



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