[asterisk-bugs] [JIRA] (ASTERISK-24585) One way speech when performing attended Xfer issue appears to be SIP invite related when switching back from native_rtp bridge

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24585?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=223949#comment-223949 ] 

Rusty Newton commented on ASTERISK-24585:
-----------------------------------------

Due to the issues Richard mentioned, you'll need to re-attach your debug. However I was able to look at it in the history tab and it does not include the "DEBUG" logger channel type.

Here are some instructions to help you get the right settings turned on: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

Once you have CLI logs with the DEBUG logger channel turned on (level 5 or above please and showing in the log) and SIP trace integrated then attach it to the issue (not as a contribution, so it'll show up).

> One way speech when performing attended Xfer issue appears to be SIP invite related when switching back from native_rtp bridge
> ------------------------------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-24585
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24585
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Bridges/bridge_native_rtp, Bridges/bridge_simple
>    Affects Versions: 13.0.1
>         Environment: Server: Virtual image, VSphere 5.5, Guest OS: Ubuntu 14.10.
> Network: Simple LAN behind NAT'ing firewall.
> VoIP Service: External, IAX trunks providing external connectivity, registrations passing out through stateful firewall, no pinholing.
> Internal client devices: Linksys SPA942 SIP phones.
>            Reporter: Chris Wiltshire
>            Assignee: Rusty Newton
>         Attachments: Asterisk_One_way_speech_native_rtp_to_simple_bridge.txt, Asterisk_One_way_speech_native_rtp_to_simple_bridge.txt
>
>
> Outlined in additional detail in forum thread:
> http://forums.asterisk.org/viewtopic.php?f=1&t=91945&p=204700#p204700
> One way speech occurs after attended transfer of inward call.
> Parties: Caller, Party A, Party B.
> Caller calls in via IAX trunk and passes to Party A. Party A then performs an attended transfer and speaks to Party B. During this Caller hears music on hold. After introduction Party A completes attended transfer and connects Caller to Party B. At this point one way speech occurs, Party B can hear the Caller, but the Caller cannot hear Party B.
> Work-around: suspend bridge technology native_rtp.
> Hypothosis: When the two local parties talk during the attended transfer the bridging mode is switched to native_rtp. The IAX Caller channel is remote and cannot support rtp, so a switch back from native_rtp bridge mode to simple is attempted. During this switch back, it appears that there may be an issue with SIP commands issued?.
> Investigations: 
> - A straight forward non-attended transfer does not bring about this issues.
> - An attended transfer (exactly the same usecase) with native_rtp suspended does not bring about this issue.
> Our experience in the forum was helpful with SIP debug and further tracing being performed. It was then that we were encouraged to log an issue here.
> I have CLI trace for both active and suspended native_rtp test cases. I will upload them to this issue thread shortly (as attachments).



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