[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

[Chris Wiltshire (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24585?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Chris Wiltshire updated ASTERISK-24585:
---------------------------------------

    Attachment: issue_24585_full_log

Full debug and sip trace for failing call. Can you please take a look at this and if you need me to get into providing segments of my dialplan or other specific config files please let me know. - This was running a freshly compiled build from the very latest SVN checkout.

Asterisk SVN--r430470

> 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: Chris Wiltshire
>         Attachments: Asterisk_One_way_speech_native_rtp_to_simple_bridge.txt, Asterisk_One_way_speech_native_rtp_to_simple_bridge.txt, issue_24585_full_log
>
>
> 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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