[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)
noreply at issues.asterisk.org
Mon Dec 8 01:22:28 CST 2014
[ https://issues.asterisk.org/jira/browse/ASTERISK-24585?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Chris Wiltshire updated ASTERISK-24585:
---------------------------------------
Attachment: Asterisk_One_way_speech_native_rtp_to_simple_bridge.txt
CLI trace and debug.
> 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
>
>
> 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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