[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
Tue Jan 13 18:59:34 CST 2015


    [ https://issues.asterisk.org/jira/browse/ASTERISK-24585?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=224435#comment-224435 ] 

Chris Wiltshire commented on ASTERISK-24585:
--------------------------------------------

Just to ensure that I was clear previously:
- Call from external party arrives over IAX, presented directly to 707.
- 707 starts an attended Xfer and speaks to 706. - This is when the native rtp bridge functions, and this works fine.
- When 707 pushes the caller through to 706, this is when the speech path from 706 to the caller is never established. 
- Its when 706 transitions back from native rtp (with 707) to simple bridge (with caller) that the 706 speech path into that bridge (or the caller's listening path from the bridge) is not correctly established.

You've said it's likely to cause problems with native_rtp, however that portion between the two local SIP peers is working fine.
You've asked specifically for the config relating to 707, however the issue is more closely related to 706.

It's these two points which made me want to re-state the problem. Can you please let me know what portions of the configs you're specifically interested in. This is our production system so I'm likely to need to pull portions out as needed.

Thanks,

Chris.

> 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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