[asterisk-bugs] [JIRA] (ASTERISK-30428) bridging: Music on hold continues after INVITE with replaces

[David Middleton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-30428?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=261625#comment-261625 ] 

David Middleton commented on ASTERISK-30428:
--------------------------------------------

Good morning Henning.
I'm happy to say that your patch has also fixed our issue.
As soon as the INVITE with replaces was processed, the held party stops receiving moh audio and starts to receive the bridged audio from the other call leg (and there was just a single RTP stream.)
Awesome, thanks.

(I don't suppose you fancy looking at ASTERISK-30454 - I suspect that one's not quite as straightforward!)

> bridging: Music on hold continues after INVITE with replaces
> ------------------------------------------------------------
>
>                 Key: ASTERISK-30428
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-30428
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Bridges/bridge_native_rtp, Bridges/bridge_simple, Bridges/bridge_softmix
>    Affects Versions: 18.14.0
>            Reporter: David Middleton
>            Assignee: Unassigned
>         Attachments: 20230130_teams-pstn_teams-pstn_merge_pjsip_nok_asterisk_anon.pcap, 20230214 pcap breakdown.txt, 2023022000_debug_log_ASTERISK-30428.txt, 2023022100_debug_log_ASTERISK-30428.txt
>
>
> Party A calls Party B.
> Party A puts Party B on hold and Asterisk plays moh to Party B.
> Party A then calls Party C.
> Party A then merges the two calls (to Party B and Party C) - a new INVITE is received by asterisk replacing the original held call to Party B.
> Asterisk continues to send music on hold to Party B, as well as the media from the new replacement call, all using the same SSRC and UDP ports, but with different timestamps and sequence numbers (causing Party B to hear broken moh and audio). Media from party B is ok, as well as media to/from Parties A and C.
> I've attached an anonymised pcap that shows all the SIP signalling, but for simplification, just the egress RTP (to/from Party B and C).
> I've also attached a pcap breakdown text file.
> I did try unloading 'bridge_simple' as suggested in ASTERISK-29273 but the issue remained.



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