[asterisk-bugs] [JIRA] (ASTERISK-30428) asterisk continues to play music on hold after INVITE with replaces

David Middleton (JIRA) noreply at issues.asterisk.org
Tue Feb 21 10:35:03 CST 2023


    [ https://issues.asterisk.org/jira/browse/ASTERISK-30428?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=261406#comment-261406 ] 

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

Should have added to the above:
When Party A merges the two calls
ingress: PJSIP/219-astpjsip-00000018 (replaces PJSIP/219-astpjsip-00000014)

PJSIP/proxy-00000015 is the channel that now has two RTP streams with the same SSRC; the music on hold continued from when PJSIP/219-astpjsip-00000014 put the call on hold plus the media from the replacement PJSIP/219-astpjsip-00000018, resulting in the called party hearing broken bits of both streams.

> asterisk continues to play music on hold 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_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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