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

[Henning Westerholt (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-30428?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=261552#comment-261552 ] 

Henning Westerholt commented on ASTERISK-30428:
-----------------------------------------------

We observed a similar problem:

- A (client behind SBC/SIP proxy) originates outbound call to PSTN gateway to B number
- A put party B on hold - client sends Inactive in SDP, Asterisk sends sendonly towards proxy
- A originates outbound call to PSTN gateway C party - call is answered - in call
- A choose to merge the calls from the client
- Client sends new INVITE with replaces header and instruct Asterisk to do call swap
- Asterisk do call swap so B is in call with C
- Asterisk hangup call with A as is supposed to do based on the INVITE with replaces header
- Asterisk doesn't send any re-INVITE towards PSTN gateway to release B party from hold
- Call between B and C stays in call until B or C party hangup but B is having MoH. C can hear B
- A party can leave call without interrupting call between B and C

Expected behaviour would be that A is in a call together with B and C. This was tested successfully with another PBX.

In case of eventual patches, just let me know, we can easily support with testing.

> 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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