[asterisk-bugs] [JIRA] (ASTERISK-29618) ConfBridge errors on creation conference room

[George Joseph (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-29618?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

George Joseph reassigned ASTERISK-29618:
----------------------------------------

    Assignee: George Joseph  (was: Unassigned)

> ConfBridge errors on creation conference room
> ---------------------------------------------
>
>                 Key: ASTERISK-29618
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29618
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_confbridge, Resources/res_pjsip
>    Affects Versions: 16.20.0
>         Environment: CentOS Linux 7.4.170, kernel 3.10.0-1127.13.1.el7.x86_64, libsrtp 1.5.4
>            Reporter: Alexander
>            Assignee: George Joseph
>              Labels: webrtc
>         Attachments: aster_cli_16.20.0.log, aster_cli_16.9.0.log, debug_log_29618_16.20.0_bad_call.log, debug_log_29618_16.9.0_good_call.log, debug_log_29618_20210907.log
>
>
> After Asterisk upgrade from 16.9.0 to 16.20.0, we began to experience problems with the application when creating ConfBridge conference rooms for 2+ participants. Video streams for some participants do not appear.
> Connection schema: Asterisk - Kamailio - WebRTC Users
> {code:title=logs|borderStyle=solid}
> [Aug 25 11:11:53] ERROR[106619][C-00000001]: bridge_softmix.c:669 sfu_topologies_on_join:  CBAnn/conference_room_37197-00000000;2: Couldn't request topology change
> [Aug 25 11:12:08] ERROR[106435]: res_pjsip_session.c:2158 resolve_refresh_media_states:  State not consistent
> [Aug 25 11:12:08] WARNING[106435]: res_pjsip_session.c:2299 sip_session_refresh:  PJSIP/webrtc-00000000: Unable to merge media states
> {code}
> {code:title=extensions.conf|borderStyle=solid}
> [conference_room]
> exten => create,1,NoOp('Create conference room')
>  same => n,GoSub(add_member,1)
> exten => add_member,1,NoOp('Add member to conference room')
>  same => n,GoSub(configure_client,1)
>  same => n,GoSub(configure_bridge,1)
>  same => n,GoTo(join,1)
> exten => configure_bridge,1,NoOp('Configure bridge profile')
>  same => n,Set(CONFBRIDGE(bridge,template)=conference_members_bridge)
>  same => n,Return()
> exten => configure_client,1,NoOp('Configure client params')
>  same => n,Set(user_profile=client_profile)
>  same => n,Set(user_menu=client_menu)
>  same => n,Return()
> exten => join,1,NoOp(Join to ConfBridge: conf_name=${conf_name}, user_profile=${user_profile}, user_menu=${user_menu})
>  same => n,ConfBridge(${conf_name},,${user_profile},${user_menu})
>  same => n,Hangup()
> {code}
> {code:title=confbridge.conf|borderStyle=solid}
> [client_profile]
> type=user
> end_marked=yes
> jitterbuffer=no ; Jitterbuffer should be disabled when video is used.
> quiet=yes
> startmuted=no
> dsp_drop_silence=no
> talk_detection_events=no
> [conference_members_bridge]
> type=bridge
> video_mode=sfu
> max_members=15
> record_file_timestamp=no
> mixing_interval=10
> video_update_discard=1000
> remb_send_interval=1000
> remb_behavior=average_all
> {code}
> Steps:
> {code:title=ARI|borderStyle=solid}
> 1. Add first user to the conference:
> POST http://asterisk.node:8088/ari/channels
> {"endpoint": "PJSIP/81014504875 at webrtc", "callerId": "<8007709999>", "context": "conference_room", "extension": "create", "variables": {"conf_name": "conference_room_37368", "formats": "opus,vp8" ... }
> 2. Add other participants. For example:
> POST http://asterisk.node:8088/ari/channels
> {"endpoint": "PJSIP/81014504876 at webrtc", "callerId": "<8007709999>", "context": "conference_room", "extension": "add_member", "variables": {"conf_name": "conference_room_37368", "originator": "37368" ...}
> Where "originator": "37368" - first user's channelId
> {code}



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