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

Benjamin Keith Ford (JIRA) noreply at issues.asterisk.org
Tue Aug 31 10:27:33 CDT 2021


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

Benjamin Keith Ford updated ASTERISK-29618:
-------------------------------------------

    Assignee: Alexander  (was: Unassigned)
      Status: Waiting for Feedback  (was: Triage)

Can you try a basic scenario with just Asterisk and client by a simpler setup or following the instructions on this wiki page [1] to see if the problem is reproduced there as well? This is just to eliminate possibilities of what might be causing the problem. It does look like there may be an issue with requesting a topology change; I'm curious to see if the same happens here.

> 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: Alexander
>              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
>
>
> 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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