Running a test using asterisk 16.1.1 and two PCs with Firefox browsers. I'm running the cmp2k demo.
I place calls into the same asterisk and using AMI answer the calls and then add them into the same confbridge. Video mode is configured to follow_talker. However, the Remote Video displayed to both browsers is always the video of the opposite call. It's not following whoever talked last. I have the talk_detection_events enabled and looking at the event it seems asterisk isn't always detecting the talking correctly. Next question.... If asterisk is reporting channel PJSIP/webrtc_client1-0000000e is talking, shouldn't the confbridge follow_talker setting change this confbridge VideoSource to be the Uniqueid of this channel? Event: ConfbridgeTalking^M Privilege: call,all^M Conference: Bridge2^M BridgeUniqueid: ea6e7cb6-cc86-413e-885c-b0a25be887ee^M BridgeType: base^M BridgeTechnology: softmix^M BridgeCreator: ConfBridge^M BridgeName: Bridge2^M BridgeNumChannels: 3^M BridgeVideoSourceMode: talker^M BridgeVideoSource: 1552492132.33^M Channel: PJSIP/webrtc_client1-0000000e^M ChannelState: 6^M ChannelStateDesc: Up^M CallerIDNum: webrtc_client1^M CallerIDName: <unknown>^M ConnectedLineNum: <unknown>^M ConnectedLineName: <unknown>^M Language: en^M AccountCode: 19^M Context: ABC^M Exten: 55555^M Priority: 14^M Uniqueid: 1552492117.32^M Linkedid: 1552492117.32^M TalkingStatus: on^M Admin: No^M Last question.... I see times where it seems the talk detection seems to become stuck for a channel. I see this event happening and the confbridge videosource becomes this channel's uniqueid. I have not talked into this PC/browser's mic in hours. Literally went to lunch and it seems stuck in the TalkingStatus: on state. I have hundreds of ConfbridgeTalking events for the other channel (on and off) over the next several hours, but that channel's TalkingStatus seems stuck. [03/13 10:49:00.595] DEBUG[49360] manager.c: Examining AMI event: Event: ConfbridgeTalking^M Privilege: call,all^M Conference: OpBridge2^M BridgeUniqueid: ea6e7cb6-cc86-413e-885c-b0a25be887ee^M BridgeType: base^M BridgeTechnology: softmix^M BridgeCreator: ConfBridge^M BridgeName: OpBridge2^M BridgeNumChannels: 2^M BridgeVideoSourceMode: talker^M BridgeVideoSource: 1552492132.33^M Channel: PJSIP/webrtc_client1-0000000f^M ChannelState: 6^M ChannelStateDesc: Up^M CallerIDNum: webrtc_client1^M CallerIDName: <unknown>^M ConnectedLineNum: <unknown>^M ConnectedLineName: <unknown>^M Language: en^M AccountCode: 19^M Context: ABC^M Exten: 4444^M Priority: 14^M Uniqueid: 1552492132.33^M Linkedid: 1552492132.33^M TalkingStatus: on^M Admin: No^M The templates I'm using are Action: SetVar ActionID: C173 Channel: PJSIP/webrtc_client1-0000000f Variable: CONFBRIDGE(bridge,template) Value: 2 Action: SetVar ActionID: C174 Channel: PJSIP/webrtc_client1-0000000f Variable: CONFBRIDGE(user,template) Value: 4 Action: SetVar ActionID: C176 Channel: PJSIP/webrtc_client1-0000000e Variable: CONFBRIDGE(bridge,template) Value: 2 Action: SetVar ActionID: C177 Channel: PJSIP/webrtc_client1-0000000e Variable: CONFBRIDGE(user,template) Value: 4 [2] type = bridge language = en internal_sample_rate = 0 mixing_interval = 20 record_file_append = no max_members = 10 video_mode = follow_talker [4] type = user admin = no marked = no startmuted = no music_on_hold_when_empty = no quiet = yes wait_marked = no end_marked = no dsp_drop_silence = yes dsp_silence_threshold = 2500 dsp_talking_threshold = 160 denoise = no jitterbuffer = yes talk_detection_events = yes dtmf_passthrough = no announce_user_count = no announce_join_leave = no announce_user_count_all = no announce_only_user = no send_events = no echo_events = no announce_join_leave_review = no
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