Jitsi-meet getGlobalOnErrorHandler on Firefox


#1

Hi team,

I’m testing with version 3135 of Jitsi, Jicofo and JVB on Firefox 61. When 3 users join the conference, sometimes, I got this error: Error: container of type undefined doesn’t exist. (I attack in this below image). This happened when someone click refresh and only excuse when all user use Firefox 59 or 61.


#2

Hi, this error happened when someone who doesn’t have both camera and microphone join the conference.
This is the log:

[modules/RTC/TraceablePeerConnection.js] : createAnswerOnError Error: “Cannot convert to Unified Plan because m-lines that are not bundled were found.”
toUnifiedPlanhttps://vcrxdev4.topica.vn/libs/lib-jitsi-meet.min.js?v=139:2:987582setLocalDescriptionhttps://vcrxdev4.topica.vn/libs/lib-jitsi-meet.min.js?v=139:2:959494valuehttps://vcrxdev4.topica.vn/libs/lib-jitsi-meet.min.js?v=139:2:759331ahttps://vcrxdev4.topica.vn/libs/lib-jitsi-meet.min.js?v=139:2:968351 type: answer

v=0

o=mozilla…THIS_IS_SDPARTA-61.0.1 7420598274005848892 2 IN IP4 0.0.0.0

s=-

t=0 0

a=fingerprint:sha-256 9A:51:EF:77:09:46:39:DD:D2:E6:86:C1:42:1D:5E:3E:0B:F7:D5:D3:8F:8D:BA:7B:57:17:FF:3B:10:DE:17:75

a=ice-options:trickle

a=msid-semantic: WMS *

a=group:BUNDLE audio data

m=audio 7318 RTP/SAVPF 111 126

c=IN IP4 1.55.145.30

a=rtpmap:111 opus/48000/2

a=rtpmap:126 telephone-event/8000/1

a=fmtp:111 maxplaybackrate=48000;stereo=1;useinbandfec=1

a=fmtp:126 0-15

a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level

a=setup:active

a=mid:audio

a=recvonly

a=ice-ufrag:9d6e31d5

a=ice-pwd:d018c779fc88b45ed6a56e1b190238b3

a=candidate:0 1 UDP 2122252543 192.168.20.74 61980 typ host

a=candidate:1 1 TCP 2105524479 192.168.20.74 9 typ host tcptype active

a=ssrc:2529979872 cname:{849d040a-c21f-4d92-b6f6-2fa1e0c72cd4}

a=ssrc:3721767689 cname:{849d040a-c21f-4d92-b6f6-2fa1e0c72cd4}

a=ssrc:4165435615 cname:{849d040a-c21f-4d92-b6f6-2fa1e0c72cd4}

a=rtcp-mux

m=video 9 RTP/SAVPF 100

c=IN IP4 0.0.0.0

a=rtpmap:100 VP8/90000

a=fmtp:100 max-fs=12288;max-fr=60

a=rtcp-fb:100 nack

a=rtcp-fb:100 nack pli

a=rtcp-fb:100 ccm fir

a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time

a=setup:active

a=mid:video

a=inactive

a=ice-ufrag:9d6e31d5

a=ice-pwd:d018c779fc88b45ed6a56e1b190238b3

a=ssrc:3125150237 cname:{849d040a-c21f-4d92-b6f6-2fa1e0c72cd4}

a=ssrc:3174272565 cname:{849d040a-c21f-4d92-b6f6-2fa1e0c72cd4}

a=rtcp-mux

m=application 9 DTLS/SCTP 5000

c=IN IP4 0.0.0.0

a=setup:active

a=mid:data

a=sendrecv

a=ice-ufrag:9d6e31d5

a=ice-pwd:d018c779fc88b45ed6a56e1b190238b3

a=rtcp-mux

a=sctpmap:5000 webrtc-datachannel 256

a=max-message-size:1073741823

Logger.js:124:12
[modules/xmpp/JingleSessionPC.js] <value/</<>: addRemoteStream failed: createAnswer failed: Error: Cannot convert to Unified Plan because m-lines that are not bundled were found.