[asterisk-bugs] [JIRA] (ASTERISK-28986) video over audio is not switching in webrtc with asterisk 16

vineet singh (JIRA) noreply at issues.asterisk.org
Fri Jul 10 09:59:25 CDT 2020


    [ https://issues.asterisk.org/jira/browse/ASTERISK-28986?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=251401#comment-251401 ] 

vineet singh commented on ASTERISK-28986:
-----------------------------------------

yes i can't add video to audio call

i am getting this error in chrome browser


reinvite {originator: "local", type: "offer", sdp: "v=0
↵o=- 3343487241041739495 2 IN IP4 127.0.0.1
↵s…1438 label:009c00e5-6877-43b7-8bf2-b140c1bf737f
↵"}originator: "local"sdp: "v=0
↵o=- 3343487241041739495 2 IN IP4 127.0.0.1
↵s=-
↵t=0 0
↵a=group:BUNDLE 0
↵a=msid-semantic: WMS 3LceJzWmsXzSu8gxQD0S0L1Vpo7rAehjy3vW
↵m=audio 62862 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
↵c=IN IP4 49.36.143.23
↵a=rtcp:9 IN IP4 0.0.0.0
↵a=candidate:828268432 1 udp 2122260223 192.168.29.237 62862 typ host generation 0 network-id 1 network-cost 10
↵a=candidate:2145231712 1 tcp 1518280447 192.168.29.237 9 typ host tcptype active generation 0 network-id 1 network-cost 10
↵a=candidate:4258488323 1 udp 1686052607 49.36.143.23 62862 typ srflx raddr 192.168.29.237 rport 62862 generation 0 network-id 1 network-cost 10
↵a=ice-ufrag:1dla
↵a=ice-pwd:Co+iu7gYAHlDhW5/u9F13WWO
↵a=ice-options:trickle
↵a=fingerprint:sha-256 F8:63:DF:57:C1:F4:A5:55:0A:33:E1:28:0D:02:89:0E:A1:E0:7F:9B:66:0E:97:DA:0F:8D:86:1B:F1:7D:F1:91
↵a=setup:actpass
↵a=mid:0
↵a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
↵a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
↵a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
↵a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
↵a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
↵a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
↵a=sendrecv
↵a=msid:3LceJzWmsXzSu8gxQD0S0L1Vpo7rAehjy3vW 009c00e5-6877-43b7-8bf2-b140c1bf737f
↵a=rtcp-mux
↵a=rtpmap:111 opus/48000/2
↵a=rtcp-fb:111 transport-cc
↵a=fmtp:111 minptime=10;useinbandfec=1
↵a=rtpmap:103 ISAC/16000
↵a=rtpmap:104 ISAC/32000
↵a=rtpmap:9 G722/8000
↵a=rtpmap:0 PCMU/8000
↵a=rtpmap:8 PCMA/8000
↵a=rtpmap:106 CN/32000
↵a=rtpmap:105 CN/16000
↵a=rtpmap:13 CN/8000
↵a=rtpmap:110 telephone-event/48000
↵a=rtpmap:112 telephone-event/32000
↵a=rtpmap:113 telephone-event/16000
↵a=rtpmap:126 telephone-event/8000
↵a=ssrc:3542451438 cname:KzEfld+ptcjJ2F+M
↵a=ssrc:3542451438 msid:3LceJzWmsXzSu8gxQD0S0L1Vpo7rAehjy3vW 009c00e5-6877-43b7-8bf2-b140c1bf737f
↵a=ssrc:3542451438 mslabel:3LceJzWmsXzSu8gxQD0S0L1Vpo7rAehjy3vW
↵a=ssrc:3542451438 label:009c00e5-6877-43b7-8bf2-b140c1bf737f
↵"type: "offer"__proto__: Object
opr-call-handler.component.ts:711 reinvite 

> video over audio is not switching in webrtc with asterisk 16
> ------------------------------------------------------------
>
>                 Key: ASTERISK-28986
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28986
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Codecs/General
>    Affects Versions: 16.8.0
>         Environment: centos, asterisk 16, webrtc
>            Reporter: vineet singh
>            Assignee: vineet singh
>              Labels: webrtc
>
> 5557]
> type=endpoint
> aors=5557
> auth=5557-auth
> tos_audio=ef
> tos_video=af41
> ;videosupport=yes
> cos_audio=4
> ;canreinvite=no
> ;trustrpid=no
> ;nat=force_rport,comedia
> ;qualify=yes
> ;force_avp=yes
> cos_video=4
> disallow=all
> allow=alaw,ulaw,h264,vp8,g722,g729,gsm,mpeg4,h263,h261
> context=from-internal
> callerid=5557 <5557>
> dtmf_mode=rfc4733
> direct_media=no
> mailboxes=5557 at default
> mwi_subscribe_replaces_unsolicited=yes
> transport=0.0.0.0-ws
> aggregate_mwi=no
> use_avpf=yes
> rtcp_mux=yes
> max_audio_streams=1
> max_video_streams=1
> bundle=yes
> ice_support=yes
> media_use_received_transport=yes
> trust_id_inbound=yes
> user_eq_phone=no
> send_connected_line=yes
> media_encryption=sdes
> timers=yes
> webrtc=yes
> timers_min_se=90
> media_encryption_optimistic=yes
> refer_blind_progress=yes
> refer_blind_progress=yes
> send_pai=yes
> rtp_symmetric=yes
> rewrite_contact=yes
> force_rport=yes
> language=en
> one_touch_recording=on
> record_on_feature=apprecord
> record_off_feature=apprecord
> message_context=messages
> media_encryption=dtls
> dtls_verify=no
> dtls_setup=actpass
> dtls_rekey=0
> dtls_cert_file=/etc/asterisk/keys/voip1.operrtel.com.crt
> dtls_private_key=/etc/asterisk/keys/voip1.operrtel.com.key
> video call is going but audio to video switch is not happening.



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