Summary:	ASTERISK-28986: video over audio is not switching in webrtc with asterisk 16
Reporter:	vineet singh (singhvineet179@gmail.com)	Labels:	webrtc
Date Opened:	2020-07-09 15:29:18	Date Closed:	2020-07-10 10:06:19
Priority:	Major	Regression?
Status:	Closed/Complete	Components:	Codecs/General
Versions:	16.8.0	Frequency of Occurrence	Constant
Related Issues:
Environment:	centos, asterisk 16, webrtc	Attachments:
Description:	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@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.
Comments:	By: Asterisk Team (asteriskteam) 2020-07-09 15:29:20.453-0500 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur. By: Joshua C. Colp (jcolp) 2020-07-09 16:01:06.065-0500 What does "video call is going but audio to video switch is not happening." mean? If you mean you can't add video to an audio call, then Asterisk 16.8.0 does not have that functionality. You would need to upgrade to the latest version of Asterisk. If that is not what you mean then you will need to further expand and provide a SIP trace (pjsip set logger on) showing what is going on. By: vineet singh (singhvineet179@gmail.com) 2020-07-10 09:58:40.752-0500 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 By: Joshua C. Colp (jcolp) 2020-07-10 10:06:19.931-0500 As you have a community forum post going as well, please continue discussion there.