vIdeo call is going but audio to video switch is not happening. with webrtc and asterisk 16

Asterisk Asterisk WebRTC
1

pjsip file

[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. with webrtc and asterisk 16

2

At least at one time, the video stream needed to be included in the initial SDP.

3

So should i upgrade the asterisk

4

I don’t know if any version supports adding a video stream that didn’t exist at the start of the call. I only said “at least earlier versions” as I’m not fully up to speed in this area, and it is possible the feature has been added, although I doubt it.

5

It’s a semi-recent addition to PJSIP[1].

[1] https://blogs.asterisk.org/2020/02/19/adding-and-removing-media-streams/

6

getting this in chrome console logs

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

7

On the issue you filed as well you were using a version of Asterisk that did not support this. Did you upgrade and the problem is still experienced? Please use a pastebin for large logs.

8

i upgraded the asterisk and now its 17.5 but still i am not able to switch to video from audio call

9

image
image1600×900 199 KB

getting dom exception.

10

Your client software seems to have crashed. I wouldn’t expect that traceback even if adding a video stream had been refused.

11

but video conferencing is working

12

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