Webrtc Confbridge Audio working Video not

Hi all,
I’m trying to produce a working confbridge with video using Cyber Mega Phone 2K Ultimate Dynamic Edition and uphonex as client.
I’m running Asterisk 17.1.0 on a machine under nat.

Audio is working correctly both side but I can’t still see remote video.
I have route all rtp ports (Udp) in my firewall
my settigns are:
[pjsip.conf]

[transport-wss]
type = transport
protocol = wss
bind = 0.0.0.0
local_net=<my internal net>/24
external_media_address=<my public ip>
external_signaling_address=<my public ip>


[webrtc_client]
type=aor
max_contacts=5
remove_existing=yes

[webrtc_client]
type=auth
auth_type=userpass
username=webrtc_client
password=webrtc_client


[webrtc_client]
type=endpoint
aors=webrtc_client
auth=webrtc_client
dtls_auto_generate_cert=yes
max_audio_streams=10
max_video_streams=10
webrtc=yes
context=default
disallow=all
allow=opus
allow=ulaw

and the same for other endpoints
modules.conf

[modules]
autoload=yes
require = chan_pjsip.so
(..)
noload => chan_sip.so

rtp.conf

[general]
rtpstart=10000
rtpend=20000

[ice_host_candidates]
<my internal ip> => <my public ip>

running asterisk -rvvvvv
I can’t see any error in console
Both side view only their own videos, audio is working correctly.
Streamecho produce same resuts.

I’m an asterisk newbie

Thanks in advance,
Pier Paolo

What is the configuration for confbridge.conf? Have you set the correct video_mode? What is the output of “pjsip set logger on” with an attempt?

just figured out now.
I think it was a codec problem
putting

allow=opus,h264,vp8

i pjsip.conf works at least form chrome to chrome or firefox to firefox.

my confbrige.con si very simple for now:

[fancybridge]
type=bridge
max_members=20
mixing_interval=10
internal_sample_rate=auto
video_mode=sfu

[fancyuser-admin]
type=user
marked=yes
music_on_hold_when_empty=yes
dsp_drop_silence=yes


Yes, supported codecs can vary. VP8 is generally supported by everything.

thanks a lot for quick reply.

very exiting working with asterisk.

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