Audio stream is not playable on safari

Asterisk Asterisk WebRTC
1

I am using jssip for conference video calling, chrome and Firefox video and audio working fine on safari getting audio stream but it’s not playable

Asterisk 17.6.0
My configurations

[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0

[101]
type=aor
max_contacts=5
remove_existing=yes

[101]
type=auth
auth_type=userpass
username=101
password=password

[101]
type=endpoint
context=default
aors=101
auth=101
direct_media = no
allow = !all,ulaw,vp8,h264
dtls_auto_generate_cert=yes
webrtc=yes


confbridge.conf
[general]
 
[default_bridge]
type=bridge
video_mode=sfu
 
[default_user]
type=user
music_on_hold_when_empty=yes
music_on_hold_class=default

extensions.conf
[default]
exten => _1XX,1,Answer()
  same => n,ConfBridge(101)
  same => n,Hangup()
2

Safari is not a platform that has been really tested or developed for. It is entirely possible that there is an issue in Asterisk which prevents it from working, or there is some things you have to do to make it work within the HTML/Javascript itself.

3

i have tried different way to play audio stream, when i make audio conference stream working fine in chrome and safari but not able to play with video calling.
is there any settings or configuration like codec or user configuration which i missed in asterisk?

4

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