WebRTC video with PJSIP gives black screen only

[yasiru]

Hi all,

I looking for features that I can improve confbridge application solutions and found WebRTC feature.
I started testing it using sipml5 and audio work well but video calls only get black screen.

Asterisk 16.2.1
Debian GNU/Linux 10 (buster)

My PJSIP config is like below,

---

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

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

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

[webrtc_client]
type=endpoint
aors=webrtc_client
auth=webrtc_client
dtls_auto_generate_cert=yes
webrtc=yes
context=execonf
disallow=all
allow=gsm,opus,ulaw,h264,h263p,vp8
max_audio_streams=10
max_video_streams=10
direct_media=nonat,update
ice_support=yes
tos_video=af41
cos_video=4

[webrtc_client2]
type=aor
max_contacts=5
remove_existing=yes

[webrtc_client2]
type=auth
auth_type=userpass
username=webrtc_client2
password=xxx

[webrtc_client2]
type=endpoint
aors=webrtc_client2
auth=webrtc_client2
dtls_auto_generate_cert=yes
webrtc=yes
context=execonf
disallow=all
allow=gsm,opus,ulaw,h264,h263p,vp8
max_audio_streams=10
max_video_streams=10
direct_media=nonat,update
ice_support=yes
tos_video=af41
cos_video=4

---

I use simple dialplan like below,

exten => _200,1,Ringing()
exten => _200,n,Wait(2)
exten => _200,n,Dial(PJSIP/webrtc_client2,60)
exten => _200,n,Hangup()

I’m testing this on AWS with allowing all traffic.

I’m kinda lost with what to check next. I also experiencing long time to connect the call.

Hope some experts can help me with this issue.

Thank you!

[david551]

The current 16 sub-version is 16.16.1.

If you are using WebRTC you need to be able to understand the logging, but you haven’t even provided any.

[jcolp]

I have a blog post of some of the common areas[1] but there is no magic bullet answer for WebRTC, and unless you are using a complete product it’s up to you to understand and learn everything involved, of which there is a lot, to support and debug things. It could be network, it could be encryption, it could be browser issues, it can be a lot of things.

[1] WebRTC and Asterisk: When It Goes Wrong ⋆ Open Source Communications Software | Asterisk Official Site