[asterisk-users] WebRTC using SIPML5 question

Dan Cropp dan at amtelco.com
Wed Dec 12 10:52:16 CST 2018

I had SIPML5 working with my Asterisk 16 last week.  Not sure what I changed, but I'm now receiving the following in asterisk whenever I try to login.

Can anyone provide some guidance on what I should be looking at or how to diagnose the problem?

[12/12 08:46:18.161] DEBUG[7322] http.c: HTTP opening session.  Top level
[12/12 08:46:18.161] DEBUG[7322] iostream.c: TLS non-recoverable I/O error occurred: error:00000005:lib(0):func(0):DH lib, System call EOF
[12/12 08:46:18.161] DEBUG[7322] http.c: HTTP closing session.  Top level

I followed the Asterisk wiki instructions for setting up asterisk for WebRTC.

enabled = yes
bindaddr =
bindport = 8088
tlsenable = yes
tlsbindaddr =
tlscertfile = /etc/asterisk/keys/asterisk.pem
tlsprivatekey = /etc/asterisk/keys/asterisk.pem

type = transport
bind =
protocol = wss

type = aor
max_contacts = 5
remove_existing = yes

type = auth
username = webrtc_client
password = webrtc_client

type = endpoint
context = IS
transport = transport2
auth = auth10
aors = webrtc_client
accountcode = 17
dtmf_mode = inband
device_state_busy_at = 2
disallow = all
allow = opus,ulaw
webrtc = yes
dtls_auto_generate_cert = yes

For the sipML5 demo, I have
Display Name: WebRTC Client
Private Identity: webrtc_client
Public Identity: sip:webrtc_client at
Password: webrtc_client
Realm: asterisk.org

In the sipML5 Expert settings I have
Disable Video checked
WebSocket Server URL: wss//
Disable 3GPP Early IMS: checked
Disable debug messages: checked
Cache the media stream: checked
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