WebRTC Using SIPML5 Question

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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.

http.conf
[general]
enabled = yes bindaddr = 0.0.0.0
bindport = 8088
tlsenable = yes tlsbindaddr = 0.0.0.0:8089
tlscertfile = /etc/asterisk/keys/asterisk.pem tlsprivatekey = /etc/asterisk/keys/asterisk.pem

pjsip.conf
[transport2]
type = transport bind = 0.0.0.0
protocol = wss

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

[auth10]
type = auth username = webrtc_client password = webrtc_client

[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@192.168.33.33
Password: webrtc_client Realm: asterisk.org

In the sipML5 Expert settings I have Disable Video checked WebSocket Server URL: wss//192.168.33.33:8089/ws Disable 3GPP Early IMS: checked Disable debug messages: checked Cache the media stream: checked

One thought on - WebRTC Using SIPML5 Question

  • I figured out my problem. Cleared all browser settings at one point. Once I visited the secure IP address 8089/ws web page directly (accepting unsafe browsing) I was able to login.

    From: asterisk-users On Behalf Of Dan Cropp Sent: Wednesday, December 12, 2018 10:52 AM
    To: asterisk-users@lists.digium.com Subject: [asterisk-users] WebRTC using SIPML5 question

    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.

    http.conf
    [general]
    enabled = yes bindaddr = 0.0.0.0
    bindport = 8088
    tlsenable = yes tlsbindaddr = 0.0.0.0:8089
    tlscertfile = /etc/asterisk/keys/asterisk.pem tlsprivatekey = /etc/asterisk/keys/asterisk.pem

    pjsip.conf
    [transport2]
    type = transport bind = 0.0.0.0
    protocol = wss

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

    [auth10]
    type = auth username = webrtc_client password = webrtc_client

    [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@192.168.33.33
    Password: webrtc_client Realm: asterisk.org

    In the sipML5 Expert settings I have Disable Video checked WebSocket Server URL: wss//192.168.33.33:8089/ws Disable 3GPP Early IMS: checked Disable debug messages: checked Cache the media stream: checked