sipML5 Disconnected

hello everyone
I refer to this https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients Configuring WebRTC,But I can’t register.When i login,the Asterisk is error.And it’s going to be shut down.

The error message:
*CLI> [Jun 25 10:25:30] ERROR[4372]: iostream.c:633 ast_iostream_start_tls: Problem setting up ssl connection: error:00000001:lib(0):func(0):reason(1), Internal SSL error
[Jun 25 10:25:30] ERROR[4372]: iostream.c:538 ast_iostream_close: SSL_shutdown() failed: error:00000001:lib(0):func(0):reason(1), Internal SSL error
== WebSocket connection from ‘192.168.1.150:1311’ for protocol ‘sip’ accepted using version ‘13’
asterisk: symbol lookup error: asterisk: undefined symbol: json_vsprintf

Can you help me. Thank you very much.

Issue seems to be the encryption for call signaling are you sure key files for the ssl are correctly set

I used the default settings. I only change the /etc/asterisk/pjsip.conf and http.conf for asterisk wiki

You need to setup the TLS connection setting up correctly the SSL files

1 Like

Sorry, I’m a rookie.Can you tell me something specific? Maybe you give me some link,I can learn for this
。 Thank you very much!!!

Webrtc might be complex even for experienced users, a lot things involved start by setting the TLS connection properly and then move forward there are many links on the Internet learn to search is par of your learning process

OK ! thank you again.

Anyone konw the problem,I need some help!
“asterisk: symbol lookup error: asterisk: undefined symbol: json_vsprintf”

You are running against the wrong version of a JSON library.

yes!I change the json 2.10.Solve this problem
WebRTC can Connected
thank you very much

Now,I can Connected sipML5 but i can‘t call 200 display the “Not Acceptable Here”

this is my /etc/asterisk/extensions.conf
[default]
exten => 200,1,Answer()
same => n,Playback(demo-congrats)
same => n,Hangu()

mistake:
ERROR[3861]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo(“cc444b8e-2385-4562-9f24-fae7301779fe.local”, “(null)”, …): Name or service not known

That will occur if the configuration is incorrect. What is the precise configuration?

this is my PJSIP.conf configure
[2000]
type=aor
max_contacts=5
remove_existing=yes

[2000]
type=auth
auth_type=userpass
username=2000
password=2000
realm=hiastar.com

[2000]
type=endpoint
aors=2000
auth=2000
dtls_auto_generate_cert=yes
webrtc=yes
; Setting webrtc=yes is a shortcut for setting the following options:
; use_avpf=yes
; media_encryption=dtls
; dtls_verify=fingerprint
; dtls_setup=actpass
; ice_support=yes
; media_use_received_transport=yes
; rtcp_mux=yes
context=test
disallow=all
allow=opus,ulaw

this is extensions.conf configure
[test]
exten => 200,1,Answer()
same=> n,Playback(demo-congrats)
same=> n,Hangup()

this test for LAN,Do you need additional configurations?

And I Generate certificates config is:
sudo contrib/scripts/ast_tls_cert -C localhost -O "WebRTC" -d /etc/asterisk/keys

And the SIP trace? (pjsip set logger on)

Thank you for your reply!I can login sipML5 but When i called 200 there will be an error!
this is my pjsip logger:

localhost*CLI> [Jun 27 03:00:19] ERROR[4612]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo(“a8a6d784-f0dc-4364-b6f3-02c1d0b5569a.local”, “(null)”, …): Name or service not known
<— Transmitting SIP response (405 bytes) to WSS:192.168.1.150:5639 —>
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=5639;received=192.168.1.150;branch=z9hG4bKTvsttHooWflKaZySbiFmnq6QERK4oiAq
Call-ID: b63c4b1c-8017-0d0a-f939-130b7a030a94
From: “2000” sip:2000@192.168.1.128;tag=7NlxvOETMFXUv7RKATso
To: sip:200@192.168.1.128;tag=d584769e-232b-4254-820f-053f5c29f32c
CSeq: 47047 INVITE
Server: Asterisk PBX 15.7.2
Content-Length: 0

[Jun 27 03:00:19] ERROR[4612]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo(“a8a6d784-f0dc-4364-b6f3-02c1d0b5569a.local”, “(null)”, …): Name or service not known
<— Transmitting SIP response (405 bytes) to WSS:192.168.1.150:5639 —>
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=5639;received=192.168.1.150;branch=z9hG4bKTvsttHooWflKaZySbiFmnq6QERK4oiAq
Call-ID: b63c4b1c-8017-0d0a-f939-130b7a030a94
From: “2000” sip:2000@192.168.1.128;tag=7NlxvOETMFXUv7RKATso
To: sip:200@192.168.1.128;tag=d584769e-232b-4254-820f-053f5c29f32c
CSeq: 47047 INVITE
Server: Asterisk PBX 15.7.2
Content-Length: 0

localhost*CLI> <— Received SIP request (363 bytes) from WSS:192.168.1.150:5639 —>
ACK sip:200@192.168.1.128 SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKTvsttHooWflKaZySbiFmnq6QERK4oiAq;rport
From: "2000"sip:2000@192.168.1.128;tag=7NlxvOETMFXUv7RKATso
To: sip:200@192.168.1.128;tag=d584769e-232b-4254-820f-053f5c29f32c
Call-ID: b63c4b1c-8017-0d0a-f939-130b7a030a94
CSeq: 47047 ACK
Content-Length: 0
Max-Forwards: 70

<— Received SIP request (363 bytes) from WSS:192.168.1.150:5639 —>
ACK sip:200@192.168.1.128 SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKTvsttHooWflKaZySbiFmnq6QERK4oiAq;rport
From: "2000"sip:2000@192.168.1.128;tag=7NlxvOETMFXUv7RKATso
To: sip:200@192.168.1.128;tag=d584769e-232b-4254-820f-053f5c29f32c
Call-ID: b63c4b1c-8017-0d0a-f939-130b7a030a94
CSeq: 47047 ACK
Content-Length: 0
Max-Forwards: 70

That is not the complete SIP trace. It doesn’t include the incoming call, just the rejection. As well I always suggest using a current supported version because WebRTC can change and break things, which are then resolved in supported versions.

I am using a Asterisk version 16.4.0 and 2.12 json,But it will Error,I can‘t login sipML5

== WebSocket connection from ‘192.168.1.150:1311’ for protocol ‘sip’ accepted using version ‘13’
asterisk: symbol lookup error: asterisk: undefined symbol: json_vsprintf

Now,I use Asterisk version 15.4 and 2.10 ison,can login sipML5

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.