WebRTC registration failed: unable to bind contact

Asterisk Asterisk SIP
1

Hi, I’ve asterisk 18.20.0 installed, and I’m trying to register a webrtc endpoint with this pjsip configuration.

[general]
udpbindaddr = 0.0.0.0

[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0:8089

[webrtc_client]
type=aor
max_contacts=1
remove_existing=yes

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

[webrtc_client]
type=endpoint
aors=webrtc_client
outbound_auth=webrtc_client
dtls_auto_generate_cert=yes
webrtc=yes
context=xentric-test-env
transport=transport-wss
disallow=all
allow=ulaw,alaw,gsm
direct_media=no
ice_support=yes

Trying to connect from the browser, I got this Error:

[Jan  5 13:22:15]   == WebSocket connection from 'xx.xx.xx.31:45542' for protocol 'sip' accepted using version '13'
[Jan  5 13:22:15] <--- Received SIP request (587 bytes) from WSS:xx.xx.xx.31:45542 --->
[Jan  5 13:22:15] REGISTER sip:xx.xx.xx.187 SIP/2.0
[Jan  5 13:22:15] Via: SIP/2.0/WSS 190l58ipsvtm.invalid;branch=z9hG4bK8413957
[Jan  5 13:22:15] Max-Forwards: 69
[Jan  5 13:22:15] To: <sip:webrtc_client@xx.xx.xx.187>
[Jan  5 13:22:15] From: "webrtc_client" <sip:webrtc_client@xx.xx.xx.187>;tag=ka0ri1ckch
[Jan  5 13:22:15] Call-ID: 9fe8jjl0giacocalb7fnf3
[Jan  5 13:22:15] CSeq: 1 REGISTER
[Jan  5 13:22:15] Contact: <sip:qrigmur6@190l58ipsvtm.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:166aa563-867e-4480-ac3b-92415beaa923>";expires=600
[Jan  5 13:22:15] Expires: 600
[Jan  5 13:22:15] Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
[Jan  5 13:22:15] Supported: path,gruu,outbound
[Jan  5 13:22:15] User-Agent: JsSIP 3.10.0
[Jan  5 13:22:15] Content-Length: 0
[Jan  5 13:22:15] 
[Jan  5 13:22:15] 
[Jan  5 13:22:15] WARNING[847519]: db.c:360 ast_db_put: Couldn't execute statement: SQL logic error
[Jan  5 13:22:15] ERROR[847519]: res_pjsip_registrar.c:861 register_aor_core: Unable to bind contact 'sip:qrigmur6@xx.xx.xx.31:45542;transport=ws;x-ast-orig-host=190l58ipsvtm.invalid:0' to AOR 'webrtc_client'
[Jan  5 13:22:15] <--- Transmitting SIP response (389 bytes) to WSS:xx.xx.xx.31:45542 --->
[Jan  5 13:22:15] SIP/2.0 200 OK
[Jan  5 13:22:15] Via: SIP/2.0/WSS 190l58ipsvtm.invalid;rport=45542;received=xx.xx.xx.31;branch=z9hG4bK8413957
[Jan  5 13:22:15] Call-ID: 9fe8jjl0giacocalb7fnf3
[Jan  5 13:22:15] From: "webrtc_client" <sip:webrtc_client@xx.xx.xx.187>;tag=ka0ri1ckch
[Jan  5 13:22:15] To: <sip:webrtc_client@xx.xx.xx.187>;tag=z9hG4bK8413957
[Jan  5 13:22:15] CSeq: 1 REGISTER
[Jan  5 13:22:15] Date: Fri, 05 Jan 2024 13:22:15 GMT
[Jan  5 13:22:15] Expires: 600
[Jan  5 13:22:15] Server: Asterisk PBX 18.20.0
[Jan  5 13:22:15] Content-Length:  0
[Jan  5 13:22:15] 
[Jan  5 13:22:15] 
[Jan  5 13:22:18]   == WebSocket connection from 'xx.xx.xx.31:45542' closed

I’have been trying different solutions for a week, but nothing seems to work out.

2

Howdy!

Excellent scrubbing of your IPs from the logs!

Does that even load ? It looks like some old chan_sip stuff, not chan_pjsip. Maybe take it out…

Don’t worry about the port in this section.

You might want to consider changing these sooner rather than later, as they are popular defaults per the related HOWTO in the docs.

Maybe try just “auth” ?

Does that context exist in your extensions.conf file ?

It does appear your http.conf is correct enough because you got this far to see the SIP output in your logs, so probably no need to post that.

3

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