hi,
i have webrtc client chrome69/jssip which is connecting to asterisk
13.23.1/pjsip
i have strange problem where pjsip aor stays in status "created"
sip trace on asterisk looks ok.
do you think if this can be bug?
test*CLI> pjsip show aors
Aor: <Aor..............................................> <MaxContact>
Contact: <Aor/ContactUri............................> <Hash....>
<Status> <RTT(ms)..>
==========================================================================================
Aor: vr1k50 1
Contact: vr1k50/sip:[email protected]:34434;tran b2ad914030
Created 0.000
<--- Received SIP request (566 bytes) from WSS:1.1.1.1:34434 --->
REGISTER sip:sip.example.com SIP/2.0
Via: SIP/2.0/WSS v0i0at11ojbn.invalid;branch=z9hG4bK2155317
Max-Forwards: 69
To: <sip:[email protected]>
From: "vr1k50" <sip:[email protected]>;tag=d56ij3vuo3
Call-ID: 0mm678kf72bc9b5ur7ea8d
CSeq: 13 REGISTER
Contact:
<sip:[email protected];transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:41c3d275-9c22-42ff-aeb3-987cb48902c7>";expires=60
Expires: 60
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: path,gruu,outbound
User-Agent: JsSIP 3.2.9
Content-Length: 0
<--- Transmitting SIP response (484 bytes) to WSS:1.1.1.1:34434 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS
v0i0at11ojbn.invalid;rport=34434;received=1.1.1.1;branch=z9hG4bK2155317
Call-ID: 0mm678kf72bc9b5ur7ea8d
From: "vr1k50" <sip:[email protected]>;tag=d56ij3vuo3
To: <sip:[email protected]>;tag=z9hG4bK2155317
CSeq: 13 REGISTER
WWW-Authenticate: Digest
realm="asterisk",nonce="1540467808/121f72ae15612cc46a72e2861657a940",opaque="3060464337b28725",algorithm=md5,qop="auth"
Server: Asterisk PBX 13.23.1
Content-Length: 0
<--- Received SIP request (837 bytes) from WSS:1.1.1.1:34434 --->
REGISTER sip:sip.example.com SIP/2.0
Via: SIP/2.0/WSS v0i0at11ojbn.invalid;branch=z9hG4bK9799804
Max-Forwards: 69
To: <sip:[email protected]>
From: "vr1k50" <sip:[email protected]>;tag=d56ij3vuo3
Call-ID: 0mm678kf72bc9b5ur7ea8d
CSeq: 14 REGISTER
Authorization: Digest algorithm=MD5, username="vr1k50",
realm="asterisk", nonce="1540467808/121f72ae15612cc46a72e2861657a940",
uri="sip:sip.example.com", response="376b4ac58b01dde2e043931467bba55a",
opaque="3060464337b28725", qop=auth, cnonce="v8i7444gio8r", nc=00000001
Contact:
<sip:[email protected];transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:41c3d275-9c22-42ff-aeb3-987cb48902c7>";expires=60
Expires: 60
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: path,gruu,outbound
User-Agent: JsSIP 3.2.9
Content-Length: 0
<--- Transmitting SIP response (446 bytes) to WSS:1.1.1.1:34434 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS
v0i0at11ojbn.invalid;rport=34434;received=1.1.1.1;branch=z9hG4bK9799804
Call-ID: 0mm678kf72bc9b5ur7ea8d
From: "vr1k50" <sip:[email protected]>;tag=d56ij3vuo3
To: <sip:[email protected]>;tag=z9hG4bK9799804
CSeq: 14 REGISTER
Date: Thu, 25 Oct 2018 11:43:28 GMT
Contact: <sip:[email protected]:34434;transport=ws>;expires=59
Expires: 60
Server: Asterisk PBX 13.23.1
Content-Length: 0
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