PJ ICE Rx error status code: 370401 'Unauthorized'

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minhhanos

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May 21, 2015, 2:12:02 PM5/21/15
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Please help me!

Logs:

  
-- Executing [6001@international:1] Dial("SIP/6000-00000008", "SIP/6001,,30") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 14664
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.100.103:1157:
INVITE sip:60...@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.100.222:5060;branch=z9hG4bK421cda15;rport
Max-Forwards: 70
From: "Joe User" <sip:60...@192.168.100.222>;tag=as0650f4c5
To: <sip:60...@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
Contact: <sip:60...@192.168.100.222:5060;transport=WS>
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-12-r434356
Date: Sun, 17 May 2015 06:46:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 637

v=0
o=root 1331264635 1331264635 IN IP4 192.168.100.222
s=Asterisk PBX SVN-branch-12-r434356
c=IN IP4 192.168.100.222
t=0 0
m=audio 14664 RTP/SAVPF 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=ice-ufrag:59f53a20039d84ae60e9fdd4174b48f7
a=ice-pwd:34f276c80e4d0b136a7efbd66bd25186
a=candidate:Hc0a864de 1 UDP 2130706431 192.168.100.222 14664 typ host
a=candidate:Hc0a864de 2 UDP 2130706430 192.168.100.222 14665 typ host
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:8B267PV6naH9uumtmsjyFcY8il0aN09P6NWZCKOV

---
    -- Called SIP/6001
[May 17 13:46:56] WARNING[19306][C-00000004]: res_rtp_asterisk.c:2083 __rtp_recvfrom: PJ ICE Rx error status code: 370401 'Unauthorized'.
[May 17 13:46:56] WARNING[19306][C-00000004]: res_rtp_asterisk.c:4234 ast_rtp_read: RTP Read error: Unspecified.  Hanging up.
Scheduling destruction of SIP dialog '6321688643d44451...@192.168.100.222:5060' in 6400 ms (Method: INVITE)
  == Spawn extension (international, 6001, 1) exited non-zero on 'SIP/6000-00000008'
Scheduling destruction of SIP dialog 'c2932881-659a-6b66-182f-a01d9df9b8f4' in 6400 ms (Method: INVITE)

<--- Reliably Transmitting (NAT) to 192.168.100.128:20937 --->
SIP/2.0 603 Declined
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKLiV9a4IdW6wvW9BOH5E9h2UXfF3kd1rn;received=192.168.100.128;rport=20937
From: "6000"<sip:60...@192.168.100.222>;tag=LtaNXTMhwn90d10cw2J0
To: <sip:60...@192.168.100.222>;tag=as3fc16953
Call-ID: c2932881-659a-6b66-182f-a01d9df9b8f4
CSeq: 14678 INVITE
Server: Asterisk PBX SVN-branch-12-r434356
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from WS:192.168.100.128:20937 --->
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKLiV9a4IdW6wvW9BOH5E9h2UXfF3kd1rn;rport
From: "6000"<sip:60...@192.168.100.222>;tag=LtaNXTMhwn90d10cw2J0
To: <sip:60...@192.168.100.222>;tag=as3fc16953
Call-ID: c2932881-659a-6b66-182f-a01d9df9b8f4
CSeq: 14678 ACK
Content-Length: 0
Max-Forwards: 70


<------------->
--- (8 headers 0 lines) ---

<--- SIP read from WS:192.168.100.103:1157 --->
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WS 192.168.100.222:5060;rport=5060;branch=z9hG4bK421cda15
From: "Joe User"<sip:60...@192.168.100.222>;tag=as0650f4c5
To: <sip:60...@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
CSeq: 102 INVITE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Reliably Transmitting (NAT) to 192.168.100.103:1157:
CANCEL sip:60...@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.100.222:5060;branch=z9hG4bK421cda15;rport
Max-Forwards: 70
From: "Joe User" <sip:60...@192.168.100.222>;tag=as0650f4c5
To: <sip:60...@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
CSeq: 102 CANCEL
User-Agent: Asterisk PBX SVN-branch-12-r434356
Content-Length: 0


---
Scheduling destruction of SIP dialog '6321688643d44451...@192.168.100.222:5060' in 6400 ms (Method: INVITE)
[May 17 13:46:56] ERROR[19231]: pjsip:0 <?>:    icess0xa612364 ..Error sending STUN request: Network is unreachable

<--- SIP read from WS:192.168.100.103:1157 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.100.222:5060;rport=5060;branch=z9hG4bK421cda15
From: "Joe User"<sip:60...@192.168.100.222>;tag=as0650f4c5
To: <sip:60...@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=0BFAWpL1fAkB1CG0UrFy
Contact: <sip:60...@df7jal23ls0d.invalid;transport=ws>
CSeq: 102 CANCEL
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---

<--- SIP read from WS:192.168.100.103:1157 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 192.168.100.222:5060;rport=5060;branch=z9hG4bK421cda15
From: "Joe User"<sip:60...@192.168.100.222>;tag=as0650f4c5
To: <sip:60...@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=0BFAWpL1fAkB1CG0UrFy
Contact: <sip:60...@df7jal23ls0d.invalid;transport=ws>
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE


<------------->
--- (9 headers 0 lines) ---
list_route: route/path hop: <sip:60...@df7jal23ls0d.invalid;transport=ws>

<--- SIP read from WS:192.168.100.103:1157 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/WS 192.168.100.222:5060;rport=5060;branch=z9hG4bK421cda15
From: "Joe User"<sip:60...@192.168.100.222>;tag=as0650f4c5
To: <sip:60...@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=0BFAWpL1fAkB1CG0UrFy
CSeq: 102 INVITE
Content-Length: 0
Reason: SIP; cause=487; text="Request Cancelled"


<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 192.168.100.103:1157:
ACK sip:60...@df7jal23ls0d.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.100.222:5060;branch=z9hG4bK421cda15;rport
Max-Forwards: 70
From: "Joe User" <sip:60...@192.168.100.222>;tag=as0650f4c5
To: <sip:60...@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=0BFAWpL1fAkB1CG0UrFy
Contact: <sip:60...@192.168.100.222:5060;transport=WS>
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-branch-12-r434356
Content-Length: 0


---
Scheduling destruction of SIP dialog '6321688643d44451...@192.168.100.222:5060' in 6400 ms (Method: INVITE)

<--- SIP read from WS:192.168.100.103:1157 --->
SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/WS 192.168.100.222:5060;rport=5060;branch=z9hG4bK421cda15
From: "Joe User"<sip:60...@192.168.100.222>;tag=as0650f4c5
To: <sip:60...@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=0BFAWpL1fAkB1CG0UrFy
CSeq: 102 ACK
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
[May 17 13:46:56] WARNING[19303][C-00000004]: chan_sip.c:24475 handle_response: Remote host can't match request ACK to call '6321688643d44451...@192.168.100.222:5060'. Giving up.
Really destroying SIP dialog 'c2932881-659a-6b66-182f-a01d9df9b8f4' Method: INVITE
Really destroying SIP dialog '6321688643d44451...@192.168.100.222:5060' Method: INVITE
huyhtpc*CLI>

Ajay Choudary

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May 28, 2015, 6:34:19 AM5/28/15
to doub...@googlegroups.com
Issue is related to Bug_id=4495, you need to patch the PJProject to get the voice from buggy chrome versions.
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