Hello,
I'm running Asterisk 13, I've configured webRTC with sipML5 following Navaismo instruction (thanks a lot 
)
All seems to work as expected, I can call system prompts, also I can make calls between two webRTC clients via sipml5.org demo, audio also works fine. But when I take a look to java debug on Chrome I get the following errors:
Code:
Not implemented SIPml-api.js?svn=230:1
tsk_utils_log_error SIPml-api.js?svn=230:1
tsip_dialog_layer.handle_incoming_message SIPml-api.js?svn=230:3 tsip_transport_layer.handle_incoming_message SIPml-api.js?svn=230:3 __tsip_transport_ws_onmessage SIPml-api.js?svn=230:3
This is sip set debug on output from Asterisk
Code:
Reliably Transmitting (no NAT) to xx.xx.xx.150:20001:
OPTIONS sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0
Via: SIP/2.0/WS yy.yy.yy.251:5060;branch=z9hG4bK59915be4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@yy.yy.yy.251>;tag=as09d99b2c
To: <sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
Contact: <sip:asterisk@yy.yy.yy.251:5060;transport=WS>
Call-ID: 4ec9b4e71a8f2bc26980194d131fe756@yy.yy.yy.251:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.3.2
Date: Wed, 13 May 2015 17:03:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from WS:xx.xx.xx.150:20001 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/WS yy.yy.yy.251:5060;branch=z9hG4bK59915be4
From: "asterisk"<sip:asterisk@yy.yy.yy.251>;tag=as09d99b2c
To: <sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
Call-ID: 4ec9b4e71a8f2bc26980194d131fe756@yy.yy.yy.251:5060
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '432665145615819734367510' Method: OPTIONS
Really destroying SIP dialog '4ec9b4e71a8f2bc26980194d131fe756@yy.yy.yy.251:5060' Method: OPTIONS
<--- SIP read from WS:xx.xx.xx.150:20001 --->
INVITE sip:200@yy.yy.yy.251 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK6D6QFdIf8JayoVDhN8Zru3CQHVN5P1zV;rport
From: "Angel"<sip:6001@yy.yy.yy.251>;tag=mLwgfkfK1HS3rO6ZekKu
To: <sip:200@yy.yy.yy.251>
Contact: "Angel"<sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
Call-ID: 05118c7e-97cd-4119-9ee3-a528547bd25a
CSeq: 64296 INVITE
Content-Type: application/sdp
Content-Length: 2535
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
v=0
o=- 2230222985729852200 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS Ymc7ob59XyWiHVy4uIG78bw9u7iABIqCOua3
m=audio 60668 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 xx.xx.xx.150
a=rtcp:60668 IN IP4 xx.xx.xx.150
a=candidate:3013953624 1 udp 2122260223 192.168.1.100 60668 typ host generation 0
a=candidate:3013953624 2 udp 2122260223 192.168.1.100 60668 typ host generation 0
a=candidate:174257638 1 udp 2122194687 192.168.146.1 60669 typ host generation 0
a=candidate:174257638 2 udp 2122194687 192.168.146.1 60669 typ host generation 0
a=candidate:3284899927 1 udp 2122129151 192.168.112.1 60670 typ host generation 0
a=candidate:3284899927 2 udp 2122129151 192.168.112.1 60670 typ host generation 0
a=candidate:854413036 1 udp 1686052607 xx.xx.xx.150 60668 typ srflx raddr 192.168.1.100 rport 60668 generation 0
a=candidate:854413036 2 udp 1686052607 xx.xx.xx.150 60668 typ srflx raddr 192.168.1.100 rport 60668 generation 0
a=candidate:4247172264 1 tcp 1518280447 192.168.1.100 0 typ host tcptype active generation 0
a=candidate:4247172264 2 tcp 1518280447 192.168.1.100 0 typ host tcptype active generation 0
a=candidate:1155598614 1 tcp 1518214911 192.168.146.1 0 typ host tcptype active generation 0
a=candidate:1155598614 2 tcp 1518214911 192.168.146.1 0 typ host tcptype active generation 0
a=candidate:2370331815 1 tcp 1518149375 192.168.112.1 0 typ host tcptype active generation 0
a=candidate:2370331815 2 tcp 1518149375 192.168.112.1 0 typ host tcptype active generation 0
a=ice-ufrag:vpLhvEmFtIJA9K2Y
a=ice-pwd:fCmjWQmjIMk4tzefPz2Weo07
a=ice-options:google-ice
a=fingerprint:sha-256 5D:66:87:30:62:C3:ED:79:A2:36:DE:24:7E:1C:3D:3B:1B:71:1D:FF:14:D2:FD:EE:53:2D:A4:9F:67:60:D3:3A
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10; useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:707164240 cname:8ZoL+rGq1tIvvfAB
a=ssrc:707164240 msid:Ymc7ob59XyWiHVy4uIG78bw9u7iABIqCOua3 38e85135-7ff7-41c6-a035-320c761f35cc
a=ssrc:707164240 mslabel:Ymc7ob59XyWiHVy4uIG78bw9u7iABIqCOua3
a=ssrc:707164240 label:38e85135-7ff7-41c6-a035-320c761f35cc
<------------->
--- (12 headers 49 lines) ---
Using INVITE request as basis request - 05118c7e-97cd-4119-9ee3-a528547bd25a
Found peer '6001' for '6001' from xx.xx.xx.150:20001
<--- Reliably Transmitting (no NAT) to xx.xx.xx.150:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK6D6QFdIf8JayoVDhN8Zru3CQHVN5P1zV;rport;received=xx.xx.xx.150
From: "Angel"<sip:6001@yy.yy.yy.251>;tag=mLwgfkfK1HS3rO6ZekKu
To: <sip:200@yy.yy.yy.251>;tag=as6ca53ec9
Call-ID: 05118c7e-97cd-4119-9ee3-a528547bd25a
CSeq: 64296 INVITE
Server: Asterisk PBX 13.3.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="yy.yy.yy.251", nonce="50766959"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '05118c7e-97cd-4119-9ee3-a528547bd25a' in 6400 ms (Method: INVITE)
<--- SIP read from WS:xx.xx.xx.150:20001 --->
ACK sip:200@yy.yy.yy.251 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK6D6QFdIf8JayoVDhN8Zru3CQHVN5P1zV;rport
From: "Angel"<sip:6001@yy.yy.yy.251>;tag=mLwgfkfK1HS3rO6ZekKu
To: <sip:200@yy.yy.yy.251>;tag=as6ca53ec9
Call-ID: 05118c7e-97cd-4119-9ee3-a528547bd25a
CSeq: 64296 ACK
Content-Length: 0
Max-Forwards: 70
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from WS:xx.xx.xx.150:20001 --->
INVITE sip:200@yy.yy.yy.251 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKfZfX3pDTrQRmK8FT2fEwVmzHRWzDaOeV;rport
From: "Angel"<sip:6001@yy.yy.yy.251>;tag=mLwgfkfK1HS3rO6ZekKu
To: <sip:200@yy.yy.yy.251>
Contact: "Angel"<sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
Call-ID: 05118c7e-97cd-4119-9ee3-a528547bd25a
CSeq: 64297 INVITE
Content-Type: application/sdp
Content-Length: 2535
Max-Forwards: 70
Authorization: Digest username="6001",realm="yy.yy.yy.251",nonce="50766959",uri="sip:200@yy.yy.yy.251",response="ec2939473729233afb289ee48347c9b6",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
v=0
o=- 2230222985729852200 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS Ymc7ob59XyWiHVy4uIG78bw9u7iABIqCOua3
m=audio 60668 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 xx.xx.xx.150
a=rtcp:60668 IN IP4 xx.xx.xx.150
a=candidate:3013953624 1 udp 2122260223 192.168.1.100 60668 typ host generation 0
a=candidate:3013953624 2 udp 2122260223 192.168.1.100 60668 typ host generation 0
a=candidate:174257638 1 udp 2122194687 192.168.146.1 60669 typ host generation 0
a=candidate:174257638 2 udp 2122194687 192.168.146.1 60669 typ host generation 0
a=candidate:3284899927 1 udp 2122129151 192.168.112.1 60670 typ host generation 0
a=candidate:3284899927 2 udp 2122129151 192.168.112.1 60670 typ host generation 0
a=candidate:854413036 1 udp 1686052607 xx.xx.xx.150 60668 typ srflx raddr 192.168.1.100 rport 60668 generation 0
a=candidate:854413036 2 udp 1686052607 xx.xx.xx.150 60668 typ srflx raddr 192.168.1.100 rport 60668 generation 0
a=candidate:4247172264 1 tcp 1518280447 192.168.1.100 0 typ host tcptype active generation 0
a=candidate:4247172264 2 tcp 1518280447 192.168.1.100 0 typ host tcptype active generation 0
a=candidate:1155598614 1 tcp 1518214911 192.168.146.1 0 typ host tcptype active generation 0
a=candidate:1155598614 2 tcp 1518214911 192.168.146.1 0 typ host tcptype active generation 0
a=candidate:2370331815 1 tcp 1518149375 192.168.112.1 0 typ host tcptype active generation 0
a=candidate:2370331815 2 tcp 1518149375 192.168.112.1 0 typ host tcptype active generation 0
a=ice-ufrag:vpLhvEmFtIJA9K2Y
a=ice-pwd:fCmjWQmjIMk4tzefPz2Weo07
a=ice-options:google-ice
a=fingerprint:sha-256 5D:66:87:30:62:C3:ED:79:A2:36:DE:24:7E:1C:3D:3B:1B:71:1D:FF:14:D2:FD:EE:53:2D:A4:9F:67:60:D3:3A
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10; useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:707164240 cname:8ZoL+rGq1tIvvfAB
a=ssrc:707164240 msid:Ymc7ob59XyWiHVy4uIG78bw9u7iABIqCOua3 38e85135-7ff7-41c6-a035-320c761f35cc
a=ssrc:707164240 mslabel:Ymc7ob59XyWiHVy4uIG78bw9u7iABIqCOua3
a=ssrc:707164240 label:38e85135-7ff7-41c6-a035-320c761f35cc
<------------->
--- (13 headers 49 lines) ---
Using INVITE request as basis request - 05118c7e-97cd-4119-9ee3-a528547bd25a
Found peer '6001' for '6001' from xx.xx.xx.150:20001
== Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 126
Found audio description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126
Capabilities: us - (gsm|ulaw), peer - audio=(ulaw|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port xx.xx.xx.150:60668
Looking for 200 in local (domain yy.yy.yy.251)
sip_route_dump: route/path hop: <sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>
<--- Transmitting (no NAT) to xx.xx.xx.150:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKfZfX3pDTrQRmK8FT2fEwVmzHRWzDaOeV;rport;received=xx.xx.xx.150
From: "Angel"<sip:6001@yy.yy.yy.251>;tag=mLwgfkfK1HS3rO6ZekKu
To: <sip:200@yy.yy.yy.251>
Call-ID: 05118c7e-97cd-4119-9ee3-a528547bd25a
CSeq: 64297 INVITE
Server: Asterisk PBX 13.3.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:200@yy.yy.yy.251:5060;transport=WS>
Content-Length: 0
<------------>
-- Executing [200@local:1] Answer("SIP/6001-0000000b", "") in new stack
Audio is at 19766
Adding codec ulaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to xx.xx.xx.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKfZfX3pDTrQRmK8FT2fEwVmzHRWzDaOeV;rport;received=xx.xx.xx.150
From: "Angel"<sip:6001@yy.yy.yy.251>;tag=mLwgfkfK1HS3rO6ZekKu
To: <sip:200@yy.yy.yy.251>;tag=as3ac69b3b
Call-ID: 05118c7e-97cd-4119-9ee3-a528547bd25a
CSeq: 64297 INVITE
Server: Asterisk PBX 13.3.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:200@yy.yy.yy.251:5060;transport=WS>
Content-Type: application/sdp
Content-Length: 851
v=0
o=root 1347265751 1347265751 IN IP4 yy.yy.yy.251
s=Asterisk PBX 13.3.2
c=IN IP4 yy.yy.yy.251
t=0 0
m=audio 19766 RTP/SAVPF 0 3 126
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=maxptime:150
a=ice-ufrag:0c7fc36d75377acb4e3f914700da3d38
a=ice-pwd:04eed8f90f1b940d2ed92776662c8188
a=candidate:H253073fb 1 UDP 2130706431 yy.yy.yy.251 19766 typ host
a=candidate:S253073fb 1 UDP 1694498815 yy.yy.yy.251 19766 typ srflx raddr yy.yy.yy.251 rport 19766
a=candidate:H253073fb 2 UDP 2130706430 yy.yy.yy.251 19767 typ host
a=candidate:S253073fb 2 UDP 1694498814 yy.yy.yy.251 19767 typ srflx raddr yy.yy.yy.251 rport 19767
a=connection:new
a=setup:active
a=fingerprint:SHA-256 19:9D:27:CD:7E:AD:3D:15:02:11:17:A9:09:83:15:48:1B:29:A0:9B:B8:62:DC:7D:E4:41:91:B3:91:08:36:B8
a=sendrecv
<------------>
<--- SIP read from WS:xx.xx.xx.150:20001 --->
ACK sip:200@yy.yy.yy.251:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKBuNzm5oGTuAZBMUne9PC;rport
From: "Angel"<sip:6001@yy.yy.yy.251>;tag=mLwgfkfK1HS3rO6ZekKu
To: <sip:200@yy.yy.yy.251>;tag=as3ac69b3b
Contact: "Angel"<sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
Call-ID: 05118c7e-97cd-4119-9ee3-a528547bd25a
CSeq: 64297 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6001",realm="yy.yy.yy.251",nonce="50766959",uri="sip:200@yy.yy.yy.251:5060;transport=WS",response="aed534047b8fd827d896f3160311224d",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
<------------->
--- (12 headers 0 lines) ---
-- Executing [200@local:2] Playback("SIP/6001-0000000b", "demo-congrats") in new stack
-- <SIP/6001-0000000b> Playing 'demo-congrats.gsm' (language 'en')
> 0x7f8bfc025280 -- Probation passed - setting RTP source address to xx.xx.xx.150:60668
-- Executing [200@local:3] Hangup("SIP/6001-0000000b", "") in new stack
== Spawn extension (local, 200, 3) exited non-zero on 'SIP/6001-0000000b'
Scheduling destruction of SIP dialog '05118c7e-97cd-4119-9ee3-a528547bd25a' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Reliably Transmitting (no NAT) to xx.xx.xx.150:5060:
BYE sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws SIP/2.0
Via: SIP/2.0/WS yy.yy.yy.251:5060;branch=z9hG4bK4ab6d131
Max-Forwards: 70
From: <sip:200@yy.yy.yy.251>;tag=as3ac69b3b
To: "Angel"<sip:6001@yy.yy.yy.251>;tag=mLwgfkfK1HS3rO6ZekKu
Call-ID: 05118c7e-97cd-4119-9ee3-a528547bd25a
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.3.2
Proxy-Authorization: Digest username="6001", realm="yy.yy.yy.251", algorithm=MD5, uri="sip:yy.yy.yy.251", nonce="50766959", response="36676fe87a456ebbfef32d7e290b2ad9"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from WS:xx.xx.xx.150:20001 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS yy.yy.yy.251:5060;branch=z9hG4bK4ab6d131
From: <sip:200@yy.yy.yy.251>;tag=as3ac69b3b
To: "Angel"<sip:6001@yy.yy.yy.251>;tag=mLwgfkfK1HS3rO6ZekKu
Contact: <sip:6001@df7jal23ls0d.invalid;transport=ws>
Call-ID: 05118c7e-97cd-4119-9ee3-a528547bd25a
CSeq: 102 BYE
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '05118c7e-97cd-4119-9ee3-a528547bd25a' Method: INVITE
<--- SIP read from WS:xx.xx.xx.150:20001 --->
REGISTER sip:yy.yy.yy.251 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKF6HKScnIu4Aswmna7SWuLgbZmVKsMZ3X;rport
From: "Angel"<sip:6001@yy.yy.yy.251>;tag=NyTsJR4lpcLBXlItsN8f
To: "Angel"<sip:6001@yy.yy.yy.251>
Contact: "Angel"<sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: bf63c485-fa48-4d3f-4a45-a42135a4f6d4
CSeq: 27588 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6001",realm="yy.yy.yy.251",nonce="2a088823",uri="sip:yy.yy.yy.251",response="f56c52332dd34808cda88c5179ca369b",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
<------------->
--- (12 headers 0 lines) ---
<--- Transmitting (no NAT) to xx.xx.xx.150:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKF6HKScnIu4Aswmna7SWuLgbZmVKsMZ3X;rport;received=xx.xx.xx.150
From: "Angel"<sip:6001@yy.yy.yy.251>;tag=NyTsJR4lpcLBXlItsN8f
To: "Angel"<sip:6001@yy.yy.yy.251>;tag=as598a8f75
Call-ID: bf63c485-fa48-4d3f-4a45-a42135a4f6d4
CSeq: 27588 REGISTER
Server: Asterisk PBX 13.3.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="yy.yy.yy.251", nonce="4f23be69"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'bf63c485-fa48-4d3f-4a45-a42135a4f6d4' in 32000 ms (Method: REGISTER)
<--- SIP read from WS:xx.xx.xx.150:20001 --->
REGISTER sip:yy.yy.yy.251 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKc6llsLGB0hiBDXA75X9xWRvwGUzOgJ8i;rport
From: "Angel"<sip:6001@yy.yy.yy.251>;tag=NyTsJR4lpcLBXlItsN8f
To: "Angel"<sip:6001@yy.yy.yy.251>
Contact: "Angel"<sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: bf63c485-fa48-4d3f-4a45-a42135a4f6d4
CSeq: 27589 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6001",realm="yy.yy.yy.251",nonce="4f23be69",uri="sip:yy.yy.yy.251",response="878c522e3169fb16429fa80107f096c9",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom
<------------->
--- (12 headers 0 lines) ---
Reliably Transmitting (no NAT) to xx.xx.xx.150:20001:
OPTIONS sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0
Via: SIP/2.0/WS yy.yy.yy.251:5060;branch=z9hG4bK17125c1b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@yy.yy.yy.251>;tag=as2ec47ece
To: <sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
Contact: <sip:asterisk@yy.yy.yy.251:5060;transport=WS>
Call-ID: 2dc4cdef148be6a14b643eb83b9bc4e0@yy.yy.yy.251:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.3.2
Date: Wed, 13 May 2015 17:03:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- Transmitting (no NAT) to xx.xx.xx.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKc6llsLGB0hiBDXA75X9xWRvwGUzOgJ8i;rport;received=xx.xx.xx.150
From: "Angel"<sip:6001@yy.yy.yy.251>;tag=NyTsJR4lpcLBXlItsN8f
To: "Angel"<sip:6001@yy.yy.yy.251>;tag=as598a8f75
Call-ID: bf63c485-fa48-4d3f-4a45-a42135a4f6d4
CSeq: 27589 REGISTER
Server: Asterisk PBX 13.3.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 200
Contact: <sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200
Date: Wed, 13 May 2015 17:03:46 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'bf63c485-fa48-4d3f-4a45-a42135a4f6d4' in 32000 ms (Method: REGISTER)
<--- SIP read from WS:xx.xx.xx.150:20001 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/WS yy.yy.yy.251:5060;branch=z9hG4bK17125c1b
From: "asterisk"<sip:asterisk@yy.yy.yy.251>;tag=as2ec47ece
To: <sip:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
Call-ID: 2dc4cdef148be6a14b643eb83b9bc4e0@yy.yy.yy.251:5060
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '2dc4cdef148be6a14b643eb83b9bc4e0@yy.yy.yy.251:5060' Method: OPTIONS
As you can see in the code I get some:
401 Unauthorized and 405 Method Not Allowed sip messages
I'm not sure if it's related with Chrome errors.
This is my sip.conf
Code:
[general]
context=unauthenticated ; default context for incoming calls
allowguest=no ; disable unauthenticated calls
srvlookup=no ; disable DNS SRV record lookup on outbound calls
udpbindaddr=0.0.0.0 ; listen for UDP requests on all interfaces
tcpenable=no ; disable TCP support
udpbindaddr=0.0.0.0:5060
realm=yy.yy.yy.251 ;replace with your Asterisk server public IP address or host
transport=ws,udp
[6001]
host=dynamic
secret=6001
context=local
type=friend
avpf=yes
force_avp=yes
icesupport=yes
directmedia=no
disallow=all
allow=gsm
allow=ulaw
allow=allaw
dtmf=auto
transport=ws,udp
qualify=yes
encryption=yes
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is
dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS
Any help would be apreciated
Thanks in advance
Statistics : Posted by angelope • on Wed May 13, 2015 11:32 am • Replies 0 • Views 14