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SipML5 + Asterisk : Remote host can't match request ACK #197

Closed GoogleCodeExporter closed 8 years ago

GoogleCodeExporter commented 8 years ago 
a) Before posting your issue you MUST answer to the questions otherwise it
will be rejected (invalid status) by us
b) Please check the issue tacker to avoid duplication
c) Please provide network capture (Wireshark) or Javascript console log
if you want quick response

What steps will reproduce the problem?
1. Asterisk is installed in a Ubuntu VM with an IP address : 192.168.10.109
 And the two SipML5 clients are on Chrome in Windows, the IP address is : 192.168.10.102

2.  I used one client at Chrome on default mode and the other client on the 
anonymous mode :
 In http://sipml5.org/call.htm?svn=224# :
 Display Name, Private Identity, and Password : 1060 / 1061 (as follows for Client 1 and 2)
 Public Identity : sip:1060@doubango.org / sip:1061@doubango.org (as follows for  Client 1 and 2)
 Realm : doubango.org

 In the expert mode, http://sipml5.org/expert.htm in both sides I have :
 Disable Video : ticked
 Enable RTCWeb Breaker : ticked
 WebSocket Server URL : ws://192.168.10.109:8088/ws
 ICE Servers : [{url:'stun:stun.l.google.com:19302'}]

3.

What is the expected output? What do you see instead?

 It is expected for Client 2 to answer the call from Client 1. But when Chrome asks to allow the micro, by clicking on "Allow" the "Call Rejected".

What version of the product are you using? On what operating system?

 Asterisk 13.0.1 + SipML5 on Chrome 39

Please provide any additional information below.

Here's the log I got :
--------------------------------------------------------------------------
<--- SIP read from WS:192.168.10.102:50162 --->
INVITE sip:1061@doubango.org SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKQMKlkFEyIu2fstl3S3DOo30PA7mNmATe;rport
From: "1060"<sip:1060@doubango.org>;tag=8uEBBzWaND7t0A39IcpD
To: <sip:1061@doubango.org>
Contact: 
"1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=
ws>;impi=1060;ha1=11ae210c967672e2981b38443f193fd4;+g.oma.sip-im;+sip.ice;langua
ge="en,fr"
Call-ID: b6dd849a-c20e-c682-0802-7b0edb3365a1
CSeq: 65358 INVITE
Content-Type: application/sdp
Content-Length: 2153
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

v=0
o=- 8287455714875358000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 7Xm9A4y63ZJIvNk5DNkisHLnRkuESOuomh37
m=audio 49927 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 207.162.10.166
a=rtcp:49927 IN IP4 207.162.10.166
a=candidate:2999745851 1 udp 2122260223 192.168.56.1 49926 typ host generation 0
a=candidate:2999745851 2 udp 2122260223 192.168.56.1 49926 typ host generation 0
a=candidate:3782639666 1 udp 2122194687 192.168.10.102 49927 typ host 
generation 0
a=candidate:3782639666 2 udp 2122194687 192.168.10.102 49927 typ host 
generation 0
a=candidate:771473313 1 udp 1685987071 207.162.10.166 49927 typ srflx raddr 
192.168.10.102 rport 49927 generation 0
a=candidate:771473313 2 udp 1685987071 207.162.10.166 49927 typ srflx raddr 
192.168.10.102 rport 49927 generation 0
a=candidate:4233069003 1 tcp 1518280447 192.168.56.1 0 typ host tcptype active 
generation 0
a=candidate:4233069003 2 tcp 1518280447 192.168.56.1 0 typ host tcptype active 
generation 0
a=candidate:2952101058 1 tcp 1518214911 192.168.10.102 0 typ host tcptype 
active generation 0
a=candidate:2952101058 2 tcp 1518214911 192.168.10.102 0 typ host tcptype 
active generation 0
a=ice-ufrag:MwhBkSFG9Ov3z5qp
a=ice-pwd:rww0LRtT2qTNHirCQ/QNBlWm
a=ice-options:google-ice
a=fingerprint:sha-256 
B5:27:21:87:CD:21:BF:E9:92:A9:55:16:AE:FE:5F:8E:5D:7D:4F:3C:B2:EA:CA:51:CB:D2:A1
:F4:ED:03:54:E7
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
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
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:2380123242 cname:BsVHcarYr2rdtCpy
a=ssrc:2380123242 msid:7Xm9A4y63ZJIvNk5DNkisHLnRkuESOuomh37 
ab2994c3-e2e3-4903-b147-5eb3be8602db
a=ssrc:2380123242 mslabel:7Xm9A4y63ZJIvNk5DNkisHLnRkuESOuomh37
a=ssrc:2380123242 label:ab2994c3-e2e3-4903-b147-5eb3be8602db
<------------->
--- (12 headers 44 lines) ---
Using INVITE request as basis request - b6dd849a-c20e-c682-0802-7b0edb3365a1
Found peer '1060' for '1060' from 192.168.10.102:50162

<--- Reliably Transmitting (NAT) to 192.168.10.102:50162 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKQMKlkFEyIu2fstl3S3DOo30PA7mNmATe;received=192
.168.10.102;rport=50162
From: "1060"<sip:1060@doubango.org>;tag=8uEBBzWaND7t0A39IcpD
To: <sip:1061@doubango.org>;tag=as4856ab1f
Call-ID: b6dd849a-c20e-c682-0802-7b0edb3365a1
CSeq: 65358 INVITE
Server: Asterisk PBX 13.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="doubango.org", nonce="0f9f4c90"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'b6dd849a-c20e-c682-0802-7b0edb3365a1' in 
32000 ms (Method: INVITE)

<--- SIP read from WS:192.168.10.102:50162 --->
ACK sip:1061@doubango.org SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKQMKlkFEyIu2fstl3S3DOo30PA7mNmATe;rport
From: "1060"<sip:1060@doubango.org>;tag=8uEBBzWaND7t0A39IcpD
To: <sip:1061@doubango.org>;tag=as4856ab1f
Call-ID: b6dd849a-c20e-c682-0802-7b0edb3365a1
CSeq: 65358 ACK
Content-Length: 0
Max-Forwards: 70

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

<--- SIP read from WS:192.168.10.102:50162 --->
INVITE sip:1061@doubango.org SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKaRHhavVgQhNexGzeicZ856ayuCnXDk7r;rport
From: "1060"<sip:1060@doubango.org>;tag=8uEBBzWaND7t0A39IcpD
To: <sip:1061@doubango.org>
Contact: 
"1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=
ws>;impi=1060;ha1=11ae210c967672e2981b38443f193fd4;+g.oma.sip-im;+sip.ice;langua
ge="en,fr"
Call-ID: b6dd849a-c20e-c682-0802-7b0edb3365a1
CSeq: 65359 INVITE
Content-Type: application/sdp
Content-Length: 2153
Max-Forwards: 70
Authorization: Digest 
username="1060",realm="doubango.org",nonce="0f9f4c90",uri="sip:1061@doubango.org
",response="fad8bc7189034b543fa54058e369b451",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

v=0
o=- 8287455714875358000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 7Xm9A4y63ZJIvNk5DNkisHLnRkuESOuomh37
m=audio 49927 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 207.162.10.166
a=rtcp:49927 IN IP4 207.162.10.166
a=candidate:2999745851 1 udp 2122260223 192.168.56.1 49926 typ host generation 0
a=candidate:2999745851 2 udp 2122260223 192.168.56.1 49926 typ host generation 0
a=candidate:3782639666 1 udp 2122194687 192.168.10.102 49927 typ host 
generation 0
a=candidate:3782639666 2 udp 2122194687 192.168.10.102 49927 typ host 
generation 0
a=candidate:771473313 1 udp 1685987071 207.162.10.166 49927 typ srflx raddr 
192.168.10.102 rport 49927 generation 0
a=candidate:771473313 2 udp 1685987071 207.162.10.166 49927 typ srflx raddr 
192.168.10.102 rport 49927 generation 0
a=candidate:4233069003 1 tcp 1518280447 192.168.56.1 0 typ host tcptype active 
generation 0
a=candidate:4233069003 2 tcp 1518280447 192.168.56.1 0 typ host tcptype active 
generation 0
a=candidate:2952101058 1 tcp 1518214911 192.168.10.102 0 typ host tcptype 
active generation 0
a=candidate:2952101058 2 tcp 1518214911 192.168.10.102 0 typ host tcptype 
active generation 0
a=ice-ufrag:MwhBkSFG9Ov3z5qp
a=ice-pwd:rww0LRtT2qTNHirCQ/QNBlWm
a=ice-options:google-ice
a=fingerprint:sha-256 
B5:27:21:87:CD:21:BF:E9:92:A9:55:16:AE:FE:5F:8E:5D:7D:4F:3C:B2:EA:CA:51:CB:D2:A1
:F4:ED:03:54:E7
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
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
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:2380123242 cname:BsVHcarYr2rdtCpy
a=ssrc:2380123242 msid:7Xm9A4y63ZJIvNk5DNkisHLnRkuESOuomh37 
ab2994c3-e2e3-4903-b147-5eb3be8602db
a=ssrc:2380123242 mslabel:7Xm9A4y63ZJIvNk5DNkisHLnRkuESOuomh37
a=ssrc:2380123242 label:ab2994c3-e2e3-4903-b147-5eb3be8602db
<------------->
--- (13 headers 44 lines) ---
Using INVITE request as basis request - b6dd849a-c20e-c682-0802-7b0edb3365a1
Found peer '1060' for '1060' from 192.168.10.102:50162
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
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 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
failed to extend from 64 to 98
Capabilities: us - 
(ulaw|alaw|gsm|h263|g723|g726|g726aal2|adpcm|slin|slin|slin|), peer - 
audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 
(telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 207.162.10.166:49927
Peer doesn't provide video
Looking for 1061 in default (domain doubango.org)
sip_route_dump: route/path hop: 
<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>

<--- Transmitting (NAT) to 192.168.10.102:50162 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKaRHhavVgQhNexGzeicZ856ayuCnXDk7r;received=192
.168.10.102;rport=50162
From: "1060"<sip:1060@doubango.org>;tag=8uEBBzWaND7t0A39IcpD
To: <sip:1061@doubango.org>
Call-ID: b6dd849a-c20e-c682-0802-7b0edb3365a1
CSeq: 65359 INVITE
Server: Asterisk PBX 13.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1061@192.168.10.109:5060;transport=WS>
Content-Length: 0

<------------>
We think we can do text
Audio is at 16562
Video is at 192.168.10.109:19038
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding video codec h263 to SDP
Adding codec g723 to SDP
Adding codec g726 to SDP
Adding codec g726aal2 to SDP
Adding codec adpcm to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec lpc10 to SDP
Adding codec g729 to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec ilbc to SDP
Adding codec g722 to SDP
Adding codec siren7 to SDP
Adding codec siren14 to SDP
Adding codec testlaw to SDP
Adding codec g719 to SDP
Adding codec opus to SDP
Adding video codec h261 to SDP
Adding video codec h263p to SDP
Adding video codec h264 to SDP
Adding video codec mpeg4 to SDP
Adding video codec vp8 to SDP
Adding codec none to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.10.102:50163:
INVITE sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.10.109:5060;branch=z9hG4bK67ebe9be;rport
Max-Forwards: 70
From: "1060" <sip:1060@192.168.10.109>;tag=as162314ad
To: <sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Contact: <sip:1060@192.168.10.109:5060;transport=WS>
Call-ID: 69af5c6775ff2046095da6a54e22d43b@192.168.10.109:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.0.1
Date: Fri, 05 Dec 2014 17:13:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1389

v=0
o=root 85583298 85583298 IN IP4 192.168.10.109
s=Asterisk PBX 13.0.1
c=IN IP4 192.168.10.109
b=CT:384
t=0 0
m=audio 16562 UDP/TLS/RTP/SAVPF 0 8 3 4 111 112 5 10 118 7 18 110 117 119 97 9 
102 115 116 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=0
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=connection:new
a=setup:actpass
a=fingerprint:SHA-1 C5:4F:EC:21:D8:2A:23:17:02:27:83:F1:6D:D6:BE:83:AF:94:A9:B7
a=sendrecv
m=video 19038 UDP/TLS/RTP/SAVPF 34 31 98 99 104 100
a=connection:new
a=setup:actpass
a=fingerprint:SHA-1 C5:4F:EC:21:D8:2A:23:17:02:27:83:F1:6D:D6:BE:83:AF:94:A9:B7
a=rtpmap:34 H263/90000
a=rtpmap:31 H261/90000
a=rtpmap:98 h263-1998/90000
a=rtpmap:99 H264/90000
a=rtpmap:104 MP4V-ES/90000
a=rtpmap:100 VP8/90000
a=rtcp-fb:* ccm fir
a=sendrecv

---

<--- SIP read from WS:192.168.10.102:50163 --->
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WS 192.168.10.109:5060;rport=5060;branch=z9hG4bK67ebe9be
From: "1060"<sip:1060@192.168.10.109>;tag=as162314ad
To: <sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Call-ID: 69af5c6775ff2046095da6a54e22d43b@192.168.10.109:5060
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from WS:192.168.10.102:50163 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 192.168.10.109:5060;rport=5060;branch=z9hG4bK67ebe9be
From: "1060"<sip:1060@192.168.10.109>;tag=as162314ad
To: 
<sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=Vu3uUh9OTdna
T1vVhcoZ
Contact: <sip:1061@df7jal23ls0d.invalid;transport=ws>
Call-ID: 69af5c6775ff2046095da6a54e22d43b@192.168.10.109:5060
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

<------------->
--- (9 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:1061@df7jal23ls0d.invalid;transport=ws>

<--- Transmitting (NAT) to 192.168.10.102:50162 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKaRHhavVgQhNexGzeicZ856ayuCnXDk7r;received=192
.168.10.102;rport=50162
From: "1060"<sip:1060@doubango.org>;tag=8uEBBzWaND7t0A39IcpD
To: <sip:1061@doubango.org>;tag=as4715a5ca
Call-ID: b6dd849a-c20e-c682-0802-7b0edb3365a1
CSeq: 65359 INVITE
Server: Asterisk PBX 13.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1061@192.168.10.109:5060;transport=WS>
Content-Length: 0

<------------>
Really destroying SIP dialog '07b1f055-deeb-64cc-0687-cb841aada377' Method: 
REGISTER

<--- SIP read from WS:192.168.10.102:50163 --->
SIP/2.0 603 Failed to get local SDP
Via: SIP/2.0/WS 192.168.10.109:5060;rport=5060;branch=z9hG4bK67ebe9be
From: "1060"<sip:1060@192.168.10.109>;tag=as162314ad
To: 
<sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=Vu3uUh9OTdna
T1vVhcoZ
Call-ID: 69af5c6775ff2046095da6a54e22d43b@192.168.10.109:5060
CSeq: 102 INVITE
Content-Length: 0
Reason: SIP; cause=603; text="Failed to get local SDP"

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 192.168.10.102:50163:
ACK sip:1061@df7jal23ls0d.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.10.109:5060;branch=z9hG4bK67ebe9be;rport
Max-Forwards: 70
From: "1060" <sip:1060@192.168.10.109>;tag=as162314ad
To: 
<sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=Vu3uUh9OTdna
T1vVhcoZ
Contact: <sip:1060@192.168.10.109:5060;transport=WS>
Call-ID: 69af5c6775ff2046095da6a54e22d43b@192.168.10.109:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.0.1
Content-Length: 0

---

<--- Reliably Transmitting (NAT) to 192.168.10.102:50162 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKaRHhavVgQhNexGzeicZ856ayuCnXDk7r;received=192
.168.10.102;rport=50162
From: "1060"<sip:1060@doubango.org>;tag=8uEBBzWaND7t0A39IcpD
To: <sip:1061@doubango.org>;tag=as4715a5ca
Call-ID: b6dd849a-c20e-c682-0802-7b0edb3365a1
CSeq: 65359 INVITE
Server: Asterisk PBX 13.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0

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

<--- SIP read from WS:192.168.10.102:50163 --->
SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/WS 192.168.10.109:5060;rport=5060;branch=z9hG4bK67ebe9be
From: "1060"<sip:1060@192.168.10.109>;tag=as162314ad
To: 
<sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=Vu3uUh9OTdna
T1vVhcoZ
Call-ID: 69af5c6775ff2046095da6a54e22d43b@192.168.10.109:5060
CSeq: 102 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
[Dec  5 12:13:51] WARNING[2865][C-00000001]: chan_sip.c:24189 handle_response: 
Remote host can't match request ACK to call 
'69af5c6775ff2046095da6a54e22d43b@192.168.10.109:5060'. Giving up.

<--- SIP read from WS:192.168.10.102:50162 --->
ACK sip:1061@doubango.org SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKaRHhavVgQhNexGzeicZ856ayuCnXDk7r;rport
From: "1060"<sip:1060@doubango.org>;tag=8uEBBzWaND7t0A39IcpD
To: <sip:1061@doubango.org>;tag=as4715a5ca
Call-ID: b6dd849a-c20e-c682-0802-7b0edb3365a1
CSeq: 65359 ACK
Content-Length: 0
Max-Forwards: 70

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 
'69af5c6775ff2046095da6a54e22d43b@192.168.10.109:5060' Method: INVITE
Really destroying SIP dialog 'b6dd849a-c20e-c682-0802-7b0edb3365a1' Method: 
INVITE
Really destroying SIP dialog '824ee07e-2f1c-e5b4-9ff8-600ba6a2d0a0' Method: 
REGISTER
----------------------------------------------------------------------------

Original issue reported on code.google.com by aymen.ch...@gmail.com on 9 Dec 2014 at 3:19

GoogleCodeExporter commented 8 years ago 
You should share logs from the browser (JavaScript console). I don't think it's 
a SIP issue but something to to with the browser failing to access the 
microphone/camera. Try to restart your browser.

Original comment by boss...@yahoo.fr on 9 Dec 2014 at 3:41

GoogleCodeExporter commented 8 years ago 
Thank you for your response,
this is the log in Client 1 :
-----------------------------------------------------------------
SIPML5 API version = 1.4.217
call.htm?svn=224:1324 GET 
http://www.google-analytics.com/r/__utm.gif?utmwv=5.6.1&utms=1&utmn=5191573…ro
perability%3B&utmjid=138598472&utmredir=1&utmu=DAAAAAAAAAAAAAAAAAAAAAAE~ 
net::ERR_BLOCKED_BY_CLIENT
call.htm?svn=224:147 location=http://sipml5.org/call.htm?svn=224#
SIPml-api.js?svn=224:1 User-Agent=Mozilla/5.0 (Windows NT 6.1; WOW64) 
AppleWebKit/537.36 (KHTML, like Gecko) Chrome/39.0.2171.71 Safari/537.36
SIPml-api.js?svn=224:1 WebSocket supported = yes
SIPml-api.js?svn=224:1 Navigator friendly name = chrome
SIPml-api.js?svn=224:1 OS friendly name = windows
SIPml-api.js?svn=224:1 Have WebRTC = yes
SIPml-api.js?svn=224:1 Have GUM = yes
SIPml-api.js?svn=224:1 Engine initialized
SIPml-api.js?svn=224:1 s_websocket_server_url=ws://192.168.10.107:8088/ws
SIPml-api.js?svn=224:1 s_sip_outboundproxy_url=(null)
SIPml-api.js?svn=224:1 b_rtcweb_breaker_enabled=no
SIPml-api.js?svn=224:1 b_click2call_enabled=no
SIPml-api.js?svn=224:1 b_early_ims=yes
SIPml-api.js?svn=224:1 b_enable_media_stream_cache=no
SIPml-api.js?svn=224:1 o_bandwidth={}
SIPml-api.js?svn=224:1 o_video_size={}
SIPml-api.js?svn=224:1 SIP stack start: proxy='ns313841.ovh.net:10060', 
realm='<sip:192.168.10.107>', impi='1060', 
impu='"1060"<sip:1060@192.168.10.107>'
SIPml-api.js?svn=224:1 Connecting to 'ws://192.168.10.107:8088/ws'
SIPml-api.js?svn=224:1 ==stack event = starting
SIPml-api.js?svn=224:1 __tsip_transport_ws_onopen
SIPml-api.js?svn=224:1 ==stack event = started
SIPml-api.js?svn=224:1 State machine: 
tsip_dialog_register_Started_2_InProgress_X_oRegister
SIPml-api.js?svn=224:1 SEND: REGISTER sip:192.168.10.107 SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKTITlrtlRwcV0Ox7rIHKrHuOJiCMccPyf;rport
From: "1060"<sip:1060@192.168.10.107>;tag=e8siAnoxBWt74uU1XGD1
To: "1060"<sip:1060@192.168.10.107>
Contact: 
"1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200
;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 60e660a6-ad35-683d-96cf-5f1588d2e232
CSeq: 11813 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Supported: path

SIPml-api.js?svn=224:1 ==session event = connecting
SIPml-api.js?svn=224:1 ==session event = sent_request
SIPml-api.js?svn=224:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=224:1 recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport=50321;received=192.168.10.102;branch=z9hG4bKTITlrtlRw
cV0Ox7rIHKrHuOJiCMccPyf
From: "1060"<sip:1060@192.168.10.107>;tag=e8siAnoxBWt74uU1XGD1
To: "1060"<sip:1060@192.168.10.107>;tag=as699d9caa
Call-ID: 60e660a6-ad35-683d-96cf-5f1588d2e232
CSeq: 11813 REGISTER
Content-Length: 0
Server: Asterisk PBX 13.0.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest 
realm="192.168.10.107",nonce="09f42b34",stale=FALSE,algorithm=MD5

SIPml-api.js?svn=224:1 State machine: 
tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494
SIPml-api.js?svn=224:1 SEND: REGISTER sip:192.168.10.107 SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKZ2TyToQGO7JJxP0fSZXWDnL6rSHroZvT;rport
From: "1060"<sip:1060@192.168.10.107>;tag=e8siAnoxBWt74uU1XGD1
To: "1060"<sip:1060@192.168.10.107>
Contact: 
"1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200
;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 60e660a6-ad35-683d-96cf-5f1588d2e232
CSeq: 11814 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest 
username="1060",realm="192.168.10.107",nonce="09f42b34",uri="sip:192.168.10.107"
,response="40889121b2bb3581758834678327dbdb",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Supported: path

SIPml-api.js?svn=224:1 ==session event = sent_request
SIPml-api.js?svn=224:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=224:1 recv=SIP/2.0 200 OK
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport=50321;received=192.168.10.102;branch=z9hG4bKZ2TyToQGO
7JJxP0fSZXWDnL6rSHroZvT
From: "1060"<sip:1060@192.168.10.107>;tag=e8siAnoxBWt74uU1XGD1
To: "1060"<sip:1060@192.168.10.107>;tag=as699d9caa
Contact: 
<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200
Call-ID: 60e660a6-ad35-683d-96cf-5f1588d2e232
CSeq: 11814 REGISTER
Expires: 200
Content-Length: 0
Server: Asterisk PBX 13.0.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Date: 09 Dec 2014 16:12:04 GMT;09

SIPml-api.js?svn=224:1 State machine: 
tsip_dialog_register_InProgress_2_Connected_X_2xx
SIPml-api.js?svn=224:1 ==session event = connected
SIPml-api.js?svn=224:1 State machine: c0000_Started_2_Outgoing_X_oINVITE
SIPml-api.js?svn=224:1 PeerConnectionClass = function RTCPeerConnection() { 
[native code] } SessionDescriptionClass = function RTCSessionDescription() { 
[native code] } IceCandidateClass = function RTCIceCandidate() { [native code] }
SIPml-api.js?svn=224:1 ICE servers:[{"url":"stun:stun.l.google.com:19302"}]
SIPml-api.js?svn=224:1 ==stack event = m_permission_requested
SIPml-api.js?svn=224:1 ==session event = connecting
SIPml-api.js?svn=224:1 onGetUserMediaSuccess
SIPml-api.js?svn=224:1 createOffer
SIPml-api.js?svn=224:1 ==stack event = m_permission_accepted
SIPml-api.js?svn=224:1 ==session event = m_stream_audio_local_added
SIPml-api.js?svn=224:1 onCreateSdpSuccess
SIPml-api.js?svn=224:1 onSetLocalDescriptionSuccess
11SIPml-api.js?svn=224:1 onIceCandidate = undefined
SIPml-api.js?svn=224:1 ICE GATHERING COMPLETED!
SIPml-api.js?svn=224:1 onIceGatheringCompleted
SIPml-api.js?svn=224:1 SEND: INVITE sip:1061@192.168.10.107 SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKEur2tQKLQzAlVRiB2T43V0Vzst5mWZGw;rport
From: "1060"<sip:1060@192.168.10.107>;tag=WwNOy7e97IVUPEMGKT3I
To: <sip:1061@192.168.10.107>
Contact: 
"1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=w
s>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 650a9732-b311-5184-848b-ebcdc08d6505
CSeq: 23468 INVITE
Content-Type: application/sdp
Content-Length: 2153
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

v=0
o=- 7852237535019560000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS eTcN5UiJHzaAnVK6QfyYuTHMZKbEtVYG0UgP
m=audio 53718 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 207.162.10.166
a=rtcp:53718 IN IP4 207.162.10.166
a=candidate:2999745851 1 udp 2122260223 192.168.56.1 53717 typ host generation 0
a=candidate:2999745851 2 udp 2122260223 192.168.56.1 53717 typ host generation 0
a=candidate:3782639666 1 udp 2122194687 192.168.10.102 53718 typ host 
generation 0
a=candidate:3782639666 2 udp 2122194687 192.168.10.102 53718 typ host 
generation 0
a=candidate:4233069003 1 tcp 1518280447 192.168.56.1 0 typ host tcptype active 
generation 0
a=candidate:4233069003 2 tcp 1518280447 192.168.56.1 0 typ host tcptype active 
generation 0
a=candidate:2952101058 1 tcp 1518214911 192.168.10.102 0 typ host tcptype 
active generation 0
a=candidate:2952101058 2 tcp 1518214911 192.168.10.102 0 typ host tcptype 
active generation 0
a=candidate:771473313 1 udp 1685987071 207.162.10.166 53718 typ srflx raddr 
192.168.10.102 rport 53718 generation 0
a=candidate:771473313 2 udp 1685987071 207.162.10.166 53718 typ srflx raddr 
192.168.10.102 rport 53718 generation 0
a=ice-ufrag:Q6N5GHkmG4I5YFem
a=ice-pwd:wTPCoGGelQ4X8zOaBgvKhA95
a=ice-options:google-ice
a=fingerprint:sha-256 
B5:27:21:87:CD:21:BF:E9:92:A9:55:16:AE:FE:5F:8E:5D:7D:4F:3C:B2:EA:CA:51:CB:D2:A1
:F4:ED:03:54:E7
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
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
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:1253082076 cname:wmpuCtp9o08v282/
a=ssrc:1253082076 msid:eTcN5UiJHzaAnVK6QfyYuTHMZKbEtVYG0UgP 
075c1cec-3a8e-4ec9-bf52-22137c1c7bd4
a=ssrc:1253082076 mslabel:eTcN5UiJHzaAnVK6QfyYuTHMZKbEtVYG0UgP
a=ssrc:1253082076 label:075c1cec-3a8e-4ec9-bf52-22137c1c7bd4

SIPml-api.js?svn=224:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=224:1 recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport=50321;received=192.168.10.102;branch=z9hG4bKEur2tQKLQ
zAlVRiB2T43V0Vzst5mWZGw
From: "1060"<sip:1060@192.168.10.107>;tag=WwNOy7e97IVUPEMGKT3I
To: <sip:1061@192.168.10.107>;tag=as78bc62a3
Call-ID: 650a9732-b311-5184-848b-ebcdc08d6505
CSeq: 23468 INVITE
Content-Length: 0
Server: Asterisk PBX 13.0.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest 
realm="192.168.10.107",nonce="34baf90f",stale=FALSE,algorithm=MD5

SIPml-api.js?svn=224:1 SEND: ACK sip:1061@192.168.10.107 SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKEur2tQKLQzAlVRiB2T43V0Vzst5mWZGw;rport
From: "1060"<sip:1060@192.168.10.107>;tag=WwNOy7e97IVUPEMGKT3I
To: <sip:1061@192.168.10.107>;tag=as78bc62a3
Call-ID: 650a9732-b311-5184-848b-ebcdc08d6505
CSeq: 23468 ACK
Content-Length: 0
Max-Forwards: 70

SIPml-api.js?svn=224:1 State machine: x0000_Any_2_Any_X_i401_407_INVITE
SIPml-api.js?svn=224:1 SEND: INVITE sip:1061@192.168.10.107 SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKZOlg7N2QMs6dMiHDeEqzqqMd4Mx1Gsec;rport
From: "1060"<sip:1060@192.168.10.107>;tag=WwNOy7e97IVUPEMGKT3I
To: <sip:1061@192.168.10.107>
Contact: 
"1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=w
s>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 650a9732-b311-5184-848b-ebcdc08d6505
CSeq: 23469 INVITE
Content-Type: application/sdp
Content-Length: 2153
Max-Forwards: 70
Authorization: Digest 
username="1060",realm="192.168.10.107",nonce="34baf90f",uri="sip:1061@192.168.10
.107",response="5deaf9554e014267e636c3d574da24b1",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

v=0
o=- 7852237535019560000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS eTcN5UiJHzaAnVK6QfyYuTHMZKbEtVYG0UgP
m=audio 53718 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 207.162.10.166
a=rtcp:53718 IN IP4 207.162.10.166
a=candidate:2999745851 1 udp 2122260223 192.168.56.1 53717 typ host generation 0
a=candidate:2999745851 2 udp 2122260223 192.168.56.1 53717 typ host generation 0
a=candidate:3782639666 1 udp 2122194687 192.168.10.102 53718 typ host 
generation 0
a=candidate:3782639666 2 udp 2122194687 192.168.10.102 53718 typ host 
generation 0
a=candidate:4233069003 1 tcp 1518280447 192.168.56.1 0 typ host tcptype active 
generation 0
a=candidate:4233069003 2 tcp 1518280447 192.168.56.1 0 typ host tcptype active 
generation 0
a=candidate:2952101058 1 tcp 1518214911 192.168.10.102 0 typ host tcptype 
active generation 0
a=candidate:2952101058 2 tcp 1518214911 192.168.10.102 0 typ host tcptype 
active generation 0
a=candidate:771473313 1 udp 1685987071 207.162.10.166 53718 typ srflx raddr 
192.168.10.102 rport 53718 generation 0
a=candidate:771473313 2 udp 1685987071 207.162.10.166 53718 typ srflx raddr 
192.168.10.102 rport 53718 generation 0
a=ice-ufrag:Q6N5GHkmG4I5YFem
a=ice-pwd:wTPCoGGelQ4X8zOaBgvKhA95
a=ice-options:google-ice
a=fingerprint:sha-256 
B5:27:21:87:CD:21:BF:E9:92:A9:55:16:AE:FE:5F:8E:5D:7D:4F:3C:B2:EA:CA:51:CB:D2:A1
:F4:ED:03:54:E7
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
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
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:1253082076 cname:wmpuCtp9o08v282/
a=ssrc:1253082076 msid:eTcN5UiJHzaAnVK6QfyYuTHMZKbEtVYG0UgP 
075c1cec-3a8e-4ec9-bf52-22137c1c7bd4
a=ssrc:1253082076 mslabel:eTcN5UiJHzaAnVK6QfyYuTHMZKbEtVYG0UgP
a=ssrc:1253082076 label:075c1cec-3a8e-4ec9-bf52-22137c1c7bd4

SIPml-api.js?svn=224:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=224:1 recv=SIP/2.0 100 Trying
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport=50321;received=192.168.10.102;branch=z9hG4bKZOlg7N2QM
s6dMiHDeEqzqqMd4Mx1Gsec
From: "1060"<sip:1060@192.168.10.107>;tag=WwNOy7e97IVUPEMGKT3I
To: <sip:1061@192.168.10.107>
Contact: <sip:1061@192.168.10.107:5060;transport=WS>
Call-ID: 650a9732-b311-5184-848b-ebcdc08d6505
CSeq: 23469 INVITE
Content-Length: 0
Server: Asterisk PBX 13.0.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

SIPml-api.js?svn=224:1 State machine: x0000_Any_2_Any_X_i1xx
SIPml-api.js?svn=224:1 ==session event = i_ao_request
SIPml-api.js?svn=224:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=224:1 recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport=50321;received=192.168.10.102;branch=z9hG4bKZOlg7N2QM
s6dMiHDeEqzqqMd4Mx1Gsec
From: "1060"<sip:1060@192.168.10.107>;tag=WwNOy7e97IVUPEMGKT3I
To: <sip:1061@192.168.10.107>;tag=as7e5c8205
Contact: <sip:1061@192.168.10.107:5060;transport=WS>
Call-ID: 650a9732-b311-5184-848b-ebcdc08d6505
CSeq: 23469 INVITE
Content-Length: 0
Server: Asterisk PBX 13.0.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

SIPml-api.js?svn=224:1 State machine: x0000_Any_2_Any_X_i1xx
SIPml-api.js?svn=224:1 ==session event = i_ao_request
SIPml-api.js?svn=224:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=224:1 recv=SIP/2.0 486 Busy Here
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport=50321;received=192.168.10.102;branch=z9hG4bKZOlg7N2QM
s6dMiHDeEqzqqMd4Mx1Gsec
From: "1060"<sip:1060@192.168.10.107>;tag=WwNOy7e97IVUPEMGKT3I
To: <sip:1061@192.168.10.107>;tag=as7e5c8205
Call-ID: 650a9732-b311-5184-848b-ebcdc08d6505
CSeq: 23469 INVITE
Content-Length: 0
Server: Asterisk PBX 13.0.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21

SIPml-api.js?svn=224:1 SEND: ACK sip:1061@192.168.10.107 SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKZOlg7N2QMs6dMiHDeEqzqqMd4Mx1Gsec;rport
From: "1060"<sip:1060@192.168.10.107>;tag=WwNOy7e97IVUPEMGKT3I
To: <sip:1061@192.168.10.107>;tag=as7e5c8205
Call-ID: 650a9732-b311-5184-848b-ebcdc08d6505
CSeq: 23469 ACK
Content-Length: 0
Max-Forwards: 70

SIPml-api.js?svn=224:1 State machine: 
c0000_Outgoing_2_Terminated_X_i300_to_i699INVITE
SIPml-api.js?svn=224:1 === INVITE Dialog terminated ===
SIPml-api.js?svn=224:1 PeerConnection::stop()
SIPml-api.js?svn=224:1 ==session event = i_ao_request
SIPml-api.js?svn=224:1 ==session event = terminated
SIPml-api.js?svn=224:1 The FSM is in the final state
SIPml-api.js?svn=224:1 State machine: 
tsip_dialog_register_Connected_2_InProgress_X_oRegister
SIPml-api.js?svn=224:1 SEND: REGISTER sip:192.168.10.107 SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKWoLcKMxqlilmv7m8rBsXTkSltJbRVQrl;rport
From: "1060"<sip:1060@192.168.10.107>;tag=e8siAnoxBWt74uU1XGD1
To: "1060"<sip:1060@192.168.10.107>
Contact: 
"1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200
;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 60e660a6-ad35-683d-96cf-5f1588d2e232
CSeq: 11815 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest 
username="1060",realm="192.168.10.107",nonce="09f42b34",uri="sip:192.168.10.107"
,response="40889121b2bb3581758834678327dbdb",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

SIPml-api.js?svn=224:1 ==session event = sent_request
SIPml-api.js?svn=224:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=224:1 recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport=50321;received=192.168.10.102;branch=z9hG4bKWoLcKMxql
ilmv7m8rBsXTkSltJbRVQrl
From: "1060"<sip:1060@192.168.10.107>;tag=e8siAnoxBWt74uU1XGD1
To: "1060"<sip:1060@192.168.10.107>;tag=as0b93f116
Call-ID: 60e660a6-ad35-683d-96cf-5f1588d2e232
CSeq: 11815 REGISTER
Content-Length: 0
Server: Asterisk PBX 13.0.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest 
realm="192.168.10.107",nonce="54e0533e",stale=FALSE,algorithm=MD5

SIPml-api.js?svn=224:1 State machine: 
tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494
SIPml-api.js?svn=224:1 SEND: REGISTER sip:192.168.10.107 SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKc6f4TlbgCOH9onLxbFSvl4RuDfpc55ZO;rport
From: "1060"<sip:1060@192.168.10.107>;tag=e8siAnoxBWt74uU1XGD1
To: "1060"<sip:1060@192.168.10.107>
Contact: 
"1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200
;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 60e660a6-ad35-683d-96cf-5f1588d2e232
CSeq: 11816 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest 
username="1060",realm="192.168.10.107",nonce="54e0533e",uri="sip:192.168.10.107"
,response="d9278d2ecf2c888d8b5d78458df9cb9c",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

SIPml-api.js?svn=224:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=224:1 recv=SIP/2.0 200 OK
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport=50321;received=192.168.10.102;branch=z9hG4bKc6f4TlbgC
OH9onLxbFSvl4RuDfpc55ZO
From: "1060"<sip:1060@192.168.10.107>;tag=e8siAnoxBWt74uU1XGD1
To: "1060"<sip:1060@192.168.10.107>;tag=as0b93f116
Contact: 
<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200
Call-ID: 60e660a6-ad35-683d-96cf-5f1588d2e232
CSeq: 11816 REGISTER
Expires: 200
Content-Length: 0
Server: Asterisk PBX 13.0.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Date: 09 Dec 2014 16:13:44 GMT;09

SIPml-api.js?svn=224:1 State machine: 
tsip_dialog_register_InProgress_2_Connected_X_2xx
SIPml-api.js?svn=224:1 ==session event = sent_request
---------------------------------------------------------------------

Original comment by aymen.ch...@gmail.com on 9 Dec 2014 at 4:24

GoogleCodeExporter commented 8 years ago 
And this is what I got in Client 2 side :

---------------------------------------------------------------------
SIPml-api.js?svn=224:1 SIPML5 API version = 1.4.217
call.htm?svn=224:147 location=http://sipml5.org/call.htm?svn=224#
SIPml-api.js?svn=224:1 User-Agent=Mozilla/5.0 (Windows NT 6.1; WOW64) 
AppleWebKit/537.36 (KHTML, like Gecko) Chrome/39.0.2171.71 Safari/537.36
SIPml-api.js?svn=224:1 WebSocket supported = yes
SIPml-api.js?svn=224:1 Navigator friendly name = chrome
SIPml-api.js?svn=224:1 OS friendly name = windows
SIPml-api.js?svn=224:1 Have WebRTC = yes
SIPml-api.js?svn=224:1 Have GUM = yes
SIPml-api.js?svn=224:1 Engine initialized
SIPml-api.js?svn=224:1 s_websocket_server_url=ws://192.168.10.107:8088/ws
SIPml-api.js?svn=224:1 s_sip_outboundproxy_url=(null)
SIPml-api.js?svn=224:1 b_rtcweb_breaker_enabled=no
SIPml-api.js?svn=224:1 b_click2call_enabled=no
SIPml-api.js?svn=224:1 b_early_ims=yes
SIPml-api.js?svn=224:1 b_enable_media_stream_cache=no
SIPml-api.js?svn=224:1 o_bandwidth={}
SIPml-api.js?svn=224:1 o_video_size={}
SIPml-api.js?svn=224:1 SIP stack start: proxy='ns313841.ovh.net:11060', 
realm='<sip:192.168.10.107>', impi='1061', 
impu='"1061"<sip:1061@192.168.10.107>'
SIPml-api.js?svn=224:1 Connecting to 'ws://192.168.10.107:8088/ws'
SIPml-api.js?svn=224:1 ==stack event = starting
SIPml-api.js?svn=224:1 __tsip_transport_ws_onopen
SIPml-api.js?svn=224:1 ==stack event = started
SIPml-api.js?svn=224:1 State machine: 
tsip_dialog_register_Started_2_InProgress_X_oRegister
SIPml-api.js?svn=224:1 SEND: REGISTER sip:192.168.10.107 SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKNxWOUGz0pA5OvtxpPfXFAZcZZivQu6z7;rport
From: "1061"<sip:1061@192.168.10.107>;tag=TNBZqF8P8QH52dUNolrT
To: "1061"<sip:1061@192.168.10.107>
Contact: 
"1061"<sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200
;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 85ec2995-ece7-cfc3-b373-6778fc01ce64
CSeq: 3825 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Supported: path

SIPml-api.js?svn=224:1 ==session event = connecting
SIPml-api.js?svn=224:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=224:1 recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport=50332;received=192.168.10.102;branch=z9hG4bKNxWOUGz0p
A5OvtxpPfXFAZcZZivQu6z7
From: "1061"<sip:1061@192.168.10.107>;tag=TNBZqF8P8QH52dUNolrT
To: "1061"<sip:1061@192.168.10.107>;tag=as037187da
Call-ID: 85ec2995-ece7-cfc3-b373-6778fc01ce64
CSeq: 3825 REGISTER
Content-Length: 0
Server: Asterisk PBX 13.0.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest 
realm="192.168.10.107",nonce="50a21882",stale=FALSE,algorithm=MD5

SIPml-api.js?svn=224:1 State machine: 
tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494
SIPml-api.js?svn=224:1 SEND: REGISTER sip:192.168.10.107 SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bK44xUwl8ve2JHClZ1pTS9OZryu8ABUsO3;rport
From: "1061"<sip:1061@192.168.10.107>;tag=TNBZqF8P8QH52dUNolrT
To: "1061"<sip:1061@192.168.10.107>
Contact: 
"1061"<sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200
;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 85ec2995-ece7-cfc3-b373-6778fc01ce64
CSeq: 3826 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest 
username="1061",realm="192.168.10.107",nonce="50a21882",uri="sip:192.168.10.107"
,response="213c65440c8a2996865cb3c44b354bc3",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Supported: path

SIPml-api.js?svn=224:1 ==session event = sent_request
SIPml-api.js?svn=224:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=224:1 recv=SIP/2.0 200 OK
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport=50332;received=192.168.10.102;branch=z9hG4bK44xUwl8ve
2JHClZ1pTS9OZryu8ABUsO3
From: "1061"<sip:1061@192.168.10.107>;tag=TNBZqF8P8QH52dUNolrT
To: "1061"<sip:1061@192.168.10.107>;tag=as037187da
Contact: 
<sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200
Call-ID: 85ec2995-ece7-cfc3-b373-6778fc01ce64
CSeq: 3826 REGISTER
Expires: 200
Content-Length: 0
Server: Asterisk PBX 13.0.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Date: 09 Dec 2014 16:12:41 GMT;09

SIPml-api.js?svn=224:1 State machine: 
tsip_dialog_register_InProgress_2_Connected_X_2xx
SIPml-api.js?svn=224:1 ==session event = sent_request
SIPml-api.js?svn=224:1 ==session event = connected

undefined
SIPml-api.js?svn=224:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=224:1 recv=INVITE 
sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.10.107:5060;rport;branch=z9hG4bK72abf18a
From: "1060"<sip:1060@192.168.10.107>;tag=as49d0569a
To: <sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
Contact: <sip:1060@192.168.10.107:5060;transport=WS>
Call-ID: 3e75837226b720d97fcf777a43586485@192.168.10.107:5060
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 1375
Max-Forwards: 70
User-Agent: Asterisk PBX 13.0.1
Date: 09 Dec 2014 16:13:37 GMT;09
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

v=0
o=root 241710367 241710367 IN IP4 192.168.10.107
s=Asterisk PBX 13.0.1
c=IN IP4 192.168.10.107
b=CT:384
t=0 0
m=audio 17304 RTP/SAVPF 0 8 3 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 
116 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=0
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=connection:new
a=setup:actpass
a=fingerprint:SHA-1 80:6D:58:AD:C8:F2:7D:19:B7:04:B8:F5:86:1E:63:FF:53:76:D7:CA
a=sendrecv
m=video 14790 RTP/SAVPF 34 31 98 99 104 100
a=connection:new
a=setup:actpass
a=fingerprint:SHA-1 80:6D:58:AD:C8:F2:7D:19:B7:04:B8:F5:86:1E:63:FF:53:76:D7:CA
a=rtpmap:34 H263/90000
a=rtpmap:31 H261/90000
a=rtpmap:98 h263-1998/90000
a=rtpmap:99 H264/90000
a=rtpmap:104 MP4V-ES/90000
a=rtpmap:100 VP8/90000
a=rtcp-fb:* ccm fir
a=sendrecv

SIPml-api.js?svn=224:1 State machine: 
tsip_transac_ist_Started_2_Proceeding_X_INVITE
SIPml-api.js?svn=224:1 SEND: SIP/2.0 100 Trying (sent from the Transaction 
Layer)
Via: SIP/2.0/WS 192.168.10.107:5060;rport=5060;branch=z9hG4bK72abf18a
From: "1060"<sip:1060@192.168.10.107>;tag=as49d0569a
To: <sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
Call-ID: 3e75837226b720d97fcf777a43586485@192.168.10.107:5060
CSeq: 102 INVITE
Content-Length: 0

SIPml-api.js?svn=224:1 PeerConnectionClass = function RTCPeerConnection() { 
[native code] } SessionDescriptionClass = function RTCSessionDescription() { 
[native code] } IceCandidateClass = function RTCIceCandidate() { [native code] }
SIPml-api.js?svn=224:1 Video Contraints:{"mandatory":{},"optional":[]}
SIPml-api.js?svn=224:1 ICE servers:[{"url":"stun:stun.l.google.com:19302"}]
SIPml-api.js?svn=224:1 setRemoteDescription(offer)
v=0
o=root 241710367 241710367 IN IP4 192.168.10.107
s=Asterisk PBX 13.0.1
c=IN IP4 192.168.10.107
b=CT:384
t=0 0
m=audio 17304 RTP/SAVPF 0 8 3 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 
116 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=0
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=connection:new
a=setup:actpass
a=fingerprint:SHA-1 80:6D:58:AD:C8:F2:7D:19:B7:04:B8:F5:86:1E:63:FF:53:76:D7:CA
a=sendrecv
m=video 14790 RTP/SAVPF 34 31 98 99 104 100
a=connection:new
a=setup:actpass
a=fingerprint:SHA-1 80:6D:58:AD:C8:F2:7D:19:B7:04:B8:F5:86:1E:63:FF:53:76:D7:CA
a=rtpmap:34 H263/90000
a=rtpmap:31 H261/90000
a=rtpmap:98 h263-1998/90000
a=rtpmap:99 H264/90000
a=rtpmap:104 MP4V-ES/90000
a=rtpmap:100 VP8/90000
a=rtcp-fb:* ccm fir
a=sendrecv

SIPml-api.js?svn=224:1 State machine: s0000_Started_2_Ringing_X_iINVITE
SIPml-api.js?svn=224:1 ==stack event = m_permission_requested
SIPml-api.js?svn=224:1 onSetRemoteDescriptionError
SIPml-api.js?svn=224:1 Failed to set remote offer sdp: Called with SDP without 
ice-ufrag and ice-pwd.
SIPml-api.js?svn=224:1 State machine: 
tsip_transac_ist_Proceeding_2_Proceeding_X_1xx
SIPml-api.js?svn=224:1 SEND: SIP/2.0 180 Ringing
Via: SIP/2.0/WS 192.168.10.107:5060;rport=5060;branch=z9hG4bK72abf18a
From: "1060"<sip:1060@192.168.10.107>;tag=as49d0569a
To: 
<sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=3tIpzu7i9nORF
3g1pgFp
Contact: <sip:1061@df7jal23ls0d.invalid;transport=ws>
Call-ID: 3e75837226b720d97fcf777a43586485@192.168.10.107:5060
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

SIPml-api.js?svn=224:1 ==stack event = i_new_call
SIPml-api.js?svn=224:1 onGetUserMediaSuccess
SIPml-api.js?svn=224:1 createAnswer
SIPml-api.js?svn=224:1 ==stack event = m_permission_accepted
SIPml-api.js?svn=224:1 ==session event = m_stream_video_local_added
SIPml-api.js?svn=224:1 ==session event = m_stream_audio_local_added
SIPml-api.js?svn=224:1 onCreateSdpError
SIPml-api.js?svn=224:1 CreateAnswer can't be called before SetRemoteDescription.
SIPml-api.js?svn=224:1 State machine: s0000_Ringing_2_Terminated_X_Reject
SIPml-api.js?svn=224:1 === INVITE Dialog terminated ===
SIPml-api.js?svn=224:1 PeerConnection::stop()
SIPml-api.js?svn=224:1 State machine: 
tsip_transac_ist_Proceeding_2_Completed_X_300_to_699
SIPml-api.js?svn=224:1 SEND: SIP/2.0 603 Failed to get local SDP
Via: SIP/2.0/WS 192.168.10.107:5060;rport=5060;branch=z9hG4bK72abf18a
From: "1060"<sip:1060@192.168.10.107>;tag=as49d0569a
To: 
<sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=3tIpzu7i9nORF
3g1pgFp
Call-ID: 3e75837226b720d97fcf777a43586485@192.168.10.107:5060
CSeq: 102 INVITE
Content-Length: 0
Reason: SIP; cause=603; text="Failed to get local SDP"

SIPml-api.js?svn=224:1 This/PeerConnection is null: unexpected
SIPml-api.js?svn=224:1 ==session event = terminated
SIPml-api.js?svn=224:1 State machine: tsip_transac_ist_Any_2_Terminated_X_cancel
f64fa88e-d46e-437c-8b8a-a0a53721784a:1 GET 
blob:http%3A//sipml5.org/f64fa88e-d46e-437c-8b8a-a0a53721784a 404 (Not Found)
SIPml-api.js?svn=224:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=224:1 recv=ACK sip:1061@df7jal23ls0d.invalid;transport=ws 
SIP/2.0
Via: SIP/2.0/WS 192.168.10.107:5060;rport;branch=z9hG4bK72abf18a
From: "1060"<sip:1060@192.168.10.107>;tag=as49d0569a
To: 
<sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=3tIpzu7i9nORF
3g1pgFp
Contact: <sip:1060@192.168.10.107:5060;transport=WS>
Call-ID: 3e75837226b720d97fcf777a43586485@192.168.10.107:5060
CSeq: 102 ACK
Content-Length: 0
Max-Forwards: 70
User-Agent: Asterisk PBX 13.0.1

SIPml-api.js?svn=224:1 SEND: SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/WS 192.168.10.107:5060;rport=5060;branch=z9hG4bK72abf18a
From: "1060"<sip:1060@192.168.10.107>;tag=as49d0569a
To: 
<sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=3tIpzu7i9nORF
3g1pgFp
Call-ID: 3e75837226b720d97fcf777a43586485@192.168.10.107:5060
CSeq: 102 ACK
Content-Length: 0

SIPml-api.js?svn=224:1 State machine: 
tsip_dialog_register_Connected_2_InProgress_X_oRegister
SIPml-api.js?svn=224:1 SEND: REGISTER sip:192.168.10.107 SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKDhREQ0cqUwFP7Fvtq62pC1bzOkE6EPAb;rport
From: "1061"<sip:1061@192.168.10.107>;tag=TNBZqF8P8QH52dUNolrT
To: "1061"<sip:1061@192.168.10.107>
Contact: 
"1061"<sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200
;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 85ec2995-ece7-cfc3-b373-6778fc01ce64
CSeq: 3827 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest 
username="1061",realm="192.168.10.107",nonce="50a21882",uri="sip:192.168.10.107"
,response="213c65440c8a2996865cb3c44b354bc3",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

SIPml-api.js?svn=224:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=224:1 recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport=50332;received=192.168.10.102;branch=z9hG4bKDhREQ0cqU
wFP7Fvtq62pC1bzOkE6EPAb
From: "1061"<sip:1061@192.168.10.107>;tag=TNBZqF8P8QH52dUNolrT
To: "1061"<sip:1061@192.168.10.107>;tag=as62120c01
Call-ID: 85ec2995-ece7-cfc3-b373-6778fc01ce64
CSeq: 3827 REGISTER
Content-Length: 0
Server: Asterisk PBX 13.0.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest 
realm="192.168.10.107",nonce="0f3f44fb",stale=FALSE,algorithm=MD5

SIPml-api.js?svn=224:1 State machine: 
tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494
SIPml-api.js?svn=224:1 SEND: REGISTER sip:192.168.10.107 SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bK57oaDxpB36JO65VpuyubXudWC6Zyp0Gk;rport
From: "1061"<sip:1061@192.168.10.107>;tag=TNBZqF8P8QH52dUNolrT
To: "1061"<sip:1061@192.168.10.107>
Contact: 
"1061"<sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200
;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 85ec2995-ece7-cfc3-b373-6778fc01ce64
CSeq: 3828 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest 
username="1061",realm="192.168.10.107",nonce="0f3f44fb",uri="sip:192.168.10.107"
,response="49fa38daa6186938262311519d5d0947",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

SIPml-api.js?svn=224:1 ==session event = sent_request
SIPml-api.js?svn=224:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=224:1 recv=SIP/2.0 200 OK
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport=50332;received=192.168.10.102;branch=z9hG4bK57oaDxpB3
6JO65VpuyubXudWC6Zyp0Gk
From: "1061"<sip:1061@192.168.10.107>;tag=TNBZqF8P8QH52dUNolrT
To: "1061"<sip:1061@192.168.10.107>;tag=as62120c01
Contact: 
<sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200
Call-ID: 85ec2995-ece7-cfc3-b373-6778fc01ce64
CSeq: 3828 REGISTER
Expires: 200
Content-Length: 0
Server: Asterisk PBX 13.0.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Date: 09 Dec 2014 16:14:21 GMT;09

SIPml-api.js?svn=224:1 State machine: 
tsip_dialog_register_InProgress_2_Connected_X_2xx
SIPml-api.js?svn=224:1 ==session event = sent_request
SIPml-api.js?svn=224:1 State machine: 
tsip_dialog_register_Connected_2_InProgress_X_oRegister
SIPml-api.js?svn=224:1 SEND: REGISTER sip:192.168.10.107 SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKb5Uf352DcTOdEEkMo2I4ARGobDJPFKqT;rport
From: "1061"<sip:1061@192.168.10.107>;tag=TNBZqF8P8QH52dUNolrT
To: "1061"<sip:1061@192.168.10.107>
Contact: 
"1061"<sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200
;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 85ec2995-ece7-cfc3-b373-6778fc01ce64
CSeq: 3829 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest 
username="1061",realm="192.168.10.107",nonce="0f3f44fb",uri="sip:192.168.10.107"
,response="49fa38daa6186938262311519d5d0947",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

SIPml-api.js?svn=224:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=224:1 recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport=50332;received=192.168.10.102;branch=z9hG4bKb5Uf352Dc
TOdEEkMo2I4ARGobDJPFKqT
From: "1061"<sip:1061@192.168.10.107>;tag=TNBZqF8P8QH52dUNolrT
To: "1061"<sip:1061@192.168.10.107>;tag=as05048f11
Call-ID: 85ec2995-ece7-cfc3-b373-6778fc01ce64
CSeq: 3829 REGISTER
Content-Length: 0
Server: Asterisk PBX 13.0.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest 
realm="192.168.10.107",nonce="00dd2eee",stale=FALSE,algorithm=MD5

SIPml-api.js?svn=224:1 State machine: 
tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494
SIPml-api.js?svn=224:1 SEND: REGISTER sip:192.168.10.107 SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKpRLNb7uSrKJIrNdyxTRwW83n7PIO4H9v;rport
From: "1061"<sip:1061@192.168.10.107>;tag=TNBZqF8P8QH52dUNolrT
To: "1061"<sip:1061@192.168.10.107>
Contact: 
"1061"<sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200
;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 85ec2995-ece7-cfc3-b373-6778fc01ce64
CSeq: 3830 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest 
username="1061",realm="192.168.10.107",nonce="00dd2eee",uri="sip:192.168.10.107"
,response="d0895894ae9968d51403499dcf6a2e91",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

2SIPml-api.js?svn=224:1 ==session event = sent_request
SIPml-api.js?svn=224:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=224:1 recv=SIP/2.0 200 OK
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport=50332;received=192.168.10.102;branch=z9hG4bKpRLNb7uSr
KJIrNdyxTRwW83n7PIO4H9v
From: "1061"<sip:1061@192.168.10.107>;tag=TNBZqF8P8QH52dUNolrT
To: "1061"<sip:1061@192.168.10.107>;tag=as05048f11
Contact: 
<sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200
Call-ID: 85ec2995-ece7-cfc3-b373-6778fc01ce64
CSeq: 3830 REGISTER
Expires: 200
Content-Length: 0
Server: Asterisk PBX 13.0.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Date: 09 Dec 2014 16:16:01 GMT;09

SIPml-api.js?svn=224:1 State machine: 
tsip_dialog_register_InProgress_2_Connected_X_2xx
----------------------------------------------------------------------------

I've tried restarting the browser, and even testing from another browser 
(Chrome Canary V41) but still the same problem. 
Please, help me out !

Original comment by aymen.ch...@gmail.com on 9 Dec 2014 at 4:24

GoogleCodeExporter commented 8 years ago 
From logs:  "Failed to set remote offer sdp: Called with SDP without ice-ufrag 
and ice-pwd."
This means: INVITE from Asterisk doesn't have ICE attributes. You must enable 
ICE on Asterisk ("icesupport=yes").
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5

Original comment by boss...@yahoo.fr on 9 Dec 2014 at 5:14