ICE support problem!

Hi! I’m facing a problem of ICE support with my asterisk, In fact I want to connect a sipml5 application to asterisk in order to call a sophtfone. I have asterisk 11.11.0 installed in a centos.
I read this tow posts carefully: forums.digium.com/viewtopic.php? … a0b7ce5fe9 . (And also other posts very helpful of mr navaismo )
I have installed ces paquet: yum install libuuid libuuid-devel uuid uuid-devel
But I still have a problem of ICE support!

Rtp debug

Got  RTP packet from    192.168.1.105:8000 (type 00, seq 051632, ts 1304102107, len 000160) 
Sent RTP packet to      192.168.1.102:59567 (type 00, seq 046353, ts 1304102104, len 000160) 
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 051633, ts 1304102267, len 000160) 
Sent RTP packet to      192.168.1.102:59567 (type 00, seq 046354, ts 1304102264, len 000160) 
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 051634, ts 1304102427, len 000160) 
Sent RTP packet to      192.168.1.102:59567 (type 00, seq 046355, ts 1304102424, len 000160) 

http.conf

[general] 
enabled=yes 
bindaddr=192.168.2.114 
bindport=8088 

rtp.conf

[general] rtpstart=10000 rtpend=20000 icesupport=yes stunaddr=stun.l.google.com:19302
extension.conf

[default] 
[from-internal] 
exten => 6002,1,Dial(SIP/6002,15) 
exten => 6001,1,Dial(SIP/6001,15) [/code]

sip.conf :[code]
[general] 

context=guest 
transport=udp,ws 
rtcachefriends=yes 
allowguest=yes 
limitonpeers=yes 
callcounter=yes 
allowoverlap=n 
udpbindaddr=0.0.0.0 
;externhost=set_your_externhost_here 
externrefresh=150 
;localnet=set_your_localnet_here   ;i.e. 10.0.1.0/255.255.255.0 
disallow=all 
;allow=g729                   
allow=gsm 
allow=ulaw                     
allow=alaw                     
language=en           
callcounter=yes 
limitonpeers=yes 
callevents=yes 
useragent=Digital-Merge_UA 
realm=192.168.1.105
nat=force_rport,comedia 

[6002] 
type=friend 
secret=azerty 
host=dynamic 
context=from-internal 
disallow=all 
allow=ulaw 
allow=alaw 
;allow=g729 
;allow=gsm 
;allow=h263p 
;allow=h264 
dtmf=auto 
;videosupport=yes 
transport=ws,udp 
avpf=yes 
nat=force_rport,comedia 
callerid=6002<6002> 
username=6002 
encryption=yes 
qualify=yes 
icesupport=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 



[6001] 
host=dynamic 
context=from-internal 
username=6001 
callerid=6001 <6001> 
secret=azerty 
type=friend 
disallow=all 
icesupport=yes 
allow=ulaw 
allow=alaw 

sip debug

[code]Asterisk 11.11.0, Copyright © 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk 11.11.0 currently running on localhost (pid = 1498)
localhost*CLI> sip set debug on
SIP Debugging re-enabled
Really destroying SIP dialog ‘f7d90ff3-2a84-2e1f-fd3b-5820f9c526c8’ Method: REGISTER

<— SIP read from WS:192.168.1.102:58901 —>
INVITE sip:6001@192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKtBmgIO5IIVJkiov2ovQhBlnZSG3Rpcmf;rport
From: "amal"sip:6002@192.168.1.105;tag=pftb72pR0692WBcIOIDA
To: sip:6001@192.168.1.105
Contact: "amal"sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws;impi=6002;ha1=00b7bc0aba57392cc96d2a266ff7863f;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 9ff1e65f-9471-cd85-b3fa-d75f726bb6f0
CSeq: 60769 INVITE
Content-Type: application/sdp
Content-Length: 1519
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

v=0
o=- 1784870157862090200 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS WLw8W11ta3J7r1Fky2ENRBWlxv7c4D4QW64Q
m=audio 38180 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.1.102
a=rtcp:38180 IN IP4 192.168.1.102
a=candidate:430735571 1 udp 2122260223 192.168.1.102 38180 typ host generation 0
a=candidate:430735571 2 udp 2122260223 192.168.1.102 38180 typ host generation 0
a=candidate:1462729763 1 tcp 1518280447 192.168.1.102 0 typ host generation 0
a=candidate:1462729763 2 tcp 1518280447 192.168.1.102 0 typ host generation 0
a=ice-ufrag:4ZND0eh+ZRy6bq27
a=ice-pwd:Gjw60R8306qgUorx0Jn8Mj3s
a=ice-options:google-ice
a=fingerprint:sha-256 8B:5C:36:9A:7F:D1:6E:B6:2E:17:9B:8F:B3:AC:79:4E:6A:F9:05:16:EA:2C:EE:73:1E:4E:DF:6D:4D:F2:B9:73
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:2147547 cname:h08tJ8pteYyXvhsh
a=ssrc:2147547 msid:WLw8W11ta3J7r1Fky2ENRBWlxv7c4D4QW64Q 45c3bbf4-9c0c-44e8-897b-858033cfb429
a=ssrc:2147547 mslabel:WLw8W11ta3J7r1Fky2ENRBWlxv7c4D4QW64Q
a=ssrc:2147547 label:45c3bbf4-9c0c-44e8-897b-858033cfb429
<------------->
— (12 headers 38 lines) —
Using INVITE request as basis request - 9ff1e65f-9471-cd85-b3fa-d75f726bb6f0
Found peer ‘6002’ for ‘6002’ from 192.168.1.102:58901

<— Reliably Transmitting (NAT) to 192.168.1.102:58901 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKtBmgIO5IIVJkiov2ovQhBlnZSG3Rpcmf;received=192.168.1.102;rport=58901
From: "amal"sip:6002@192.168.1.105;tag=pftb72pR0692WBcIOIDA
To: sip:6001@192.168.1.105;tag=as3d15adc1
Call-ID: 9ff1e65f-9471-cd85-b3fa-d75f726bb6f0
CSeq: 60769 INVITE
Server: Digital-Merge_UA
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“192.168.1.105”, nonce="1a519962"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘9ff1e65f-9471-cd85-b3fa-d75f726bb6f0’ in 6400 ms (Method: INVITE)

<— SIP read from WS:192.168.1.102:58901 —>
ACK sip:6001@192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKtBmgIO5IIVJkiov2ovQhBlnZSG3Rpcmf;rport
From: "amal"sip:6002@192.168.1.105;tag=pftb72pR0692WBcIOIDA
To: sip:6001@192.168.1.105;tag=as3d15adc1
Call-ID: 9ff1e65f-9471-cd85-b3fa-d75f726bb6f0
CSeq: 60769 ACK
Content-Length: 0
Max-Forwards: 70

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

<— SIP read from WS:192.168.1.102:58901 —>
INVITE sip:6001@192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKPOp5RSMaQC6vq5wmATmAnf062XF38HaF;rport
From: "amal"sip:6002@192.168.1.105;tag=pftb72pR0692WBcIOIDA
To: sip:6001@192.168.1.105
Contact: "amal"sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws;impi=6002;ha1=00b7bc0aba57392cc96d2a266ff7863f;+g.oma.sip-im;+sip.ice;language=“en,fr"
Call-ID: 9ff1e65f-9471-cd85-b3fa-d75f726bb6f0
CSeq: 60770 INVITE
Content-Type: application/sdp
Content-Length: 1519
Max-Forwards: 70
Authorization: Digest username=“6002”,realm=“192.168.1.105”,nonce=“1a519962”,uri="sip:6001@192.168.1.105”,response=“2110662960425a99fe224b3c37d3a45c”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

v=0
o=- 1784870157862090200 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS WLw8W11ta3J7r1Fky2ENRBWlxv7c4D4QW64Q
m=audio 38180 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.1.102
a=rtcp:38180 IN IP4 192.168.1.102
a=candidate:430735571 1 udp 2122260223 192.168.1.102 38180 typ host generation 0
a=candidate:430735571 2 udp 2122260223 192.168.1.102 38180 typ host generation 0
a=candidate:1462729763 1 tcp 1518280447 192.168.1.102 0 typ host generation 0
a=candidate:1462729763 2 tcp 1518280447 192.168.1.102 0 typ host generation 0
a=ice-ufrag:4ZND0eh+ZRy6bq27
a=ice-pwd:Gjw60R8306qgUorx0Jn8Mj3s
a=ice-options:google-ice
a=fingerprint:sha-256 8B:5C:36:9A:7F:D1:6E:B6:2E:17:9B:8F:B3:AC:79:4E:6A:F9:05:16:EA:2C:EE:73:1E:4E:DF:6D:4D:F2:B9:73
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:2147547 cname:h08tJ8pteYyXvhsh
a=ssrc:2147547 msid:WLw8W11ta3J7r1Fky2ENRBWlxv7c4D4QW64Q 45c3bbf4-9c0c-44e8-897b-858033cfb429
a=ssrc:2147547 mslabel:WLw8W11ta3J7r1Fky2ENRBWlxv7c4D4QW64Q
a=ssrc:2147547 label:45c3bbf4-9c0c-44e8-897b-858033cfb429
<------------->
— (13 headers 38 lines) —
Using INVITE request as basis request - 9ff1e65f-9471-cd85-b3fa-d75f726bb6f0
Found peer ‘6002’ for ‘6002’ from 192.168.1.102:58901
== 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 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 unknown media 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
Capabilities: us - (ulaw|alaw), 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 192.168.1.102:38180
Looking for 6001 in from-internal (domain 192.168.1.105)
list_route: hop: sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws

<— Transmitting (NAT) to 192.168.1.102:58901 —>
SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKPOp5RSMaQC6vq5wmATmAnf062XF38HaF;received=192.168.1.102;rport=58901
From: "amal"sip:6002@192.168.1.105;tag=pftb72pR0692WBcIOIDA
To: sip:6001@192.168.1.105
Call-ID: 9ff1e65f-9471-cd85-b3fa-d75f726bb6f0
CSeq: 60770 INVITE
Server: Digital-Merge_UA
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:6001@192.168.1.105:5060;transport=WS
Content-Length: 0

<------------>
– Executing [6001@from-internal:1] Dial(“SIP/6002-00000008”, “SIP/6001,15”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 14698
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.105:5061:
INVITE sip:6001@192.168.1.105:5061;rinstance=8fcb7712fb70d213;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK32ac72bf;rport
Max-Forwards: 70
From: “6002” sip:6002@192.168.1.105;tag=as0eb1d78a
To: sip:6001@192.168.1.105:5061;rinstance=8fcb7712fb70d213;transport=UDP
Contact: sip:6002@192.168.1.105:5060
Call-ID: 3229b0841109dddc363fd7ce290870cb@192.168.1.105:5060
CSeq: 102 INVITE
User-Agent: Digital-Merge_UA
Date: Mon, 18 Aug 2014 13:57:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 1629463152 1629463152 IN IP4 192.168.1.105
s=Asterisk PBX 11.11.0
c=IN IP4 192.168.1.105
t=0 0
m=audio 14698 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


-- Called SIP/6001 

<— SIP read from UDP:192.168.1.105:5061 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK32ac72bf;rport=5060
To: sip:6001@192.168.1.105:5061;rinstance=8fcb7712fb70d213;transport=UDP
From: “6002” sip:6002@192.168.1.105;tag=as0eb1d78a
Call-ID: 3229b0841109dddc363fd7ce290870cb@192.168.1.105:5060
CSeq: 102 INVITE
Content-Length: 0

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

<— SIP read from UDP:192.168.1.105:5061 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK32ac72bf;rport=5060
Contact: sip:6001@197.31.95.108:5061;rinstance=8fcb7712fb70d213;transport=UDP
To: sip:6001@192.168.1.105:5061;rinstance=8fcb7712fb70d213;transport=UDP;tag=ff47ce61
From: "6002"sip:6002@192.168.1.105;tag=as0eb1d78a
Call-ID: 3229b0841109dddc363fd7ce290870cb@192.168.1.105:5060
CSeq: 102 INVITE
User-Agent: Zoiper r21155
Content-Length: 0

<------------->
— (9 headers 0 lines) —
list_route: hop: sip:6001@197.31.95.108:5061;rinstance=8fcb7712fb70d213;transport=UDP
– SIP/6001-00000009 is ringing

<— Transmitting (NAT) to 192.168.1.102:58901 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKPOp5RSMaQC6vq5wmATmAnf062XF38HaF;received=192.168.1.102;rport=58901
From: "amal"sip:6002@192.168.1.105;tag=pftb72pR0692WBcIOIDA
To: sip:6001@192.168.1.105;tag=as074b1e6e
Call-ID: 9ff1e65f-9471-cd85-b3fa-d75f726bb6f0
CSeq: 60770 INVITE
Server: Digital-Merge_UA
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:6001@192.168.1.105:5060;transport=WS
Content-Length: 0 [/code]

there is the result of the command: rpm -qa | grep uuid && echo && ldd /usr/lib/asterisk/modules/res_rtp_asterisk.so && echo && ls -lha /lib/libuu* && cat /etc/issue

[code]libuuid-2.17.2-12.14.el6_5.i686
libuuid-devel-2.17.2-12.14.el6_5.i686
uuid-devel-1.6.1-10.el6.i686
uuid-1.6.1-10.el6.i686

linux-gate.so.1 =>  (0x002ef000) 
libuuid.so.1 => /lib/libuuid.so.1 (0x00298000) 
libpthread.so.0 => /lib/libpthread.so.0 (0x0038c000) 
libc.so.6 => /lib/libc.so.6 (0x007ab000) 
/lib/ld-linux.so.2 (0x004b0000) 

lrwxrwxrwx. 1 root root 16 Aug 13 13:16 /lib/libuuid.so.1 -> libuuid.so.1.3.0
-rwxr-xr-x. 1 root root 15K Apr 28 08:05 /lib/libuuid.so.1.3.0
CentOS release 6.5 (Final)
Kernel \r on an \m [/code]

this is the chrome debug:

[code]YOUR ARE USING DEBUG CODE. PLEASE USE CODE UNDER ‘release’ FOLDER or check https://code.google.com/p/sipml5/wiki/FAQ#How_to_reduce_the_size_of_the_scripts_before_deploying for more information on how to amalgamate the code.
Uncaught ReferenceError: tmedia_session_jsep01 is not defined 192.168.1.102/myphone/SIPml-api.js?svn=224:29
location=http://192.168.1.102/myphone/call.htm?svn=224#
User-Agent=Mozilla/5.0 (X11; Linux i686) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/35.0.1916.153 Safari/537.36
WebSocket supported = yes
Navigator friendly name = chrome
OS friendly name = linux
Have WebRTC = yes
Have GUM = yes
Engine initialized
s_websocket_server_url=ws://192.168.1.105:8088/ws
s_sip_outboundproxy_url=(null)
b_rtcweb_breaker_enabled=yes
b_click2call_enabled=no
b_early_ims=yes
b_enable_media_stream_cache=no
o_bandwidth={}
o_video_size={}
SIP stack start: proxy=‘ns313841.ovh.net:12062’, realm=‘sip:192.168.1.105’, impi=‘6002’, impu=’"amal"sip:6002@192.168.1.105
Connecting to ‘ws://192.168.1.105:8088/ws’
==stack event = starting
__tsip_transport_ws_onopen
==stack event = started
State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister
SEND: REGISTER sip:192.168.1.105 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKMq5e2Ufy5UkD2SP6JQXaosLnnu8qAK7T;rport From: "amal"sip:6002@192.168.1.105;tag=vNwF1TRFNTufsfTxCw6e To: "amal"sip:6002@192.168.1.105 Contact: "amal"sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language=“en,fr” Call-ID: 37db0c94-d2df-b83b-5d4a-0a5210242ab6 CSeq: 4743 REGISTER Content-Length: 0 Max-Forwards: 70 User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18 Organization: Doubango Telecom Supported: path
==session event = connecting
==session event = sent_request
__tsip_transport_ws_onmessage
recv=SIP/2.0 401 Unauthorized Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=59554;received=192.168.1.102;branch=z9hG4bKMq5e2Ufy5UkD2SP6JQXaosLnnu8qAK7T From: "amal"sip:6002@192.168.1.105;tag=vNwF1TRFNTufsfTxCw6e To: "amal"sip:6002@192.168.1.105;tag=as721657d8 Call-ID: 37db0c94-d2df-b83b-5d4a-0a5210242ab6 CSeq: 4743 REGISTER Content-Length: 0 Server: Digital-Merge_UA Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer WWW-Authenticate: Digest realm=“192.168.1.105”,nonce=“6df95fec”,stale=FALSE,algorithm=MD5
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494
SEND: REGISTER sip:192.168.1.105 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKMiVmptrZfkOOJm2VerJfeWCUHCSxgCEq;rport From: "amal"sip:6002@192.168.1.105;tag=vNwF1TRFNTufsfTxCw6e To: "amal"sip:6002@192.168.1.105 Contact: "amal"sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language=“en,fr” Call-ID: 37db0c94-d2df-b83b-5d4a-0a5210242ab6 CSeq: 4744 REGISTER Content-Length: 0 Max-Forwards: 70 Authorization: Digest username=“6002”,realm=“192.168.1.105”,nonce=“6df95fec”,uri=“sip:192.168.1.105”,response=“c2fa2b68ecd61704ef86bb6491698dd4”,algorithm=MD5 User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18 Organization: Doubango Telecom Supported: path
==session event = sent_request
__tsip_transport_ws_onmessage
recv=OPTIONS sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.1.105:5060;rport;branch=z9hG4bK01a6db6c From: "asterisk"sip:asterisk@192.168.1.105;tag=as72fde194 To: sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws Contact: sip:asterisk@192.168.1.105:5060;transport=WS Call-ID: 473f70d07abc25c42160cdbd5602ed28@192.168.1.105:5060 CSeq: 102 OPTIONS Content-Length: 0 Max-Forwards: 70 User-Agent: Digital-Merge_UA Date: 18 Aug 2014 14:10:25 GMT;18 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer

  1. Not implemented
    SEND: SIP/2.0 405 Method Not Allowed Via: SIP/2.0/WS 192.168.1.105:5060;rport=5060;branch=z9hG4bK01a6db6c From: "asterisk"sip:asterisk@192.168.1.105;tag=as72fde194 To: sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws Call-ID: 473f70d07abc25c42160cdbd5602ed28@192.168.1.105:5060 CSeq: 102 OPTIONS Content-Length: 0
    __tsip_transport_ws_onmessage
    recv=SIP/2.0 200 OK Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=59554;received=192.168.1.102;branch=z9hG4bKMiVmptrZfkOOJm2VerJfeWCUHCSxgCEq From: "amal"sip:6002@192.168.1.105;tag=vNwF1TRFNTufsfTxCw6e To: "amal"sip:6002@192.168.1.105;tag=as721657d8 Contact: sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200 Call-ID: 37db0c94-d2df-b83b-5d4a-0a5210242ab6 CSeq: 4744 REGISTER Expires: 200 Content-Length: 0 Server: Digital-Merge_UA Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer Date: 18 Aug 2014 14:10:25 GMT;18
    State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx
    ==session event = connected
    State machine: c0000_Started_2_Outgoing_X_oINVITE
    PeerConnectionClass = function RTCPeerConnection() { [native code] } SessionDescriptionClass = function RTCSessionDescription() { [native code] } IceCandidateClass = function RTCIceCandidate() { [native code] }
    ICE servers:[{“url”:“stun:null”}]
    ==stack event = m_permission_requested
    ==session event = connecting
    onGetUserMediaSuccess
    createOffer
    onCreateSdpSuccess
    ==stack event = m_permission_accepted
    ==session event = m_stream_audio_local_added
    onSetLocalDescriptionSuccess
    5onIceCandidate = undefined
    ICE GATHERING COMPLETED!
    onIceGatheringCompleted
    SEND: INVITE sip:6001@192.168.1.105 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKGQPRMQs3Tu8mlLNH6VXDaHUlTG8xNrEE;rport From: "amal"sip:6002@192.168.1.105;tag=U7dn8KonDu5BWZh4dJTz To: sip:6001@192.168.1.105 Contact: "amal"sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws;impi=6002;ha1=00b7bc0aba57392cc96d2a266ff7863f;+g.oma.sip-im;+sip.ice;language=“en,fr” Call-ID: 39b22dfb-b28b-8247-51fb-0bd8babcf0e3 CSeq: 54755 INVITE Content-Type: application/sdp Content-Length: 1531 Max-Forwards: 70 User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18 Organization: Doubango Telecom v=0 o=- 6044190987274923000 2 IN IP4 127.0.0.1 s=Doubango Telecom - chrome t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS NsnymbMVTwDbsvJf5VW44CkzuDFvhh1n5e6m m=audio 38820 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 192.168.1.102 a=rtcp:38820 IN IP4 192.168.1.102 a=candidate:430735571 1 udp 2122260223 192.168.1.102 38820 typ host generation 0 a=candidate:430735571 2 udp 2122260223 192.168.1.102 38820 typ host generation 0 a=candidate:1462729763 1 tcp 1518280447 192.168.1.102 0 typ host generation 0 a=candidate:1462729763 2 tcp 1518280447 192.168.1.102 0 typ host generation 0 a=ice-ufrag:5jMOwc7aT9m06vRh a=ice-pwd:JAbeQG7uf/gwjczgsw1A8Eo6 a=ice-options:google-ice a=fingerprint:sha-256 8B:5C:36:9A:7F:D1:6E:B6:2E:17:9B:8F:B3:AC:79:4E:6A:F9:05:16:EA:2C:EE:73:1E:4E:DF:6D:4D:F2:B9:73 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:4142699940 cname:EKVFRXb9M3xHZxn5 a=ssrc:4142699940 msid:NsnymbMVTwDbsvJf5VW44CkzuDFvhh1n5e6m 4ea69ef7-7872-4550-b4a9-7dfd5fbd2ee0 a=ssrc:4142699940 mslabel:NsnymbMVTwDbsvJf5VW44CkzuDFvhh1n5e6m a=ssrc:4142699940 label:4ea69ef7-7872-4550-b4a9-7dfd5fbd2ee0
    __tsip_transport_ws_onmessage
    recv=SIP/2.0 401 Unauthorized Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=59554;received=192.168.1.102;branch=z9hG4bKGQPRMQs3Tu8mlLNH6VXDaHUlTG8xNrEE From: "amal"sip:6002@192.168.1.105;tag=U7dn8KonDu5BWZh4dJTz To: sip:6001@192.168.1.105;tag=as1947f0cb Call-ID: 39b22dfb-b28b-8247-51fb-0bd8babcf0e3 CSeq: 54755 INVITE Content-Length: 0 Server: Digital-Merge_UA Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer WWW-Authenticate: Digest realm=“192.168.1.105”,nonce=“392cafe4”,stale=FALSE,algorithm=MD5
    SEND: ACK sip:6001@192.168.1.105 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKGQPRMQs3Tu8mlLNH6VXDaHUlTG8xNrEE;rport From: "amal"sip:6002@192.168.1.105;tag=U7dn8KonDu5BWZh4dJTz To: sip:6001@192.168.1.105;tag=as1947f0cb Call-ID: 39b22dfb-b28b-8247-51fb-0bd8babcf0e3 CSeq: 54755 ACK Content-Length: 0 Max-Forwards: 70
    State machine: x0000_Any_2_Any_X_i401_407_INVITE
    SEND: INVITE sip:6001@192.168.1.105 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKXagwrbhSxfP4kV519oksCQcVPrec4rOQ;rport From: "amal"sip:6002@192.168.1.105;tag=U7dn8KonDu5BWZh4dJTz To: sip:6001@192.168.1.105 Contact: “amal"sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws;impi=6002;ha1=00b7bc0aba57392cc96d2a266ff7863f;+g.oma.sip-im;+sip.ice;language=“en,fr” Call-ID: 39b22dfb-b28b-8247-51fb-0bd8babcf0e3 CSeq: 54756 INVITE Content-Type: application/sdp Content-Length: 1531 Max-Forwards: 70 Authorization: Digest username=“6002”,realm=“192.168.1.105”,nonce=“392cafe4”,uri="sip:6001@192.168.1.105”,response=“d580ffcd49a84d8464447607dd2dd427”,algorithm=MD5 User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18 Organization: Doubango Telecom v=0 o=- 6044190987274923000 2 IN IP4 127.0.0.1 s=Doubango Telecom - chrome t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS NsnymbMVTwDbsvJf5VW44CkzuDFvhh1n5e6m m=audio 38820 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 192.168.1.102 a=rtcp:38820 IN IP4 192.168.1.102 a=candidate:430735571 1 udp 2122260223 192.168.1.102 38820 typ host generation 0 a=candidate:430735571 2 udp 2122260223 192.168.1.102 38820 typ host generation 0 a=candidate:1462729763 1 tcp 1518280447 192.168.1.102 0 typ host generation 0 a=candidate:1462729763 2 tcp 1518280447 192.168.1.102 0 typ host generation 0 a=ice-ufrag:5jMOwc7aT9m06vRh a=ice-pwd:JAbeQG7uf/gwjczgsw1A8Eo6 a=ice-options:google-ice a=fingerprint:sha-256 8B:5C:36:9A:7F:D1:6E:B6:2E:17:9B:8F:B3:AC:79:4E:6A:F9:05:16:EA:2C:EE:73:1E:4E:DF:6D:4D:F2:B9:73 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:4142699940 cname:EKVFRXb9M3xHZxn5 a=ssrc:4142699940 msid:NsnymbMVTwDbsvJf5VW44CkzuDFvhh1n5e6m 4ea69ef7-7872-4550-b4a9-7dfd5fbd2ee0 a=ssrc:4142699940 mslabel:NsnymbMVTwDbsvJf5VW44CkzuDFvhh1n5e6m a=ssrc:4142699940 label:4ea69ef7-7872-4550-b4a9-7dfd5fbd2ee0
    __tsip_transport_ws_onmessage
    recv=SIP/2.0 100 Trying Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=59554;received=192.168.1.102;branch=z9hG4bKXagwrbhSxfP4kV519oksCQcVPrec4rOQ From: "amal"sip:6002@192.168.1.105;tag=U7dn8KonDu5BWZh4dJTz To: sip:6001@192.168.1.105 Contact: sip:6001@192.168.1.105:5060;transport=WS Call-ID: 39b22dfb-b28b-8247-51fb-0bd8babcf0e3 CSeq: 54756 INVITE Content-Length: 0 Server: Digital-Merge_UA Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer
    State machine: x0000_Any_2_Any_X_i1xx
    ==session event = i_ao_request
    __tsip_transport_ws_onmessage
    recv=SIP/2.0 180 Ringing Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=59554;received=192.168.1.102;branch=z9hG4bKXagwrbhSxfP4kV519oksCQcVPrec4rOQ From: "amal"sip:6002@192.168.1.105;tag=U7dn8KonDu5BWZh4dJTz To: sip:6001@192.168.1.105;tag=as61a318c4 Contact: sip:6001@192.168.1.105:5060;transport=WS Call-ID: 39b22dfb-b28b-8247-51fb-0bd8babcf0e3 CSeq: 54756 INVITE Content-Length: 0 Server: Digital-Merge_UA Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer
    State machine: x0000_Any_2_Any_X_i1xx
    ==session event = i_ao_request
    __tsip_transport_ws_onmessage
    recv=SIP/2.0 200 OK Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=59554;received=192.168.1.102;branch=z9hG4bKXagwrbhSxfP4kV519oksCQcVPrec4rOQ From: "amal"sip:6002@192.168.1.105;tag=U7dn8KonDu5BWZh4dJTz To: sip:6001@192.168.1.105;tag=as61a318c4 Contact: sip:6001@192.168.1.105:5060;transport=WS Call-ID: 39b22dfb-b28b-8247-51fb-0bd8babcf0e3 CSeq: 54756 INVITE Content-Type: application/sdp Content-Length: 425 Server: Digital-Merge_UA Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer v=0 o=root 1171354543 1171354543 IN IP4 192.168.1.105 s=Asterisk PBX 11.11.0 c=IN IP4 192.168.1.105 t=0 0 m=audio 13154 UDP/TLS/RTP/SAVPF 0 8 126 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:126 telephone-event/8000 a=fmtp:126 0-16 a=ptime:20 a=connection:new a=setup:active a=fingerprint:SHA-256 34:02:5C:C2:93:7A:36:01:F0:56:F0:18:E0:8E:09:F7:06:9B:1D:FD:0C:1A:8B:E3:C3:EE:1E:85:66:22:EB:F0 a=sendrecv
    State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE
    setRemoteDescription(answer) v=0 o=root 1171354543 1171354543 IN IP4 192.168.1.105 s=Asterisk PBX 11.11.0 c=IN IP4 192.168.1.105 t=0 0 m=audio 13154 RTP/SAVPF 0 8 126 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:126 telephone-event/8000 a=fmtp:126 0-16 a=ptime:20 a=connection:new a=setup:active a=fingerprint:SHA-256 34:02:5C:C2:93:7A:36:01:F0:56:F0:18:E0:8E:09:F7:06:9B:1D:FD:0C:1A:8B:E3:C3:EE:1E:85:66:22:EB:F0 a=sendrecv
    SEND: ACK sip:6001@192.168.1.105:5060;transport=WS SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKkWFLStF2R7D6mJ5Vg0go;rport From: "amal"sip:6002@192.168.1.105;tag=U7dn8KonDu5BWZh4dJTz To: sip:6001@192.168.1.105;tag=as61a318c4 Contact: "amal"sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws;+g.oma.sip-im;+sip.ice;language=“en,fr” Call-ID: 39b22dfb-b28b-8247-51fb-0bd8babcf0e3 CSeq: 54756 ACK Content-Length: 0 Max-Forwards: 70 Authorization: Digest username=“6002”,realm=“192.168.1.105”,nonce=“392cafe4”,uri=“sip:6001@192.168.1.105:5060;transport=WS”,response=“95f4a7089ef7a4549bfda59f2991a01b”,algorithm=MD5 User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18 Organization: Doubango Telecom
    onSetRemoteDescriptionError
  2. Failed to set remote answer sdp: Called with SDP without ice-ufrag and ice-pwd.
    ==session event = m_early_media
    ==session event = connected
    __tsip_transport_ws_onmessage
    recv=BYE sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.1.105:5060;rport;branch=z9hG4bK010bd840 From: sip:6001@192.168.1.105;tag=as61a318c4 To: "amal"sip:6002@192.168.1.105;tag=U7dn8KonDu5BWZh4dJTz Call-ID: 39b22dfb-b28b-8247-51fb-0bd8babcf0e3 CSeq: 102 BYE Content-Length: 0 Max-Forwards: 70 User-Agent: Digital-Merge_UA Proxy-Authorization: Digest username=“6002”,realm=“192.168.1.105”,nonce=“392cafe4”,uri=“sip:192.168.1.105”,response=“a798a0a6b15aee5d80400a64cfad1320”,algorithm=MD5 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16
    State machine: x0000_Any_2_Terminated_X_iBYE
    === INVITE Dialog terminated ===
    PeerConnection::stop()
    SEND: SIP/2.0 200 OK Via: SIP/2.0/WS 192.168.1.105:5060;rport=5060;branch=z9hG4bK010bd840 From: sip:6001@192.168.1.105;tag=as61a318c4 To: "amal"sip:6002@192.168.1.105;tag=U7dn8KonDu5BWZh4dJTz Contact: sip:6002@df7jal23ls0d.invalid;transport=ws Call-ID: 39b22dfb-b28b-8247-51fb-0bd8babcf0e3 CSeq: 102 BYE Content-Length: 0
    ==session event = terminated
    The FSM is in the final state
    __tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
    recv=OPTIONS sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.1.105:5060;rport;branch=z9hG4bK2eb84a0b From: "asterisk"sip:asterisk@192.168.1.105;tag=as14e23d7d To: sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws Contact: sip:asterisk@192.168.1.105:5060;transport=WS Call-ID: 690e6e101ce3467951f128231637f2e6@192.168.1.105:5060 CSeq: 102 OPTIONS Content-Length: 0 Max-Forwards: 70 User-Agent: Digital-Merge_UA Date: 18 Aug 2014 14:11:25 GMT;18 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer tsk_utils.js?svn=224:116
  3. Not implemented tsk_utils.js?svn=224:128
    SEND: SIP/2.0 405 Method Not Allowed Via: SIP/2.0/WS 192.168.1.105:5060;rport=5060;branch=z9hG4bK2eb84a0b From: "asterisk"sip:asterisk@192.168.1.105;tag=as14e23d7d To: sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws Call-ID: 690e6e101ce3467951f128231637f2e6@192.168.1.105:5060 CSeq: 102 OPTIONS Content-Length: 0 [/code]

I will be really grateful for your help!

64 or 32 bits machine? If 64 try building a Virtual Machine with 32 bits system, somewhere I saw(once) a issue with 64bits machines, maybe there is a bug but this need to be confirmed.

It’s Centos 6.5 32 bits

If I install webrtc2sip is there a chance to get this work? I’m really stuck! I have to finish the project this Saturday to succeed this year :frowning:

Yes you can try that. But compile the gateway is little difficult than native way.

otherwise, is there any other thing I could try, replace, or correct? Thank you so much for all your response

As last try before compilingvthe GW you can wipe out your OS install and update the latest CentOS, install all again and try. I was testing in 32bits machine without Issues.