Failed to set remote answer sdp: Called with SDP without ice

samira · August 2, 2014, 9:37pm

Hello! I can’t get sound when I make a call between Sipml5 and my webphone(Zoiper), I have Asterisk installed and configured as follows:
sip.conf:

[general]
udpbindaddr=0.0.0.0:5060
realm=192.168.1.105
transport=udp,ws
[6001]
host=dynamic
context=from-internal
username=6001
secret=azerty
type=friend
disallow=all
allow=ulaw
allow=alaw
[6002]
type=friend
context=from-internal
host=dynamic
username=6002
secret=azerty
encryption=yes
avpf=yes
icesupport=yes
directmedia=no
disallow=all
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass

http.conf

[general]
enabled=yes
bindport=8088
bindaddr=0.0.0.0
enablestatic=yes

rtp.conf:

[general]
icesupport=yes
stunaddr=

extensions.conf

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

rtp debug:

Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032398, ts 2264605027, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028050, ts 2264605024, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032399, ts 2264605187, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028051, ts 2264605184, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032400, ts 2264605347, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028052, ts 2264605344, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032401, ts 2264605507, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028053, ts 2264605504, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032402, ts 2264605667, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028054, ts 2264605664, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032403, ts 2264605827, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028055, ts 2264605824, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032404, ts 2264605987, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028056, ts 2264605984, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032405, ts 2264606147, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028057, ts 2264606144, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032406, ts 2264606307, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028058, ts 2264606304, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032407, ts 2264606467, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028059, ts 2264606464, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032408, ts 2264606627, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028060, ts 2264606624, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032409, ts 2264606787, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028061, ts 2264606784, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032410, ts 2264606947, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028062, ts 2264606944, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032411, ts 2264607107, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028063, ts 2264607104, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032412, ts 2264607267, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028064, ts 2264607264, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032413, ts 2264607427, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028065, ts 2264607424, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032414, ts 2264607587, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028066, ts 2264607584, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032415, ts 2264607747, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028067, ts 2264607744, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032416, ts 2264607907, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028068, ts 2264607904, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032417, ts 2264608067, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028069, ts 2264608064, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032418, ts 2264608227, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028070, ts 2264608224, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032419, ts 2264608387, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028071, ts 2264608384, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032420, ts 2264608547, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028072, ts 2264608544, len 000160)
Got  RTP packet from    192.168.1.105:8000 (type 00, seq 032421, ts 2264608707, len 000160)
Sent RTP packet to      192.168.1.104:52117 (type 00, seq 028073, ts 2264608704, len 000160)
....

my sip debug:

<--- SIP read from WS:192.168.1.104:59439 ---> 
INVITE sip:6001@192.168.1.105 SIP/2.0 
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKDvVLWqTerDmfvU7ntzHJQt1sO5nFm5S2;rport 
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3 
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: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0 
CSeq: 62542 INVITE 
Content-Type: application/sdp 
Content-Length: 1533 
Max-Forwards: 70 
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18 
Organization: Doubango Telecom 
v=0 
o=- 2577394367678132000 2 IN IP4 127.0.0.1 
s=Doubango Telecom - chrome 
t=0 0 
a=group:BUNDLE audio 
a=msid-semantic: WMS HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM 
m=audio 42985 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 
c=IN IP4 192.168.1.104 
a=rtcp:42985 IN IP4 192.168.1.104 
a=candidate:1019731727 1 udp 2122260223 192.168.1.104 42985 typ host generation 0 
a=candidate:1019731727 2 udp 2122260223 192.168.1.104 42985 typ host generation 0 
a=candidate:1917068287 1 tcp 1518280447 192.168.1.104 0 typ host generation 0 
a=candidate:1917068287 2 tcp 1518280447 192.168.1.104 0 typ host generation 0 
a=ice-ufrag:fHkUinzS5ohjJhal 
a=ice-pwd:6Q37zMp1s4jjos+z63Gy3u+3 
a=ice-options:google-ice 
a=fingerprint:sha-256 17:AE:2A:F4:FE:8A:C9:AF:E0:32:1C:93:AA:58:C9:91:79:4A:6B:2A:63:5B:67:F2:81:1A:2E:8A:EC:DC:33:D4 
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:1974193999 cname:bB9ulmmnutOJVAtz 
a=ssrc:1974193999 msid:HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM d16640b3-40b2-4f7d-95cc-5808611d201a 
a=ssrc:1974193999 mslabel:HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM 
a=ssrc:1974193999 label:d16640b3-40b2-4f7d-95cc-5808611d201a 
<-------------> 
--- (12 headers 38 lines) --- 
Using INVITE request as basis request - c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0 
Found peer '6002' for '6002' from 192.168.1.104:59439 

<--- Reliably Transmitting (no NAT) to 192.168.1.104:5060 ---> 
SIP/2.0 401 Unauthorized 
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKDvVLWqTerDmfvU7ntzHJQt1sO5nFm5S2;rport;received=192.168.1.104 
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3 
To: <sip:6001@192.168.1.105>;tag=as03c5c0eb 
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0 
CSeq: 62542 INVITE 
Server: Asterisk PBX 11.11.0 
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="6ad3c739" 
Content-Length: 0 


<------------> 
Scheduling destruction of SIP dialog 'c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0' in 32000 ms (Method: INVITE) 

<--- SIP read from WS:192.168.1.104:59439 ---> 
ACK sip:6001@192.168.1.105 SIP/2.0 
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKDvVLWqTerDmfvU7ntzHJQt1sO5nFm5S2;rport 
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3 
To: <sip:6001@192.168.1.105>;tag=as03c5c0eb 
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0 
CSeq: 62542 ACK 
Content-Length: 0 
Max-Forwards: 70 

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

<--- SIP read from WS:192.168.1.104:59439 ---> 
INVITE sip:6001@192.168.1.105 SIP/2.0 
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK1X2JJVn1ZhXRITLTzwAVwszBoWwztiOn;rport 
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3 
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: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0 
CSeq: 62543 INVITE 
Content-Type: application/sdp 
Content-Length: 1533 
Max-Forwards: 70 
Authorization: Digest username="6002",realm="192.168.1.105",nonce="6ad3c739",uri="sip:6001@192.168.1.105",response="455fa3f20baf557b4a1bf192fdb1d935",algorithm=MD5 
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18 
Organization: Doubango Telecom 

v=0 
o=- 2577394367678132000 2 IN IP4 127.0.0.1 
s=Doubango Telecom - chrome 
t=0 0 
a=group:BUNDLE audio 
a=msid-semantic: WMS HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM 
m=audio 42985 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 
c=IN IP4 192.168.1.104 
a=rtcp:42985 IN IP4 192.168.1.104 
a=candidate:1019731727 1 udp 2122260223 192.168.1.104 42985 typ host generation 0 
a=candidate:1019731727 2 udp 2122260223 192.168.1.104 42985 typ host generation 0 
a=candidate:1917068287 1 tcp 1518280447 192.168.1.104 0 typ host generation 0 
a=candidate:1917068287 2 tcp 1518280447 192.168.1.104 0 typ host generation 0 
a=ice-ufrag:fHkUinzS5ohjJhal 
a=ice-pwd:6Q37zMp1s4jjos+z63Gy3u+3 
a=ice-options:google-ice 
a=fingerprint:sha-256 17:AE:2A:F4:FE:8A:C9:AF:E0:32:1C:93:AA:58:C9:91:79:4A:6B:2A:63:5B:67:F2:81:1A:2E:8A:EC:DC:33:D4 
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:1974193999 cname:bB9ulmmnutOJVAtz 
a=ssrc:1974193999 msid:HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM d16640b3-40b2-4f7d-95cc-5808611d201a 
a=ssrc:1974193999 mslabel:HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM 
a=ssrc:1974193999 label:d16640b3-40b2-4f7d-95cc-5808611d201a 
<-------------> 
--- (13 headers 38 lines) --- 
Using INVITE request as basis request - c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0 
Found peer '6002' for '6002' from 192.168.1.104:59439 
  == 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), peer - audio=(ulaw|alaw)/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 192.168.1.104:42985 
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 (no NAT) to 192.168.1.104:5060 ---> 
SIP/2.0 100 Trying 
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK1X2JJVn1ZhXRITLTzwAVwszBoWwztiOn;rport;received=192.168.1.104 
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3 
To: <sip:6001@192.168.1.105> 
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0 
CSeq: 62543 INVITE 
Server: Asterisk PBX 11.11.0 
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/6002-00000006", "SIP/6001,15") in new stack 
  == Using SIP RTP CoS mark 5 
Audio is at 25382 
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@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP SIP/2.0 
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK7e0e84b5;rport 
Max-Forwards: 70 
From: "6002" <sip:6002@192.168.1.105>;tag=as698a9a90 
To: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP> 
Contact: <sip:6002@192.168.1.105:5060> 
Call-ID: 4c35c0026845ff8313b0c6e14ebe9679@192.168.1.105:5060 
CSeq: 102 INVITE 
User-Agent: Asterisk PBX 11.11.0 
Date: Sat, 02 Aug 2014 21:06:40 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 2039524556 2039524556 IN IP4 192.168.1.105 
s=Asterisk PBX 11.11.0 
c=IN IP4 192.168.1.105 
t=0 0 
m=audio 25382 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=z9hG4bK7e0e84b5;rport=5060 
To: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP> 
From: "6002" <sip:6002@192.168.1.105>;tag=as698a9a90 
Call-ID: 4c35c0026845ff8313b0c6e14ebe9679@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=z9hG4bK7e0e84b5;rport=5060 
Contact: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP> 
To: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP>;tag=b31dcf34 
From: "6002"<sip:6002@192.168.1.105>;tag=as698a9a90 
Call-ID: 4c35c0026845ff8313b0c6e14ebe9679@192.168.1.105:5060 
CSeq: 102 INVITE 
User-Agent: Zoiper r21155 
Content-Length: 0 

<-------------> 
--- (9 headers 0 lines) --- 
list_route: hop: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP> 
    -- SIP/6001-00000007 is ringing 

<--- Transmitting (no NAT) to 192.168.1.104:5060 ---> 
SIP/2.0 180 Ringing 
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK1X2JJVn1ZhXRITLTzwAVwszBoWwztiOn;rport;received=192.168.1.104 
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3 
To: <sip:6001@192.168.1.105>;tag=as6731b4d6 
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0 
CSeq: 62543 INVITE 
Server: Asterisk PBX 11.11.0 
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 


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

<--- SIP read from UDP:192.168.1.105:5061 ---> 


<-------------> 
Really destroying SIP dialog '9f5b3baf-2d7e-6900-494f-c4773252203c' Method: REGISTER 
       > 0x92f18b0 -- Probation passed - setting RTP source address to 192.168.1.105:8000 

<--- SIP read from UDP:192.168.1.105:5061 ---> 
SIP/2.0 200 OK 
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK7e0e84b5;rport=5060 
Contact: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP> 
To: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP>;tag=b31dcf34 
From: "6002"<sip:6002@192.168.1.105>;tag=as698a9a90 
Call-ID: 4c35c0026845ff8313b0c6e14ebe9679@192.168.1.105:5060 
CSeq: 102 INVITE 
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE 
Content-Type: application/sdp 
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri 
User-Agent: Zoiper r21155 
Allow-Events: presence, kpml 
Content-Length: 308 

v=0 
o=Z 0 3 IN IP4 197.31.123.195 
s=Z 
c=IN IP4 197.31.123.195 
t=0 0 
m=audio 8000 RTP/AVP 0 3 110 98 8 101 
a=rtpmap:0 PCMU/8000 
a=rtpmap:3 GSM/8000 
a=rtpmap:110 speex/8000 
a=rtpmap:98 iLBC/8000 
a=fmtp:98 mode=30 
a=rtpmap:8 PCMA/8000 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-15 
a=sendrecv 
<-------------> 
--- (13 headers 15 lines) --- 
Found RTP audio format 0 
Found RTP audio format 3 
Found RTP audio format 110 
Found RTP audio format 98 
Found RTP audio format 8 
Found RTP audio format 101 
Found audio description format PCMU for ID 0 
Found audio description format GSM for ID 3 
Found audio description format speex for ID 110 
Found audio description format iLBC for ID 98 
Found audio description format PCMA for ID 8 
Found audio description format telephone-event for ID 101 
Capabilities: us - (ulaw|alaw), peer - audio=(gsm|ulaw|alaw|speex|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) 
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) 
Peer audio RTP is at port 197.31.123.195:8000 
list_route: hop: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP> 
set_destination: Parsing <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP> for address/port to send to 
set_destination: set destination to 197.31.123.195:5061 
Transmitting (NAT) to 192.168.1.105:5061: 
ACK sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP SIP/2.0 
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK6403d8d9;rport 
Max-Forwards: 70 
From: "6002" <sip:6002@192.168.1.105>;tag=as698a9a90 
To: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP>;tag=b31dcf34 
Contact: <sip:6002@192.168.1.105:5060> 
Call-ID: 4c35c0026845ff8313b0c6e14ebe9679@192.168.1.105:5060 
CSeq: 102 ACK 
User-Agent: Asterisk PBX 11.11.0 
Content-Length: 0 


--- 
    -- SIP/6001-00000007 answered SIP/6002-00000006 
Audio is at 21190 
Adding codec 100003 (ulaw) to SDP 
Adding non-codec 0x1 (telephone-event) to SDP 

<--- Reliably Transmitting (no NAT) to 192.168.1.104:5060 ---> 
SIP/2.0 200 OK 
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK1X2JJVn1ZhXRITLTzwAVwszBoWwztiOn;rport;received=192.168.1.104 
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3 
To: <sip:6001@192.168.1.105>;tag=as6731b4d6 
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0 
CSeq: 62543 INVITE 
Server: Asterisk PBX 11.11.0 
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-Type: application/sdp 
Content-Length: 401 

v=0 
o=root 2120884642 2120884642 IN IP4 192.168.1.105 
s=Asterisk PBX 11.11.0 
c=IN IP4 192.168.1.105 
t=0 0 
m=audio 21190 UDP/TLS/RTP/SAVPF 0 126 
a=rtpmap:0 PCMU/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 02:9A:35:5F:0C:1B:69:0B:47:1B:B6:FB:BC:D7:96:09:D9:BC:BA:62:12:A9:4F:83:D2:CC:2C:CD:58:97:F1:51 
a=sendrecv 

<------------> 
       > 0x92f18b0 -- Probation passed - setting RTP source address to 192.168.1.105:8000 

<--- SIP read from WS:192.168.1.104:59439 ---> 
ACK sip:6001@192.168.1.105:5060;transport=WS SIP/2.0 
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKGQCwpPqNNDIZ481uSBwI;rport 
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3 
To: <sip:6001@192.168.1.105>;tag=as6731b4d6 
Contact: "amal"<sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr" 
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0 
CSeq: 62543 ACK 
Content-Length: 0 
Max-Forwards: 70 
Authorization: Digest username="6002",realm="192.168.1.105",nonce="6ad3c739",uri="sip:6001@192.168.1.105:5060;transport=WS",response="9954906e8debce0780742e3b9237ea1a",algorithm=MD5 
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18 
Organization: Doubango Telecom 

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

<--- SIP read from UDP:192.168.1.105:5061 ---> 
BYE sip:6002@192.168.1.105:5060 SIP/2.0 
Via: SIP/2.0/UDP 197.31.123.195:5061;branch=z9hG4bK-d8754z-43d5b88923bed3f0-1---d8754z- 
Max-Forwards: 70 
Contact: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP> 
To: "6002"<sip:6002@192.168.1.105>;tag=as698a9a90 
From: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP>;tag=b31dcf34 
Call-ID: 4c35c0026845ff8313b0c6e14ebe9679@192.168.1.105:5060 
CSeq: 2 BYE 
User-Agent: Zoiper r21155 
Content-Length: 0 

<-------------> 
--- (10 headers 0 lines) --- 
Sending to 192.168.1.105:5061 (NAT) 
Scheduling destruction of SIP dialog '4c35c0026845ff8313b0c6e14ebe9679@192.168.1.105:5060' in 32000 ms (Method: BYE) 

<--- Transmitting (NAT) to 192.168.1.105:5061 ---> 
SIP/2.0 200 OK 
Via: SIP/2.0/UDP 197.31.123.195:5061;branch=z9hG4bK-d8754z-43d5b88923bed3f0-1---d8754z-;received=192.168.1.105;rport=5061 
From: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP>;tag=b31dcf34 
To: "6002"<sip:6002@192.168.1.105>;tag=as698a9a90 
Call-ID: 4c35c0026845ff8313b0c6e14ebe9679@192.168.1.105:5060 
CSeq: 2 BYE 
Server: Asterisk PBX 11.11.0 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE 
Supported: replaces, timer 
Content-Length: 0 


<------------> 
  == Spawn extension (from-internal, 6001, 1) exited non-zero on 'SIP/6002-00000006' 
Scheduling destruction of SIP dialog 'c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0' in 32000 ms (Method: INVITE) 
set_destination: Parsing <sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;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 192.168.1.104:5060: 
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;branch=z9hG4bK3fda03bc 
Max-Forwards: 70 
From: <sip:6001@192.168.1.105>;tag=as6731b4d6 
To: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3 
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0 
CSeq: 102 BYE 
User-Agent: Asterisk PBX 11.11.0 
Proxy-Authorization: Digest username="6002", realm="192.168.1.105", algorithm=MD5, uri="sip:192.168.1.105", nonce="6ad3c739", response="4188a9e02c874579f65ebf7b91182d76" 
X-Asterisk-HangupCause: Normal Clearing 
X-Asterisk-HangupCauseCode: 16 
Content-Length: 0 


--- 

<--- SIP read from WS:192.168.1.104:59439 ---> 
SIP/2.0 200 OK 
Via: SIP/2.0/WS 192.168.1.105:5060;branch=z9hG4bK3fda03bc 
From: <sip:6001@192.168.1.105>;tag=as6731b4d6 
To: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3 
Contact: <sip:6002@df7jal23ls0d.invalid;transport=ws> 
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0 
CSeq: 102 BYE 
Content-Length: 0 

<-------------> 
--- (8 headers 0 lines) --- 
SIP Response message for INCOMING dialog BYE arrived 
Really destroying SIP dialog 'c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0' Method: INVITE 

<--- SIP read from UDP:192.168.1.105:5061 ---> 


<-------------> 
Really destroying SIP dialog '4c35c0026845ff8313b0c6e14ebe9679@192.168.1.105:5060' Method: BYE 
debian*CLI>

my Sipml5 debug:

State machine: c0000_Started_2_Outgoing_X_oINVITE tsk_utils.js?svn=224:116
ICE servers:[{"url":"stun:null"}] tsk_utils.js?svn=224:116
==stack event = m_permission_requested tsk_utils.js?svn=224:116
==session event = connecting tsk_utils.js?svn=224:116
onGetUserMediaSuccess tsk_utils.js?svn=224:116
createOffer tsk_utils.js?svn=224:116
==stack event = m_permission_accepted tsk_utils.js?svn=224:116
onCreateSdpSuccess tsk_utils.js?svn=224:116
==session event = m_stream_audio_local_added tsk_utils.js?svn=224:116
onSetLocalDescriptionSuccess tsk_utils.js?svn=224:116
5
onIceCandidate = undefined tsk_utils.js?svn=224:116
ICE GATHERING COMPLETED! tsk_utils.js?svn=224:116
onIceGatheringCompleted tsk_utils.js?svn=224:116
SEND: INVITE sip:6001@192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKgHNUD0sQwFgqFQTMZXkhXuh9TaLtyzrt;rport
From: "amal"<sip:6002@192.168.1.105>;tag=1Tmp2lBCMvsyOTm5DTpE
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: 75db4220-9ef3-86c9-8fcf-e7fdf4775892
CSeq: 23605 INVITE
Content-Type: application/sdp
Content-Length: 1533
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

v=0
o=- 2732090651842997000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS U3LAtQLc2XMs2izIxXXJL8xHU7f0FNKQIbx6
m=audio 44683 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.1.104
a=rtcp:44683 IN IP4 192.168.1.104
a=candidate:1019731727 1 udp 2122260223 192.168.1.104 44683 typ host generation 0
a=candidate:1019731727 2 udp 2122260223 192.168.1.104 44683 typ host generation 0
a=candidate:1917068287 1 tcp 1518280447 192.168.1.104 0 typ host generation 0
a=candidate:1917068287 2 tcp 1518280447 192.168.1.104 0 typ host generation 0
a=ice-ufrag:xokyb72LGstFdQBI
a=ice-pwd:NMP9WcQbj49BeA4LXb6sGPkH
a=ice-options:google-ice
a=fingerprint:sha-256 17:AE:2A:F4:FE:8A:C9:AF:E0:32:1C:93:AA:58:C9:91:79:4A:6B:2A:63:5B:67:F2:81:1A:2E:8A:EC:DC:33:D4
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:2043935191 cname:9YLjUZ+QgfTBfRWd
a=ssrc:2043935191 msid:U3LAtQLc2XMs2izIxXXJL8xHU7f0FNKQIbx6 4b6559f6-6a12-429f-8f46-aafb1cb2d505
a=ssrc:2043935191 mslabel:U3LAtQLc2XMs2izIxXXJL8xHU7f0FNKQIbx6
a=ssrc:2043935191 label:4b6559f6-6a12-429f-8f46-aafb1cb2d505
 tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKgHNUD0sQwFgqFQTMZXkhXuh9TaLtyzrt
From: "amal"<sip:6002@192.168.1.105>;tag=1Tmp2lBCMvsyOTm5DTpE
To: <sip:6001@192.168.1.105>;tag=as75ccf8c8
Call-ID: 75db4220-9ef3-86c9-8fcf-e7fdf4775892
CSeq: 23605 INVITE
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="192.168.1.105",nonce="1f8890a3",stale=FALSE,algorithm=MD5

 tsk_utils.js?svn=224:116
SEND: ACK sip:6001@192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKgHNUD0sQwFgqFQTMZXkhXuh9TaLtyzrt;rport
From: "amal"<sip:6002@192.168.1.105>;tag=1Tmp2lBCMvsyOTm5DTpE
To: <sip:6001@192.168.1.105>;tag=as75ccf8c8
Call-ID: 75db4220-9ef3-86c9-8fcf-e7fdf4775892
CSeq: 23605 ACK
Content-Length: 0
Max-Forwards: 70

 tsk_utils.js?svn=224:116
State machine: x0000_Any_2_Any_X_i401_407_INVITE tsk_utils.js?svn=224:116
SEND: INVITE sip:6001@192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKtDazRM2SUb5OvOKh2dnXLioEV7OcNgma;rport
From: "amal"<sip:6002@192.168.1.105>;tag=1Tmp2lBCMvsyOTm5DTpE
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: 75db4220-9ef3-86c9-8fcf-e7fdf4775892
CSeq: 23606 INVITE
Content-Type: application/sdp
Content-Length: 1533
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="1f8890a3",uri="sip:6001@192.168.1.105",response="488af2ac260ef24356958095f0b45cc1",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

v=0
o=- 2732090651842997000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS U3LAtQLc2XMs2izIxXXJL8xHU7f0FNKQIbx6
m=audio 44683 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.1.104
a=rtcp:44683 IN IP4 192.168.1.104
a=candidate:1019731727 1 udp 2122260223 192.168.1.104 44683 typ host generation 0
a=candidate:1019731727 2 udp 2122260223 192.168.1.104 44683 typ host generation 0
a=candidate:1917068287 1 tcp 1518280447 192.168.1.104 0 typ host generation 0
a=candidate:1917068287 2 tcp 1518280447 192.168.1.104 0 typ host generation 0
a=ice-ufrag:xokyb72LGstFdQBI
a=ice-pwd:NMP9WcQbj49BeA4LXb6sGPkH
a=ice-options:google-ice
a=fingerprint:sha-256 17:AE:2A:F4:FE:8A:C9:AF:E0:32:1C:93:AA:58:C9:91:79:4A:6B:2A:63:5B:67:F2:81:1A:2E:8A:EC:DC:33:D4
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:2043935191 cname:9YLjUZ+QgfTBfRWd
a=ssrc:2043935191 msid:U3LAtQLc2XMs2izIxXXJL8xHU7f0FNKQIbx6 4b6559f6-6a12-429f-8f46-aafb1cb2d505
a=ssrc:2043935191 mslabel:U3LAtQLc2XMs2izIxXXJL8xHU7f0FNKQIbx6
a=ssrc:2043935191 label:4b6559f6-6a12-429f-8f46-aafb1cb2d505
 tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKtDazRM2SUb5OvOKh2dnXLioEV7OcNgma
From: "amal"<sip:6002@192.168.1.105>;tag=1Tmp2lBCMvsyOTm5DTpE
To: <sip:6001@192.168.1.105>
Contact: <sip:6001@192.168.1.105:5060;transport=WS>
Call-ID: 75db4220-9ef3-86c9-8fcf-e7fdf4775892
CSeq: 23606 INVITE
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

 tsk_utils.js?svn=224:116
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js?svn=224:116
==session event = i_ao_request tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKtDazRM2SUb5OvOKh2dnXLioEV7OcNgma
From: "amal"<sip:6002@192.168.1.105>;tag=1Tmp2lBCMvsyOTm5DTpE
To: <sip:6001@192.168.1.105>;tag=as7db2a4a3
Contact: <sip:6001@192.168.1.105:5060;transport=WS>
Call-ID: 75db4220-9ef3-86c9-8fcf-e7fdf4775892
CSeq: 23606 INVITE
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

 tsk_utils.js?svn=224:116
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js?svn=224:116
==session event = i_ao_request tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKtDazRM2SUb5OvOKh2dnXLioEV7OcNgma
From: "amal"<sip:6002@192.168.1.105>;tag=1Tmp2lBCMvsyOTm5DTpE
To: <sip:6001@192.168.1.105>;tag=as7db2a4a3
Contact: <sip:6001@192.168.1.105:5060;transport=WS>
Call-ID: 75db4220-9ef3-86c9-8fcf-e7fdf4775892
CSeq: 23606 INVITE
Content-Type: application/sdp
Content-Length: 399
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

v=0
o=root 373378232 373378232 IN IP4 192.168.1.105
s=Asterisk PBX 11.11.0
c=IN IP4 192.168.1.105
t=0 0
m=audio 22550 UDP/TLS/RTP/SAVPF 0 126
a=rtpmap:0 PCMU/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 02:9A:35:5F:0C:1B:69:0B:47:1B:B6:FB:BC:D7:96:09:D9:BC:BA:62:12:A9:4F:83:D2:CC:2C:CD:58:97:F1:51
a=sendrecv
 tsk_utils.js?svn=224:116
State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE tsk_utils.js?svn=224:116
setRemoteDescription(answer)
v=0
o=root 373378232 373378232 IN IP4 192.168.1.105
s=Asterisk PBX 11.11.0
c=IN IP4 192.168.1.105
t=0 0
m=audio 22550 RTP/SAVPF 0 126
a=rtpmap:0 PCMU/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 02:9A:35:5F:0C:1B:69:0B:47:1B:B6:FB:BC:D7:96:09:D9:BC:BA:62:12:A9:4F:83:D2:CC:2C:CD:58:97:F1:51
a=sendrecv
 tsk_utils.js?svn=224:116
SEND: ACK sip:6001@192.168.1.105:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKu31tJTM5Cr0D092qmUbt;rport
From: "amal"<sip:6002@192.168.1.105>;tag=1Tmp2lBCMvsyOTm5DTpE
To: <sip:6001@192.168.1.105>;tag=as7db2a4a3
Contact: "amal"<sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 75db4220-9ef3-86c9-8fcf-e7fdf4775892
CSeq: 23606 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="1f8890a3",uri="sip:6001@192.168.1.105:5060;transport=WS",response="85b438f7d9984c926147554e58bd9fb7",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

 tsk_utils.js?svn=224:116
onSetRemoteDescriptionError tsk_utils.js?svn=224:116
Failed to set remote answer sdp: Called with SDP without ice-ufrag and ice-pwd. tsk_utils.js?svn=224:128
==session event = m_early_media tsk_utils.js?svn=224:116
==session event = connected tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
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;branch=z9hG4bK6501c79a
From: <sip:6001@192.168.1.105>;tag=as7db2a4a3
To: "amal"<sip:6002@192.168.1.105>;tag=1Tmp2lBCMvsyOTm5DTpE
Call-ID: 75db4220-9ef3-86c9-8fcf-e7fdf4775892
CSeq: 102 BYE
Content-Length: 0
Max-Forwards: 70
User-Agent: Asterisk PBX 11.11.0
Proxy-Authorization: Digest username="6002",realm="192.168.1.105",nonce="1f8890a3",uri="sip:192.168.1.105",response="69ccda7dd5a7e67d599e8d557f5c8273",algorithm=MD5
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16

 tsk_utils.js?svn=224:116
State machine: x0000_Any_2_Terminated_X_iBYE tsk_utils.js?svn=224:116
=== INVITE Dialog terminated === tsk_utils.js?svn=224:116
PeerConnection::stop() tsk_utils.js?svn=224:116
SEND: SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.1.105:5060;branch=z9hG4bK6501c79a
From: <sip:6001@192.168.1.105>;tag=as7db2a4a3
To: "amal"<sip:6002@192.168.1.105>;tag=1Tmp2lBCMvsyOTm5DTpE
Contact: <sip:6002@df7jal23ls0d.invalid;transport=ws>
Call-ID: 75db4220-9ef3-86c9-8fcf-e7fdf4775892
CSeq: 102 BYE
Content-Length: 0

 tsk_utils.js?svn=224:116
==session event = terminated tsk_utils.js?svn=224:116
The FSM is in the final state tsk_utils.js?svn=224:122
State machine: tsip_dialog_register_Connected_2_InProgress_X_oRegister tsk_utils.js?svn=224:116
SEND: REGISTER sip:192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKikApHPC1Gtgl2bUU4fqtgrBP5Mm2baay;rport
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
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: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57812 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="3a290ce0",uri="sip:192.168.1.105",response="715632384099d84923a8698184c46be5",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

 tsk_utils.js?svn=224:116
==session event = sent_request tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKikApHPC1Gtgl2bUU4fqtgrBP5Mm2baay
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
To: "amal"<sip:6002@192.168.1.105>;tag=as389399be
Call-ID: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57812 REGISTER
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="192.168.1.105",nonce="745215ab",stale=FALSE,algorithm=MD5

 tsk_utils.js?svn=224:116
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494 tsk_utils.js?svn=224:116
SEND: REGISTER sip:192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKfTc0LKHak33wIaKbcKipSai69Z5RKOau;rport
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
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: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57813 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="745215ab",uri="sip:192.168.1.105",response="bbe9b541971cab616b0293674a9af37f",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

 tsk_utils.js?svn=224:116
==session event = sent_request tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKfTc0LKHak33wIaKbcKipSai69Z5RKOau
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
To: "amal"<sip:6002@192.168.1.105>;tag=as389399be
Contact: <sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200
Call-ID: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57813 REGISTER
Expires: 200
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Date: 02 Aug 2014 21:06:12 GMT;02

 tsk_utils.js?svn=224:116
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx tsk_utils.js?svn=224:116
State machine: c0000_Started_2_Outgoing_X_oINVITE tsk_utils.js?svn=224:116
ICE servers:[{"url":"stun:null"}] tsk_utils.js?svn=224:116
==stack event = m_permission_requested tsk_utils.js?svn=224:116
==session event = connecting tsk_utils.js?svn=224:116
onGetUserMediaSuccess tsk_utils.js?svn=224:116
createOffer tsk_utils.js?svn=224:116
onCreateSdpSuccess tsk_utils.js?svn=224:116
==stack event = m_permission_accepted tsk_utils.js?svn=224:116
==session event = m_stream_audio_local_added tsk_utils.js?svn=224:116
onSetLocalDescriptionSuccess tsk_utils.js?svn=224:116
5
onIceCandidate = undefined tsk_utils.js?svn=224:116
ICE GATHERING COMPLETED! tsk_utils.js?svn=224:116
onIceGatheringCompleted tsk_utils.js?svn=224:116
SEND: INVITE sip:6001@192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKDvVLWqTerDmfvU7ntzHJQt1sO5nFm5S2;rport
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
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: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62542 INVITE
Content-Type: application/sdp
Content-Length: 1533
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

v=0
o=- 2577394367678132000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM
m=audio 42985 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.1.104
a=rtcp:42985 IN IP4 192.168.1.104
a=candidate:1019731727 1 udp 2122260223 192.168.1.104 42985 typ host generation 0
a=candidate:1019731727 2 udp 2122260223 192.168.1.104 42985 typ host generation 0
a=candidate:1917068287 1 tcp 1518280447 192.168.1.104 0 typ host generation 0
a=candidate:1917068287 2 tcp 1518280447 192.168.1.104 0 typ host generation 0
a=ice-ufrag:fHkUinzS5ohjJhal
a=ice-pwd:6Q37zMp1s4jjos+z63Gy3u+3
a=ice-options:google-ice
a=fingerprint:sha-256 17:AE:2A:F4:FE:8A:C9:AF:E0:32:1C:93:AA:58:C9:91:79:4A:6B:2A:63:5B:67:F2:81:1A:2E:8A:EC:DC:33:D4
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:1974193999 cname:bB9ulmmnutOJVAtz
a=ssrc:1974193999 msid:HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM d16640b3-40b2-4f7d-95cc-5808611d201a
a=ssrc:1974193999 mslabel:HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM
a=ssrc:1974193999 label:d16640b3-40b2-4f7d-95cc-5808611d201a
 tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKDvVLWqTerDmfvU7ntzHJQt1sO5nFm5S2
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>;tag=as03c5c0eb
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62542 INVITE
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="192.168.1.105",nonce="6ad3c739",stale=FALSE,algorithm=MD5

 tsk_utils.js?svn=224:116
SEND: ACK sip:6001@192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKDvVLWqTerDmfvU7ntzHJQt1sO5nFm5S2;rport
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>;tag=as03c5c0eb
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62542 ACK
Content-Length: 0
Max-Forwards: 70

 tsk_utils.js?svn=224:116
State machine: x0000_Any_2_Any_X_i401_407_INVITE tsk_utils.js?svn=224:116
SEND: INVITE sip:6001@192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK1X2JJVn1ZhXRITLTzwAVwszBoWwztiOn;rport
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
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: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62543 INVITE
Content-Type: application/sdp
Content-Length: 1533
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="6ad3c739",uri="sip:6001@192.168.1.105",response="455fa3f20baf557b4a1bf192fdb1d935",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

v=0
o=- 2577394367678132000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM
m=audio 42985 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.1.104
a=rtcp:42985 IN IP4 192.168.1.104
a=candidate:1019731727 1 udp 2122260223 192.168.1.104 42985 typ host generation 0
a=candidate:1019731727 2 udp 2122260223 192.168.1.104 42985 typ host generation 0
a=candidate:1917068287 1 tcp 1518280447 192.168.1.104 0 typ host generation 0
a=candidate:1917068287 2 tcp 1518280447 192.168.1.104 0 typ host generation 0
a=ice-ufrag:fHkUinzS5ohjJhal
a=ice-pwd:6Q37zMp1s4jjos+z63Gy3u+3
a=ice-options:google-ice
a=fingerprint:sha-256 17:AE:2A:F4:FE:8A:C9:AF:E0:32:1C:93:AA:58:C9:91:79:4A:6B:2A:63:5B:67:F2:81:1A:2E:8A:EC:DC:33:D4
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:1974193999 cname:bB9ulmmnutOJVAtz
a=ssrc:1974193999 msid:HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM d16640b3-40b2-4f7d-95cc-5808611d201a
a=ssrc:1974193999 mslabel:HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM
a=ssrc:1974193999 label:d16640b3-40b2-4f7d-95cc-5808611d201a
 tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bK1X2JJVn1ZhXRITLTzwAVwszBoWwztiOn
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>
Contact: <sip:6001@192.168.1.105:5060;transport=WS>
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62543 INVITE
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

 tsk_utils.js?svn=224:116
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js?svn=224:116
==session event = i_ao_request tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bK1X2JJVn1ZhXRITLTzwAVwszBoWwztiOn
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>;tag=as6731b4d6
Contact: <sip:6001@192.168.1.105:5060;transport=WS>
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62543 INVITE
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

 tsk_utils.js?svn=224:116
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js?svn=224:116
==session event = i_ao_request tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bK1X2JJVn1ZhXRITLTzwAVwszBoWwztiOn
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>;tag=as6731b4d6
Contact: <sip:6001@192.168.1.105:5060;transport=WS>
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62543 INVITE
Content-Type: application/sdp
Content-Length: 401
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

v=0
o=root 2120884642 2120884642 IN IP4 192.168.1.105
s=Asterisk PBX 11.11.0
c=IN IP4 192.168.1.105
t=0 0
m=audio 21190 UDP/TLS/RTP/SAVPF 0 126
a=rtpmap:0 PCMU/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 02:9A:35:5F:0C:1B:69:0B:47:1B:B6:FB:BC:D7:96:09:D9:BC:BA:62:12:A9:4F:83:D2:CC:2C:CD:58:97:F1:51
a=sendrecv
 tsk_utils.js?svn=224:116
State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE tsk_utils.js?svn=224:116
setRemoteDescription(answer)
v=0
o=root 2120884642 2120884642 IN IP4 192.168.1.105
s=Asterisk PBX 11.11.0
c=IN IP4 192.168.1.105
t=0 0
m=audio 21190 RTP/SAVPF 0 126
a=rtpmap:0 PCMU/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 02:9A:35:5F:0C:1B:69:0B:47:1B:B6:FB:BC:D7:96:09:D9:BC:BA:62:12:A9:4F:83:D2:CC:2C:CD:58:97:F1:51
a=sendrecv
 tsk_utils.js?svn=224:116
SEND: ACK sip:6001@192.168.1.105:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKGQCwpPqNNDIZ481uSBwI;rport
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>;tag=as6731b4d6
Contact: "amal"<sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62543 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="6ad3c739",uri="sip:6001@192.168.1.105:5060;transport=WS",response="9954906e8debce0780742e3b9237ea1a",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

 tsk_utils.js?svn=224:116
onSetRemoteDescriptionError tsk_utils.js?svn=224:116
Failed to set remote answer sdp: Called with SDP without ice-ufrag and ice-pwd. tsk_utils.js?svn=224:128
==session event = m_early_media tsk_utils.js?svn=224:116
==session event = connected tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
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;branch=z9hG4bK3fda03bc
From: <sip:6001@192.168.1.105>;tag=as6731b4d6
To: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 102 BYE
Content-Length: 0
Max-Forwards: 70
User-Agent: Asterisk PBX 11.11.0
Proxy-Authorization: Digest username="6002",realm="192.168.1.105",nonce="6ad3c739",uri="sip:192.168.1.105",response="4188a9e02c874579f65ebf7b91182d76",algorithm=MD5
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16

 tsk_utils.js?svn=224:116
State machine: x0000_Any_2_Terminated_X_iBYE tsk_utils.js?svn=224:116
=== INVITE Dialog terminated === tsk_utils.js?svn=224:116
PeerConnection::stop() tsk_utils.js?svn=224:116
SEND: SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.1.105:5060;branch=z9hG4bK3fda03bc
From: <sip:6001@192.168.1.105>;tag=as6731b4d6
To: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
Contact: <sip:6002@df7jal23ls0d.invalid;transport=ws>
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 102 BYE
Content-Length: 0

 tsk_utils.js?svn=224:116
==session event = terminated tsk_utils.js?svn=224:116
The FSM is in the final state tsk_utils.js?svn=224:122
State machine: tsip_dialog_register_Connected_2_InProgress_X_oRegister tsk_utils.js?svn=224:116
SEND: REGISTER sip:192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKl1MKbTRKWcaQFxEWKxR6KJVXuOvhg1sO;rport
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
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: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57814 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="745215ab",uri="sip:192.168.1.105",response="bbe9b541971cab616b0293674a9af37f",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

 tsk_utils.js?svn=224:116
==session event = sent_request tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKl1MKbTRKWcaQFxEWKxR6KJVXuOvhg1sO
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
To: "amal"<sip:6002@192.168.1.105>;tag=as68f45007
Call-ID: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57814 REGISTER
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="192.168.1.105",nonce="321f2a10",stale=FALSE,algorithm=MD5

 tsk_utils.js?svn=224:116
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494 tsk_utils.js?svn=224:116
SEND: REGISTER sip:192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKQmvhBdCeANbzmq5TiKBAGgWTtegsU3RW;rport
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
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: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57815 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="321f2a10",uri="sip:192.168.1.105",response="f2bbde081390003688b3211e6b39a325",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

 tsk_utils.js?svn=224:116
==session event = sent_request tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKQmvhBdCeANbzmq5TiKBAGgWTtegsU3RW
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
To: "amal"<sip:6002@192.168.1.105>;tag=as68f45007
Contact: <sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200
Call-ID: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57815 REGISTER
Expires: 200
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Date: 02 Aug 2014 21:07:52 GMT;02

 tsk_utils.js?svn=224:116
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx tsk_utils.js?svn=224:116
State machine: tsip_dialog_register_Connected_2_InProgress_X_oRegister tsk_utils.js?svn=224:116
SEND: REGISTER sip:192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKb2xiaSxCloGbsthdprWT6ooMhSlRLp8k;rport
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
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: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57816 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="321f2a10",uri="sip:192.168.1.105",response="f2bbde081390003688b3211e6b39a325",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

 tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKb2xiaSxCloGbsthdprWT6ooMhSlRLp8k
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
To: "amal"<sip:6002@192.168.1.105>;tag=as7d54f249
Call-ID: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57816 REGISTER
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="192.168.1.105",nonce="3be23638",stale=FALSE,algorithm=MD5

 tsk_utils.js?svn=224:116
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494 tsk_utils.js?svn=224:116
SEND: REGISTER sip:192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKwXwTSJ8OyziIgtBcRzJ2Zxy3BHydOHVv;rport
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
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: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57817 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="3be23638",uri="sip:192.168.1.105",response="1905b36637ff0044f263772837a8cbc4",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

 tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKwXwTSJ8OyziIgtBcRzJ2Zxy3BHydOHVv
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
To: "amal"<sip:6002@192.168.1.105>;tag=as7d54f249
Contact: <sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200
Call-ID: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57817 REGISTER
Expires: 200
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Date: 02 Aug 2014 21
 tsk_utils.js?svn=224:116

I have Asterisk 11.11.0 installed in debian 32 bit, and I have executed apt-get install uuid-dev then ./configure for Asterisk and make && make install

navaismo · August 2, 2014, 10:27pm

[quote]ISSUE: I get this response on JSSIP or SIPML5 debug:tRemoteDescription failed: Called with an SDP without ice-ufrag and ice-pwd.
This issue is caused because you asterisk don’t have ICE support, you can solve that by installing the uuid/libuuid and uuid-devel/libuuid-devel packages on your system. Then recompile asterisk(be sure to rerun the configure script before the make command).[/quote]

samira · August 2, 2014, 10:46pm

But I have it installed! : apt-get install uuid-dev, then /.configure and make && make install for asterisk!
Is that wrong for getting ICE support? thank you for the response

brunowebbh · August 13, 2014, 6:44pm

Same problem here. I already have libs installeds and asterisk recompiled but i continue to get this message. Any Suggestion?

anilshah · April 3, 2015, 5:11pm

I am facing the same problem. did anyone solve this problem?

nkt · May 18, 2015, 1:45pm

Hi, same issue here with Sipjs and Asterisk 11.17.1.

Relevant system info:

# dpkg -l '*uuid*' | column
Desired=Unknown/Install/Remove/Purge/Hold
| Status=Not/Inst/Conf-files/Unpacked/halF-conf/Half-inst/trig-aWait/Trig-pend
|/ Err?=(none)/Reinst-required (Status,Err: uppercase=bad)
||/ Name                 Version      Architecture Description
+++-====================-============-============-============================================================
un  libossp-uuid11       <none>       <none>       (no description available)
ii  libossp-uuid16:amd64 1.6.2-1.5+b1 amd64        OSSP uuid ISO-C and C++ - shared library
ii  libuuid1:amd64       2.25.2-6     amd64        Universally Unique ID library
ii  uuid                 1.6.2-1.5+b1 amd64        Universally Unique Identifier Command-Line Tool
ii  uuid-dev:amd64       2.25.2-6     amd64        universally unique id library - headers and static libraries
un  uuid-runtime         <none>       <none>       (no description available)

# dpkg -l '*srtp*' | column                                                                                                                                                           
Desired=Unknown/Install/Remove/Purge/Hold
| Status=Not/Inst/Conf-files/Unpacked/halF-conf/Half-inst/trig-aWait/Trig-pend
|/ Err?=(none)/Reinst-required (Status,Err: uppercase=bad)
||/ Name           Version                 Architecture Description
+++-==============-=======================-============-=======================================================================
un  libsrtp-dev    <none>                  <none>       (no description available)
ii  libsrtp0       1.4.5~20130609~dfsg-1.1 amd64        Secure RTP (SRTP) and UST Reference Implementations - shared library
ii  libsrtp0-dev   1.4.5~20130609~dfsg-1.1 amd64        Secure RTP (SRTP) and UST Reference Implementations - development files
un  libsrtp1-dev   <none>                  <none>       (no description available)
un  srtp-utils     <none>                  <none>       (no description available)

# ldd /usr/lib/asterisk/modules/res_rtp_asterisk.so
        linux-vdso.so.1 (0x00007ffc49d7e000)
        libuuid.so.1 => /lib/x86_64-linux-gnu/libuuid.so.1 (0x00007fe650071000)
        libpthread.so.0 => /lib/x86_64-linux-gnu/libpthread.so.0 (0x00007fe64fe54000)
        libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x00007fe64faaa000)
        /lib64/ld-linux-x86-64.so.2 (0x00007fe650497000)

# ldconfig -v 2>/dev/null | egrep 'rtp|uuid'                                                                                                                                          
        libuuid.so.1 -> libuuid.so.1.3.0
        libossp-uuid++.so.16 -> libossp-uuid++.so.16.0.22
        libossp-uuid.so.16 -> libossp-uuid.so.16.0.22
        libsrtp.so.0 -> libsrtp.so.0.0

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Failed to set remote answer sdp: Called with SDP without ice

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