No Voice When Using WebRTC

paatalom · December 22, 2014, 2:19pm

Hello asterisk team,

I’m newbie on asterisk and this is my first post.

My Current Environment :

OS :
CentOS release 6.6 (Final)
Linux asterisk 2.6.32-504.1.3.el6.x86_64 #1 SMP Tue Nov 11 17:57:25 UTC 2014 x86_64 x86_64 x86_64 GNU/Linux
Asterisk :
Asterisk 11.14.2, Copyright © 1999 - 2013 Digium, Inc. and others.
Client Browser :
Firefox 34.0.5

I want to make a call from my simple javascript client application (which is run on host - for example 192.168.1.3) to secod machine where is run SIP Client “3CXPhone” - host for example : 192.168.1.9.

everything works fine , both client were registered successfully. call aslo established fine. but I have no voice on the both side.

Here is my asteris log files (sip set debug on) :

<--- SIP read from WS:192.168.1.3:34029 --->
INVITE sip:01@192.168.1.28 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKLsO5XprypwV1Q6gakR1IBZpv6k5eqE5x;rport
From: "1060"<sip:1060@192.168.1.28>;tag=uC3esqGT9E84McJENMU4
To: <sip:01@192.168.1.28>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=1060;ha1=1a2e73b7fac658ec7cf668edf6ac0790;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: f3903f11-37e7-f5e8-a4a9-f1d7cb2c70c0
CSeq: 52579 INVITE
Content-Type: application/sdp
Content-Length: 588
Route: <sip:192.168.1.28:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Information Center 08

v=0
o=Mozilla-SIPUA-34.0.5 7398 1 IN IP4 0.0.0.0
s=Doubango Telecom - firefox
t=0 0
a=ice-ufrag:fd694364
a=ice-pwd:97043ddb18ae70bb629675e11f22c263
a=fingerprint:sha-256 44:55:9D:6E:4B:F4:6E:9A:EA:8A:D3:2C:9A:55:A0:B3:45:3A:AF:4A:6F:AA:2D:77:0E:4E:37:A2:70:7B:06:9C
m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 0.0.0.0
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=setup:actpass
a=rtcp-mux
<------------->
--- (13 headers 20 lines) ---
Using INVITE request as basis request - f3903f11-37e7-f5e8-a4a9-f1d7cb2c70c0
Found peer '1060' for '1060' from 192.168.1.3:34029

<--- Reliably Transmitting (no NAT) to 192.168.1.3:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKLsO5XprypwV1Q6gakR1IBZpv6k5eqE5x;rport;received=192.168.1.3
From: "1060"<sip:1060@192.168.1.28>;tag=uC3esqGT9E84McJENMU4
To: <sip:01@192.168.1.28>;tag=as5190de01
Call-ID: f3903f11-37e7-f5e8-a4a9-f1d7cb2c70c0
CSeq: 52579 INVITE
Server: Asterisk PBX 11.14.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1c38f320"
Content-Length: 05


<------------>
Scheduling destruction of SIP dialog 'f3903f11-37e7-f5e8-a4a9-f1d7cb2c70c0' in 32000 ms (Method: INVITE)

<--- SIP read from WS:192.168.1.3:34029 --->
ACK sip:01@192.168.1.28 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKLsO5XprypwV1Q6gakR1IBZpv6k5eqE5x;rport
From: "1060"<sip:1060@192.168.1.28>;tag=uC3esqGT9E84McJENMU4
To: <sip:01@192.168.1.28>;tag=as5190de01
Call-ID: f3903f11-37e7-f5e8-a4a9-f1d7cb2c70c0
CSeq: 52579 ACK
Content-Length: 0
Route: <sip:192.168.1.28:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70

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

<--- SIP read from WS:192.168.1.3:34029 --->
INVITE sip:01@192.168.1.28 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKhkgg4phxQKau7CbdGTgyNntvWrRid6ac;rport
From: "1060"<sip:1060@192.168.1.28>;tag=uC3esqGT9E84McJENMU4
To: <sip:01@192.168.1.28>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=1060;ha1=1a2e73b7fac658ec7cf668edf6ac0790;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: f3903f11-37e7-f5e8-a4a9-f1d7cb2c70c0
CSeq: 52580 INVITE
Content-Type: application/sdp
Content-Length: 588
Route: <sip:192.168.1.28:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="asterisk",nonce="1c38f320",uri="sip:01@192.168.1.28",response="46b3bdf52fc76b9fd66a3288f27aeb60",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Information Center 08

v=0
o=Mozilla-SIPUA-34.0.5 7398 1 IN IP4 0.0.0.0
s=Doubango Telecom - firefox
t=0 0
a=ice-ufrag:fd694364
a=ice-pwd:97043ddb18ae70bb629675e11f22c263
a=fingerprint:sha-256 44:55:9D:6E:4B:F4:6E:9A:EA:8A:D3:2C:9A:55:A0:B3:45:3A:AF:4A:6F:AA:2D:77:0E:4E:37:A2:70:7B:06:9C
m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 0.0.0.0
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=setup:actpass
a=rtcp-mux
<------------->
--- (14 headers 20 lines) ---
Using INVITE request as basis request - f3903f11-37e7-f5e8-a4a9-f1d7cb2c70c0
Found peer '1060' for '1060' from 192.168.1.3:34029
  == Using SIP RTP CoS mark 5
Found RTP audio format 109
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format opus for ID 109
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|h263p|h264|mpeg4|opus), peer - audio=(ulaw|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|opus)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 0.0.0.0:9
Looking for 01 in 08_pbx_out (domain 192.168.1.28)
list_route: hop: <sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>

<--- Transmitting (no NAT) to 192.168.1.3:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKhkgg4phxQKau7CbdGTgyNntvWrRid6ac;rport;received=192.168.1.3
From: "1060"<sip:1060@192.168.1.28>;tag=uC3esqGT9E84McJENMU4
To: <sip:01@192.168.1.28>
Call-ID: f3903f11-37e7-f5e8-a4a9-f1d7cb2c70c0
CSeq: 52580 INVITE
Server: Asterisk PBX 11.14.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:01@192.168.1.28:5060;transport=WS>
Content-Length: 0


<------------>
[Dec 22 17:51:51] WARNING[20305]: res_odbc.c:1412 _ast_odbc_request_obj2: SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC 5.1 Driver]MySQL server has gone away
       > [INSERT INTO cel (eventtype,eventtime,userdeftype,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,peeraccount,uniqueid,linkedid,userfield,peer) VALUES ('CHAN_START',{ts '2014-12-22 17:51:51'},'','1060','1060','','','','01','08_pbx_out','SIP/1060-00000006','','',3,'','','1419256311.6','1419256311.6','','')]
    -- Executing [01@08_pbx_out:1] Goto("SIP/1060-00000006", "30") in new stack
    -- Goto (08_pbx_out,01,30)
    -- Executing [01@08_pbx_out:30] Set("SIP/1060-00000006", "CHANNEL(musicclass)=blue") in new stack
[Dec 22 17:51:51] WARNING[20305]: res_odbc.c:646 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 08S01: [MySQL][ODBC 5.1 Driver][mysqld-5.1.73]MySQL server has gone away (65)
[Dec 22 17:51:51] WARNING[20305]: res_odbc.c:658 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to asterisk [asterisk-connector-mysql]...
[Dec 22 17:51:51] WARNING[20305]: res_odbc.c:762 ast_odbc_sanity_check: Connection is down attempting to reconnect...
[Dec 22 17:51:51] NOTICE[20305]: res_odbc.c:1537 odbc_obj_connect: Connecting asterisk
    -- Executing [01@08_pbx_out:31] Monitor("SIP/1060-00000006", "wav,1060-01-20141222-175151-1419256311.6,mb") in new stack
    -- Executing [01@08_pbx_out:32] Set("SIP/1060-00000006", "CDR(userfield)=14192563116") in new stack
    -- Executing [01@08_pbx_out:33] Goto("SIP/1060-00000006", "2") in new stack
    -- Goto (08_pbx_out,01,2)
    -- Executing [01@08_pbx_out:2] GotoIf("SIP/1060-00000006", "0?5") in new stack
    -- Executing [01@08_pbx_out:3] Macro("SIP/1060-00000006", "sip_call_to_oper_local,phone_01,1,60,stat/data_in,,") in new stack
    -- Executing [s@macro-sip_call_to_oper_local:1] GotoIf("SIP/1060-00000006", "0?8") in new stack
    -- Executing [s@macro-sip_call_to_oper_local:2] Set("SIP/1060-00000006", "GROUP()=phone_01_in") in new stack
    -- Executing [s@macro-sip_call_to_oper_local:3] GotoIf("SIP/1060-00000006", "0?8") in new stack
    -- Executing [s@macro-sip_call_to_oper_local:4] Set("SIP/1060-00000006", "GROUP()=phone_01_out") in new stack
    -- Executing [s@macro-sip_call_to_oper_local:5] Set("SIP/1060-00000006", "OUTBOUND_GROUP=phone_01_in") in new stack
    -- Executing [s@macro-sip_call_to_oper_local:6] NoOp("SIP/1060-00000006", "") in new stack
    -- Executing [s@macro-sip_call_to_oper_local:7] Dial("SIP/1060-00000006", "sip/phone_01,60,Tt") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 16694
Adding codec 100004 (alaw) to SDP
Reliably Transmitting (no NAT) to 192.168.1.9:49842:
INVITE sip:phone_01@192.168.1.9:49842;rinstance=a05fdabd61cb77b6 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.28:5060;branch=z9hG4bK1899c55d
Max-Forwards: 70
From: "1060" <sip:1060@192.168.1.28>;tag=as6b37ce73
To: <sip:phone_01@192.168.1.9:49842;rinstance=a05fdabd61cb77b6>
Contact: <sip:1060@192.168.1.28:5060>
Call-ID: 2e753df60be0f8c5018bf9004213e64b@192.168.1.28:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.14.2
Date: Mon, 22 Dec 2014 13:51:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 178

v=0
o=root 921748531 921748531 IN IP4 192.168.1.28
s=Asterisk PBX 11.14.2
c=IN IP4 192.168.1.28
t=0 0
m=audio 16694 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
    -- Called sip/phone_01
[Dec 22 17:51:51] NOTICE[20305]: res_odbc.c:1569 odbc_obj_connect: res_odbc: Connected to asterisk [asterisk-connector-mysql]
       > [INSERT INTO cel (eventtype,eventtime,userdeftype,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,peeraccount,uniqueid,linkedid,userfield,peer) VALUES ('CHAN_START',{ts '2014-12-22 17:51:51'},'','01','P01','','','','s','08_pbx_out','SIP/phone_01-00000007','','',3,'','','1419256311.7','1419256311.6','','')]

<--- SIP read from UDP:192.168.1.9:49842 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.28:5060;branch=z9hG4bK1899c55d
Contact: <sip:phone_01@192.168.1.9:49842;rinstance=a05fdabd61cb77b6>
To: <sip:phone_01@192.168.1.9:49842;rinstance=a05fdabd61cb77b6>;tag=e943e55b
From: "1060"<sip:1060@192.168.1.28>;tag=as6b37ce73
Call-ID: 2e753df60be0f8c5018bf9004213e64b@192.168.1.28:5060
CSeq: 102 INVITE
User-Agent: 3CXPhone 6.0.18815.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:phone_01@192.168.1.9:49842;rinstance=a05fdabd61cb77b6>
    -- SIP/phone_01-00000007 is ringing

<--- Transmitting (no NAT) to 192.168.1.3:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKhkgg4phxQKau7CbdGTgyNntvWrRid6ac;rport;received=192.168.1.3
From: "1060"<sip:1060@192.168.1.28>;tag=uC3esqGT9E84McJENMU4
To: <sip:01@192.168.1.28>;tag=as69c9f93a
Call-ID: f3903f11-37e7-f5e8-a4a9-f1d7cb2c70c0
CSeq: 52580 INVITE
Server: Asterisk PBX 11.14.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:01@192.168.1.28:5060;transport=WS>
Content-Length: 0


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

<--- SIP read from UDP:192.168.1.9:49842 --->


<------------->
Really destroying SIP dialog 'ZmI3ODFkYTU4NzJiYmE3ZGI1MDI3MTQzNTU1ODgzNjg.' Method: REGISTER

<--- SIP read from UDP:192.168.1.9:49842 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.28:5060;branch=z9hG4bK1899c55d
Contact: <sip:phone_01@192.168.1.9:49842;rinstance=a05fdabd61cb77b6>
To: <sip:phone_01@192.168.1.9:49842;rinstance=a05fdabd61cb77b6>;tag=e943e55b
From: "1060"<sip:1060@192.168.1.28>;tag=as6b37ce73
Call-ID: 2e753df60be0f8c5018bf9004213e64b@192.168.1.28:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.18815.0
Content-Length: 150

v=0
o=3cxVCE 130881690 72452805 IN IP4 192.168.1.9
s=3cxVCE Audio Call
c=IN IP4 192.168.1.9
t=0 0
m=audio 40006 RTP/AVP 8
a=rtpmap:8 PCMA/8000
<------------->
--- (12 headers 7 lines) ---
Found RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.9:40006
list_route: hop: <sip:phone_01@192.168.1.9:49842;rinstance=a05fdabd61cb77b6>
set_destination: Parsing <sip:phone_01@192.168.1.9:49842;rinstance=a05fdabd61cb77b6> for address/port to send to
set_destination: set destination to 192.168.1.9:49842
Transmitting (no NAT) to 192.168.1.9:49842:
ACK sip:phone_01@192.168.1.9:49842;rinstance=a05fdabd61cb77b6 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.28:5060;branch=z9hG4bK103ff4c7
Max-Forwards: 70
From: "1060" <sip:1060@192.168.1.28>;tag=as6b37ce73
To: <sip:phone_01@192.168.1.9:49842;rinstance=a05fdabd61cb77b6>;tag=e943e55b
Contact: <sip:1060@192.168.1.28:5060>
Call-ID: 2e753df60be0f8c5018bf9004213e64b@192.168.1.28:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.14.2
Content-Length: 0


---
    -- SIP/phone_01-00000007 answered SIP/1060-00000006
       > [INSERT INTO cel (eventtype,eventtime,userdeftype,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,peeraccount,uniqueid,linkedid,userfield,peer) VALUES ('ANSWER',{ts '2014-12-22 17:51:56'},'','01','P01','P01','','','01','08_pbx_out','SIP/phone_01-00000007','AppDial','(Outgoing Line)',3,'','','1419256311.7','1419256311.6','','')]
Audio is at 16776
Adding codec 100030 (opus) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.1.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKhkgg4phxQKau7CbdGTgyNntvWrRid6ac;rport;received=192.168.1.3
From: "1060"<sip:1060@192.168.1.28>;tag=uC3esqGT9E84McJENMU4
To: <sip:01@192.168.1.28>;tag=as69c9f93a
Call-ID: f3903f11-37e7-f5e8-a4a9-f1d7cb2c70c0
CSeq: 52580 INVITE
Server: Asterisk PBX 11.14.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:01@192.168.1.28:5060;transport=WS>
Content-Type: application/sdp
Content-Length: 537

v=0
o=root 1107901304 1107901304 IN IP4 192.168.1.28
s=Asterisk PBX 11.14.2
c=IN IP4 192.168.1.28
t=0 0
m=audio 16776 RTP/SAVPF 109 0 8 101
a=rtpmap:109 opus/48000/2
a=maxptime:60
a=fmtp:109 maxplaybackrate=16000; stereo=0; sprop-stereo=0; useinbandfec=0
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=connection:new
a=setup:active
a=fingerprint:SHA-256 B2:72:2B:AF:A0:AC:6C:24:8D:48:04:C5:52:7C:DC:61:C9:CB:FF:EB:B4:38:93:B8:D1:42:AE:27:E7:E1:BC:E2
a=sendrecv

<------------>
       > 0x7fc588008060 -- Probation passed - setting RTP source address to 192.168.1.9:40006

<--- SIP read from WS:192.168.1.3:34029 --->
ACK sip:01@192.168.1.28:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKe1nMmm03SoZL4ZiTnhJh;rport
From: "1060"<sip:1060@192.168.1.28>;tag=uC3esqGT9E84McJENMU4
To: <sip:01@192.168.1.28>;tag=as69c9f93a
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: f3903f11-37e7-f5e8-a4a9-f1d7cb2c70c0
CSeq: 52580 ACK
Content-Length: 0
Route: <sip:192.168.1.28:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="asterisk",nonce="1c38f320",uri="sip:01@192.168.1.28:5060;transport=WS",response="b236f7f948f04b90afbdc7e306769826",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Information Center 08

<------------->
--- (13 headers 0 lines) ---
       > [INSERT INTO cel (eventtype,eventtime,userdeftype,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,peeraccount,uniqueid,linkedid,userfield,peer) VALUES ('ANSWER',{ts '2014-12-22 17:51:56'},'','1060','1060','1060','','01','s','macro-sip_call_to_oper_local','SIP/1060-00000006','Dial','sip/phone_01,60,Tt',3,'','','1419256311.6','1419256311.6','','')]
       > [INSERT INTO cel (eventtype,eventtime,userdeftype,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,peeraccount,uniqueid,linkedid,userfield,peer) VALUES ('BRIDGE_START',{ts '2014-12-22 17:51:56'},'','1060','1060','1060','','01','s','macro-sip_call_to_oper_local','SIP/1060-00000006','Dial','sip/phone_01,60,Tt',3,'','','1419256311.6','1419256311.6','','SIP/phone_01-00000007')]

<--- SIP read from WS:192.168.1.3:34029 --->
REGISTER sip:192.168.1.28 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvLUiyi2lZkb7TSHTVG8U1vlTM3TTkn4m;rport
From: "1060"<sip:1060@192.168.1.28>;tag=yw8fBDsrmt3IS0F15OoD
To: "1060"<sip:1060@192.168.1.28>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 57f8fe29-98fd-6993-7d4e-5bd62c0208dc
CSeq: 5490 REGISTER
Content-Length: 0
Route: <sip:192.168.1.28:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="asterisk",nonce="59bc2834",uri="sip:192.168.1.28",response="972eb54b77fda5fc7e14e82e4534b8de",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Information Center 08

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

<--- Transmitting (no NAT) to 192.168.1.3:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvLUiyi2lZkb7TSHTVG8U1vlTM3TTkn4m;rport;received=192.168.1.3
From: "1060"<sip:1060@192.168.1.28>;tag=yw8fBDsrmt3IS0F15OoD
To: "1060"<sip:1060@192.168.1.28>;tag=as32604e9b
Call-ID: 57f8fe29-98fd-6993-7d4e-5bd62c0208dc
CSeq: 5490 REGISTER
Server: Asterisk PBX 11.14.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5adc39a0"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '57f8fe29-98fd-6993-7d4e-5bd62c0208dc' in 32000 ms (Method: REGISTER)

<--- SIP read from WS:192.168.1.3:34029 --->
REGISTER sip:192.168.1.28 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKaILU91M4DHTIxwG2IWhA2O2qHNOozdwu;rport
From: "1060"<sip:1060@192.168.1.28>;tag=yw8fBDsrmt3IS0F15OoD
To: "1060"<sip:1060@192.168.1.28>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 57f8fe29-98fd-6993-7d4e-5bd62c0208dc
CSeq: 5491 REGISTER
Content-Length: 0
Route: <sip:192.168.1.28:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="asterisk",nonce="5adc39a0",uri="sip:192.168.1.28",response="d9770acf66a94e9db6b3645675223297",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Information Center 08

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

<--- Transmitting (no NAT) to 192.168.1.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKaILU91M4DHTIxwG2IWhA2O2qHNOozdwu;rport;received=192.168.1.3
From: "1060"<sip:1060@192.168.1.28>;tag=yw8fBDsrmt3IS0F15OoD
To: "1060"<sip:1060@192.168.1.28>;tag=as32604e9b
Call-ID: 57f8fe29-98fd-6993-7d4e-5bd62c0208dc
CSeq: 5491 REGISTER
Server: Asterisk PBX 11.14.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 200
Contact: <sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200
Date: Mon, 22 Dec 2014 13:52:00 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '57f8fe29-98fd-6993-7d4e-5bd62c0208dc' in 32000 ms (Method: REGISTER)

Also here is my client application call log (From firebug) :

Calling ..............
mysip.js (line 371)
Calling ..............1
mysip.js (line 388)
Calling ..............2
mysip.js (line 408)
State machine: c0000_Started_2_Outgoing_X_oINVITE
SIPml-api.js (line 2)

PeerConnectionClass = function mozRTCPeerConnection() {
    [native code]
} SessionDescriptionClass = function mozRTCSessionDescription() {
    [native code]
} IceCandidateClass = function mozRTCIceCandidate() {
    [native code]
}

SIPml-api.js (line 2)
ICE servers:[{"url":"stun:23.21.150.121:3478"},{"url":"stun:216.93.246.18:3478"},{"url":"stun:66.228.45.110:3478"},{"url":"stun:173.194.78.127:19302"}]
SIPml-api.js (line 2)
==stack event = m_permission_requested
SIPml-api.js (line 2)
==session event = connecting
SIPml-api.js (line 2)
onGetUserMediaSuccess
SIPml-api.js (line 2)
createOffer
SIPml-api.js (line 2)
==stack event = m_permission_accepted
SIPml-api.js (line 2)
==session event = m_stream_audio_local_added
SIPml-api.js (line 2)
onCreateSdpSuccess
SIPml-api.js (line 2)
onSetLocalDescriptionSuccess
SIPml-api.js (line 2)
onIceGatheringCompleted
SIPml-api.js (line 2)

SEND: INVITE sip:01@192.168.1.28 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKLsO5XprypwV1Q6gakR1IBZpv6k5eqE5x;rport
From: "1060"<sip:1060@192.168.1.28>;tag=uC3esqGT9E84McJENMU4
To: <sip:01@192.168.1.28>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=1060;ha1=1a2e73b7fac658ec7cf668edf6ac0790;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: f3903f11-37e7-f5e8-a4a9-f1d7cb2c70c0
CSeq: 52579 INVITE
Content-Type: application/sdp
Content-Length: 588
Route: <sip:192.168.1.28:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Information Center 08

v=0
o=Mozilla-SIPUA-34.0.5 7398 1 IN IP4 0.0.0.0
s=Doubango Telecom - firefox
t=0 0
a=ice-ufrag:fd694364
a=ice-pwd:97043ddb18ae70bb629675e11f22c263
a=fingerprint:sha-256 44:55:9D:6E:4B:F4:6E:9A:EA:8A:D3:2C:9A:55:A0:B3:45:3A:AF:4A:6F:AA:2D:77:0E:4E:37:A2:70:7B:06:9C
m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 0.0.0.0
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=setup:actpass
a=rtcp-mux

SIPml-api.js (line 2)
5
onIceCandidate = undefined
SIPml-api.js (line 2)
ICE GATHERING COMPLETED!
SIPml-api.js (line 2)
onIceGatheringCompleted
SIPml-api.js (line 2)
onIceGatheringCompleted but no local sdp request is pending
SIPml-api.js (line 2)
__tsip_transport_ws_onmessage
SIPml-api.js (line 2)

recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.3;branch=z9hG4bKLsO5XprypwV1Q6gakR1IBZpv6k5eqE5x
From: "1060"<sip:1060@192.168.1.28>;tag=uC3esqGT9E84McJENMU4
To: <sip:01@192.168.1.28>;tag=as5190de01
Call-ID: f3903f11-37e7-f5e8-a4a9-f1d7cb2c70c0
CSeq: 52579 INVITE
Content-Length: 0
Server: Asterisk PBX 11.14.2
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="asterisk",nonce="1c38f320",stale=FALSE,algorithm=MD5

SIPml-api.js (line 2)

SEND: ACK sip:01@192.168.1.28 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKLsO5XprypwV1Q6gakR1IBZpv6k5eqE5x;rport
From: "1060"<sip:1060@192.168.1.28>;tag=uC3esqGT9E84McJENMU4
To: <sip:01@192.168.1.28>;tag=as5190de01
Call-ID: f3903f11-37e7-f5e8-a4a9-f1d7cb2c70c0
CSeq: 52579 ACK
Content-Length: 0
Route: <sip:192.168.1.28:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70

SIPml-api.js (line 2)
State machine: x0000_Any_2_Any_X_i401_407_INVITE
SIPml-api.js (line 2)

SEND: INVITE sip:01@192.168.1.28 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKhkgg4phxQKau7CbdGTgyNntvWrRid6ac;rport
From: "1060"<sip:1060@192.168.1.28>;tag=uC3esqGT9E84McJENMU4
To: <sip:01@192.168.1.28>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=1060;ha1=1a2e73b7fac658ec7cf668edf6ac0790;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: f3903f11-37e7-f5e8-a4a9-f1d7cb2c70c0
CSeq: 52580 INVITE
Content-Type: application/sdp
Content-Length: 588
Route: <sip:192.168.1.28:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="asterisk",nonce="1c38f320",uri="sip:01@192.168.1.28",response="46b3bdf52fc76b9fd66a3288f27aeb60",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Information Center 08

v=0
o=Mozilla-SIPUA-34.0.5 7398 1 IN IP4 0.0.0.0
s=Doubango Telecom - firefox
t=0 0
a=ice-ufrag:fd694364
a=ice-pwd:97043ddb18ae70bb629675e11f22c263
a=fingerprint:sha-256 44:55:9D:6E:4B:F4:6E:9A:EA:8A:D3:2C:9A:55:A0:B3:45:3A:AF:4A:6F:AA:2D:77:0E:4E:37:A2:70:7B:06:9C
m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 0.0.0.0
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=setup:actpass
a=rtcp-mux

SIPml-api.js (line 2)
__tsip_transport_ws_onmessage
SIPml-api.js (line 2)

recv=SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.3;branch=z9hG4bKhkgg4phxQKau7CbdGTgyNntvWrRid6ac
From: "1060"<sip:1060@192.168.1.28>;tag=uC3esqGT9E84McJENMU4
To: <sip:01@192.168.1.28>
Contact: <sip:01@192.168.1.28:5060;transport=WS>
Call-ID: f3903f11-37e7-f5e8-a4a9-f1d7cb2c70c0
CSeq: 52580 INVITE
Content-Length: 0
Server: Asterisk PBX 11.14.2
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

SIPml-api.js (line 2)
State machine: x0000_Any_2_Any_X_i1xx
SIPml-api.js (line 2)
==session event = i_ao_request
SIPml-api.js (line 2)
__tsip_transport_ws_onmessage
SIPml-api.js (line 2)

recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.3;branch=z9hG4bKhkgg4phxQKau7CbdGTgyNntvWrRid6ac
From: "1060"<sip:1060@192.168.1.28>;tag=uC3esqGT9E84McJENMU4
To: <sip:01@192.168.1.28>;tag=as69c9f93a
Contact: <sip:01@192.168.1.28:5060;transport=WS>
Call-ID: f3903f11-37e7-f5e8-a4a9-f1d7cb2c70c0
CSeq: 52580 INVITE
Content-Length: 0
Server: Asterisk PBX 11.14.2
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

SIPml-api.js (line 2)
State machine: x0000_Any_2_Any_X_i1xx
SIPml-api.js (line 2)
==session event = i_ao_request
SIPml-api.js (line 2)
__tsip_transport_ws_onmessage
SIPml-api.js (line 2)

recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.3;branch=z9hG4bKhkgg4phxQKau7CbdGTgyNntvWrRid6ac
From: "1060"<sip:1060@192.168.1.28>;tag=uC3esqGT9E84McJENMU4
To: <sip:01@192.168.1.28>;tag=as69c9f93a
Contact: <sip:01@192.168.1.28:5060;transport=WS>
Call-ID: f3903f11-37e7-f5e8-a4a9-f1d7cb2c70c0
CSeq: 52580 INVITE
Content-Type: application/sdp
Content-Length: 537
Server: Asterisk PBX 11.14.2
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

v=0
o=root 1107901304 1107901304 IN IP4 192.168.1.28
s=Asterisk PBX 11.14.2
c=IN IP4 192.168.1.28
t=0 0
m=audio 16776 RTP/SAVPF 109 0 8 101
a=rtpmap:109 opus/48000/2
a=maxptime:60
a=fmtp:109 maxplaybackrate=16000; stereo=0; sprop-stereo=0; useinbandfec=0
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=connection:new
a=setup:active
a=fingerprint:SHA-256 B2:72:2B:AF:A0:AC:6C:24:8D:48:04:C5:52:7C:DC:61:C9:CB:FF:EB:B4:38:93:B8:D1:42:AE:27:E7:E1:BC:E2
a=sendrecv

SIPml-api.js (line 2)
State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE
SIPml-api.js (line 2)

setRemoteDescription(answer)
v=0
o=root 1107901304 1107901304 IN IP4 192.168.1.28
s=Asterisk PBX 11.14.2
c=IN IP4 192.168.1.28
t=0 0
m=audio 16776 RTP/SAVPF 109 0 8 101
a=rtpmap:109 opus/48000/2
a=maxptime:60
a=fmtp:109 maxplaybackrate=16000; stereo=0; sprop-stereo=0; useinbandfec=0
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=connection:new
a=setup:active
a=fingerprint:SHA-256 B2:72:2B:AF:A0:AC:6C:24:8D:48:04:C5:52:7C:DC:61:C9:CB:FF:EB:B4:38:93:B8:D1:42:AE:27:E7:E1:BC:E2
a=sendrecv

SIPml-api.js (line 2)

SEND: ACK sip:01@192.168.1.28:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKe1nMmm03SoZL4ZiTnhJh;rport
From: "1060"<sip:1060@192.168.1.28>;tag=uC3esqGT9E84McJENMU4
To: <sip:01@192.168.1.28>;tag=as69c9f93a
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: f3903f11-37e7-f5e8-a4a9-f1d7cb2c70c0
CSeq: 52580 ACK
Content-Length: 0
Route: <sip:192.168.1.28:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="asterisk",nonce="1c38f320",uri="sip:01@192.168.1.28:5060;transport=WS",response="b236f7f948f04b90afbdc7e306769826",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Information Center 08

SIPml-api.js (line 2)
==session event = m_early_media
SIPml-api.js (line 2)
==session event = connected
SIPml-api.js (line 2)
onSetRemoteDescriptionError
SIPml-api.js (line 2)
[object Object]
SIPml-api.js (line 2)
State machine: tsip_dialog_register_Connected_2_InProgress_X_oRegister
SIPml-api.js (line 2)

SEND: REGISTER sip:192.168.1.28 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvLUiyi2lZkb7TSHTVG8U1vlTM3TTkn4m;rport
From: "1060"<sip:1060@192.168.1.28>;tag=yw8fBDsrmt3IS0F15OoD
To: "1060"<sip:1060@192.168.1.28>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 57f8fe29-98fd-6993-7d4e-5bd62c0208dc
CSeq: 5490 REGISTER
Content-Length: 0
Route: <sip:192.168.1.28:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="asterisk",nonce="59bc2834",uri="sip:192.168.1.28",response="972eb54b77fda5fc7e14e82e4534b8de",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Information Center 08

SIPml-api.js (line 2)
==session event = sent_request
SIPml-api.js (line 2)
__tsip_transport_ws_onmessage
SIPml-api.js (line 2)

recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.3;branch=z9hG4bKvLUiyi2lZkb7TSHTVG8U1vlTM3TTkn4m
From: "1060"<sip:1060@192.168.1.28>;tag=yw8fBDsrmt3IS0F15OoD
To: "1060"<sip:1060@192.168.1.28>;tag=as32604e9b
Call-ID: 57f8fe29-98fd-6993-7d4e-5bd62c0208dc
CSeq: 5490 REGISTER
Content-Length: 0
Server: Asterisk PBX 11.14.2
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="asterisk",nonce="5adc39a0",stale=FALSE,algorithm=MD5

SIPml-api.js (line 2)
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494
SIPml-api.js (line 2)

SEND: REGISTER sip:192.168.1.28 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKaILU91M4DHTIxwG2IWhA2O2qHNOozdwu;rport
From: "1060"<sip:1060@192.168.1.28>;tag=yw8fBDsrmt3IS0F15OoD
To: "1060"<sip:1060@192.168.1.28>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 57f8fe29-98fd-6993-7d4e-5bd62c0208dc
CSeq: 5491 REGISTER
Content-Length: 0
Route: <sip:192.168.1.28:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="asterisk",nonce="5adc39a0",uri="sip:192.168.1.28",response="d9770acf66a94e9db6b3645675223297",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Information Center 08

SIPml-api.js (line 2)
==session event = sent_request
SIPml-api.js (line 2)
__tsip_transport_ws_onmessage
SIPml-api.js (line 2)

recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.3;branch=z9hG4bKaILU91M4DHTIxwG2IWhA2O2qHNOozdwu
From: "1060"<sip:1060@192.168.1.28>;tag=yw8fBDsrmt3IS0F15OoD
To: "1060"<sip:1060@192.168.1.28>;tag=as32604e9b
Contact: <sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200
Call-ID: 57f8fe29-98fd-6993-7d4e-5bd62c0208dc
CSeq: 5491 REGISTER
Expires: 200
Content-Length: 0
Server: Asterisk PBX 11.14.2
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Date: 22 Dec 2014 13:52:00 GMT;22

SIPml-api.js (line 2)
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx

Also I had some problems to install opus codec drivers but now it’s fine.
I don’t know where is the problem.

Any help will be appreciated.

Best Regards,
Paata Lominadze.

david55 · December 22, 2014, 4:11pm

Mozilla has requested the call on hold, using an obsolete way of doing so (address=0.0.0.0) and has not unheld it when the call is answered.

paatalom · December 22, 2014, 5:15pm

Thank you very much davit for your reply.

I’m very newbies on asterisk and sorry for my misunderstanding
for client I’m using simpl5 javascript library. I got it from git repository :

sipml5.googlecode.com/svn/trunk

I will post my client code and server configs more detail.
I don’t know what to change to work this example.
if have some time to help I’ll be appreciated.

http.conf

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

2.rtp.conf

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

3.sip.conf

;  Geneal Available Configurations 
[general]
context=public
allowguest=yes
allowoverlap=no
udpbindaddr=0.0.0.0:5060
bindport=5060
port=5060
realm=192.168.1.28
tcpenable=yes
tcpbindaddr=0.0.0.0
transport=udp,ws
srvlookup=yes
videosupport=no

; Sip Accounts : 
[1060] ; This will be WebRTC client
type=friend
username=1060 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=1060 ; The SIP Password for SIP.js
encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF for this peer
icesupport=yes ; Tell Asterisk to use ICE for this peer
context=08_pbx_out ; Tell Asterisk which context to use when this peer is dialing
directmedia=no ; Asterisk will relay media for this peer
transport=udp,ws,wss ; Asterisk will allow this peer to register on UDP or WebSockets
force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11
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
disallow=all
allow=opus,ulaw,alaw,h264,h263p,mpeg4   ;(or h263, h264)
;nat=no
;dial=SIP/1060
callerid=1060 <1060>



[1061] ; This will be WebRTC client
type=friend
username=1061 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=1061 ; The SIP Password for SIP.js
encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF for this peer
icesupport=yes ; Tell Asterisk to use ICE for this peer
context=08_pbx_out ; Tell Asterisk which context to use when this peer is dialing
directmedia=no ; Asterisk will relay media for this peer
transport=udp,ws,wss ; Asterisk will allow this peer to register on UDP or WebSockets
force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11
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
disallow=all
allow=opus,ulaw,alaw,h264,h263p,mpeg4   ;(or h263, h264)
;nat=no
;dial=SIP/1061
callerid=1061 <1061>

All required ports are open and firewall is disabled :

tcp        0      0 0.0.0.0:8088                0.0.0.0:*                   LISTEN
tcp        0      0 0.0.0.0:5060                0.0.0.0:*                   LISTEN
udp        0      0 0.0.0.0:5060                0.0.0.0:*

Sipml5 Sip Client Registration code :

oSipStack = new SIPml.Stack(
                                {
                                        realm : "192.168.1.28",
                                        impi : sipUsername,
                                        impu : "sip:"+sipUsername+"@192.168.1.28",
                                        password : password,
                                        display_name : sipUsername,
                                        websocket_proxy_url : 'ws://192.168.1.28:8088/ws',
                                        outbound_proxy_url : 'udp://192.168.1.28:5060',
                                        ice_servers : [{ url: 'stun:stun.l.google.com:19302'}],
                                        enable_rtcweb_breaker : (true),

                                        username :sipUsername,
                                        credential :password,

                                        events_listener : {
                                                events : '*',
                                                listener : onSipEventStack
                                        },
                                        enable_early_ims : (false), 
                                        enable_media_stream_cache : (false),
                                        bandwidth : (null), // could be redefined a session-level
                                        video_size : (null), // could be redefined a session-level
                                        sip_headers : [ {
                                                name : 'User-Agent',
                                                value : 'IM-client/OMA1.0 sipML5-v1.2014.04.18'
                                        }, {
                                                name : 'Organization',
                                                value : 'Information Center 08'
                                        } ]
                                });

                if (oSipStack.start() != 0) {
                        window.console.info('Failed to start the SIP stack');
                        return -101;
                } else
                        window.console.info('Sip Registered Successfully');
                        return 0;

Simpl5 Client call Code :

oSipSessionCall = oSipStack.newSession("call-audio", oConfigCall);
                // make call
                if (oSipSessionCall.call("01") != 0) {
                        oSipSessionCall = null;
                        window.console.info('Failed to make call');
                        return;
}

rtp.log (Small part from rtp output)

Got  RTP packet from    192.168.1.9:40018 (type 08, seq 029360, ts 112774, len 000160)
Sent RTP packet to      0.0.0.0:9 (type 109, seq 057443, ts 001920, len 000003)
Got  RTP packet from    192.168.1.9:40018 (type 08, seq 029361, ts 112934, len 000160)
Sent RTP packet to      0.0.0.0:9 (type 109, seq 057444, ts 002880, len 000003)
Got  RTP packet from    192.168.1.9:40018 (type 08, seq 029362, ts 113094, len 000160)
Sent RTP packet to      0.0.0.0:9 (type 109, seq 057445, ts 003840, len 000003)
Got  RTP packet from    192.168.1.9:40018 (type 08, seq 029363, ts 113254, len 000160)
Sent RTP packet to      0.0.0.0:9 (type 109, seq 057446, ts 004800, len 000003)
Got  RTP packet from    192.168.1.9:40018 (type 08, seq 029364, ts 113414, len 000160)
Sent RTP packet to      0.0.0.0:9 (type 109, seq 057447, ts 005760, len 000003)
Got  RTP packet from    192.168.1.9:40018 (type 08, seq 029365, ts 113574, len 000160)
Sent RTP packet to      0.0.0.0:9 (type 109, seq 057448, ts 006720, len 000003)
Got  RTP packet from    192.168.1.9:40018 (type 08, seq 029366, ts 113734, len 000160)
Sent RTP packet to      0.0.0.0:9 (type 109, seq 057449, ts 007680, len 000003)

somewhere here i found that this log must contains “(Via ICE)” text for correct usage of ice . for this i have installed uuid-devel and libuuid-devel, and after tha I have compiled asterisk again. but this text is not visible into my output.

That’s all what I have configured for this example.
I don’t know what is the correct way to call (how i understand this problem is connected to simpl5 not asterisk)
do you know how fix this ?

Thank you very much again david.

Best Regards,
Paata Lominadze

navaismo · December 22, 2014, 5:24pm

You should try wit null ice server in your js config. Seems like your client cant get the local IP addres and send 0.0.0.0

paatalom · December 22, 2014, 5:41pm

hello navaismo,

I am excited so good support from this forum.
Thank you for your reply.

Previously I had this “ice server” to null value but the result was the same. now i set this variable to null but no success.

My Client host is linux :

paata@wolf:~> cat /etc/os-release 
NAME=openSUSE
VERSION="13.2 (Harlequin)"
VERSION_ID="13.2"
PRETTY_NAME="openSUSE 13.2 (Harlequin) (x86_64)"
ID=opensuse
ANSI_COLOR="0;32"
CPE_NAME="cpe:/o:opensuse:opensuse:13.2"
BUG_REPORT_URL="https://bugs.opensuse.org"
HOME_URL="https://opensuse.org/"
ID_LIKE="suse"

I don’t know why this js api can’t get my ip address to send.

Best Regards,
Paata Lominadze.

navaismo · December 22, 2014, 6:20pm

using the Sipml demo works?

jairomolina · January 20, 2015, 7:01pm

Similar problem here (no audio), using sipML demo page.

The address in INVITE:

<— SIP read from WS:IP:port —>

Is right but in message body it is:

c=IN IP4 0.0.0.0

Asterisk 11.7, CentOS 6 x86_64, Firefox 35.0.

Tks.

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