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Uncaught ReferenceError: tsdp_message is not defined (tsdp_parser_message.js:117)

What steps will reproduce the problem?
1.
2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 31 Mar 2012 at 12:01

Set port to zero if no codec (fmt)

What steps will reproduce the problem?
1.
2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 9 Apr 2012 at 8:04

Inbound calls from Asterisk fail

Using Asterisk revision 372699 plus the sipml5 patches
sipml5-read-only svn Rev: 77
Chrome 23.0.1263.0 canary (also occurs on dev and last couple of days canary)

Outbound calls from chrome work fine but inbound fail at session negotiation 
phase.

INVITE from Asterisk looks OK but stack responds with SDP in INVITE which 
contains a huge number of mostly duplicate a=candidate lines, this overflows a 
max SDP buffer size in Asterisk and causes it to junk the response and the 
dialog eventually times out.

Attached Javascript console log and Asterisk "sip debug peer" trace.

Original issue reported on code.google.com by [email protected] on 11 Sep 2012 at 4:16

Attachments:

www.sipML5.org/call.htm doesn't work!

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

What steps will reproduce the problem?
1.using chrome 19.0.1084.52 m tp open www.sipML5.org/call.htm
2.There have a Error Code in the console of developer
3.

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


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


Please provide any additional information below.

Original issue reported on code.google.com by [email protected] on 31 May 2012 at 6:56

Attachments:

error.www.sipML5.org.png

SYNTAX_ERR: DOM Exception 12 When Incoming call comes

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

What steps will reproduce the problem?
1.get the latest copy from SVN 
2. launch it into Chrome (ver. 21.0.1180.89 m) in 2 machines in the LAN and 
register those 2 with webRTCSip proxy.
3. Call to the other extension in the LAN

What is the expected output? What do you see instead?
Expected the "Incoming call... " message.
But I do not see it but remove video panel expand with white background.

What version of the product are you using? On what operating system?
I am using latest SIPML5 sip Client with latest webrtc2sip proxy on Cent OS 6.0

I have attached the javascript console with this.
Please advice.
Best Regards,
Ajith 



Please provide any additional information below.

Original issue reported on code.google.com by [email protected] on 12 Sep 2012 at 1:04

Attachments:

error.txt

click2call

What steps will reproduce the problem?
1.
2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 30 Mar 2012 at 11:34

load specific revision?


The sipml5 load js at initialization with svn parameter. Is it really need 
specific revision or..?
http://invalid.url/src/tinyMEDIA/src/tmedia_api.js?svn=5

Original issue reported on code.google.com by [email protected] on 17 Jun 2012 at 11:49

bug with a chrome Version 20.0.1132.11 dev in Linux (ubuntu)

i use ubuntu system (12.04) and Google chrome version dev i have a  issue with 
a procedure registration a button 'login' is locked when i click this 
everything doesn't happen.And i get some error with a js so i attach some file 
about this bug

Original issue reported on code.google.com by [email protected] on 20 May 2012 at 4:41

Attachments:

488 (Reason failed to match PRACK) when PRACK is retransmitted

What steps will reproduce the problem?
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3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 30 Mar 2012 at 9:13

reg-id fro contact match (expires)

What steps will reproduce the problem?
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2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 2 Apr 2012 at 12:26

Add support for Bowser (Ericsson's WebRTC implementation)

What steps will reproduce the problem?
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What is the expected output? What do you see instead?


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Original issue reported on code.google.com by [email protected] on 22 Oct 2012 at 4:36

Auth header is not implemented for generic request

sipml5 do not add auth header to generic request and reaction on 401/407 is not 
implemented too.

tsip_dialog_generic.js
function tsip_dialog_generic_InProgress_2_InProgress_X_401_407_421_494(ao_args) 
{
    console.error("Not implemented");
    return 0;
}

Original issue reported on code.google.com by [email protected] on 22 May 2012 at 10:36

Make it more like library


It is not issue, just a feature request.
1. Ability to change default path, now tsip_api.js load everything from /src 
path. It possible to change it of course, but it will be nice to have straight 
way for this, may be root path as param?
2. Minified version

Original issue reported on code.google.com by [email protected] on 31 May 2012 at 5:31

No audio with Freeswitch

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

What steps will reproduce the problem?
1. register two sipml5 clients on freeswitch
2. send invite from one client to the other
3. the other side not ring. in js console, we can see the message come in and 
generate DOMException.
4. please reference forum thread: 
https://groups.google.com/forum/?hl=en&fromgroups=#!topic/doubango/wiiBOxlxSVo

What is the expected output? What do you see instead?
expect phone ring. but not ring and DOMException

What version of the product are you using? On what operating system?
sipml5 r78 version. Mac Chrome version 21.0.1180.89

Please provide any additional information below.

> I downloaded latest r78 codes and disabled video. 
> 
> I registered both webrtc clients on freeswitch and try to call each other. 
> I still got similar error: 
> 
> __tsip_transport_ws_onmessage 
tsk_utils.js:55<http://184.73.86.199/sipml5-read-only/src/tinySAK/src/tsk_utils.
js?svn=9> 
> recv=INVITE sip:[email protected]:63789;transport=WS SIP/2.0 Via: 
> SIP/2.0/WS 
> 87.106.69.240:5062;rport;branch=z9hG4bK-524287-1---d605ab0781319063 From: 
> "chaofeng"<sip:[email protected]>;tag=Fjv70F751c33m To: 
> 
<sip:[email protected]:5060;transport=udp;ws-src-ip=173.73.161.29;ws-src-
port=63789 > 
> Contact: <sip:[email protected]:5060> Call-ID: 
> 088b9c8e-7666-1230-a78e-12313907bd07 CSeq: 33331514 INVITE Content-Type: 
> application/sdp Content-Length: 207 Via: SIP/2.0/UDP 
> 10.209.190.245;rport=5060;received=23.23.237.155;branch=z9hG4bKBZD0UXvry2m9 D 
> Max-Forwards: 67 Allow: 
> INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,UPDATE,INFO,REGISTER,REFER,NOTIFY,PUB 
LISH,SUBSCRIBE 
> Content-Disposition: session Supported: timer,precondition,path,replaces 
> User-Agent: FreeSWITCH-mod_sofia/1.3.0+git~20120906T224301Z~a7c791d2de 
> Allow-Events: 
> talk,hold,conference,presence,dialog,line-seize,call-info,sla,include-sessi 
on-description,presence.winfo,message-summary,refer 
> Remote-Party-ID: "chaofeng" 
> <sip:[email protected]>;party=calling;screen=yes;privacy=off 
> X-FS-Support: update_display,send_info v=0 o=FreeSWITCH 1347314882 
> 1347314883 IN IP4 10.209.190.245 s=FreeSWITCH c=IN IP4 10.209.190.245 t=0 0 
> m=audio 20274 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 
> a=fmtp:101 0-16 a=ptime:20 
tsk_utils.js:55<http://184.73.86.199/sipml5-read-only/src/tinySAK/src/tsk_utils.
js?svn=9> 
> SEND: SIP/2.0 100 Trying (sent from the Transaction Layer) Via: SIP/2.0/WS 
> 87.106.69.240:5062;rport=5062;branch=z9hG4bK-524287-1---d605ab0781319063 
> From: "chaofeng"<sip:[email protected]>;tag=Fjv70F751c33m To: 
> 
<sip:[email protected]:5060;transport=udp;ws-src-ip=173.73.161.29;ws-src-
port=63789 > 
> Call-ID: 088b9c8e-7666-1230-a78e-12313907bd07 CSeq: 33331514 INVITE 
> Content-Length: 0 Via: SIP/2.0/UDP 
> 10.209.190.245;rport=5060;received=23.23.237.155;branch=z9hG4bKBZD0UXvry2m9 D 
> 
tsk_utils.js:55<http://184.73.86.199/sipml5-read-only/src/tinySAK/src/tsk_utils.
js?svn=9> 
> 
>    1. 
>    DOMException 
>    1. code: 12 
>       2. message: "SYNTAX_ERR: DOM Exception 12" 
>       3. name: "SYNTAX_ERR" 
>       4. stack: "Error: An invalid or illegal string was specified.↵ at 
>       tmedia_session_jsep.__set_ro 
>       
(http://184.73.86.199/sipml5-read-only/src/tinyMEDIA/src/tmedia_sessio... 
>       at tmedia_session_jsep.__get_lo 
>       
(http://184.73.86.199/sipml5-read-only/src/tinyMEDIA/src/tmedia_sessio... 
>       at tmedia_session.get_lo 
>       
(http://184.73.86.199/sipml5-read-only/src/tinyMEDIA/src/tmedia_sessio... 
>       at tmedia_session_mgr.get_lo 
>       
(http://184.73.86.199/sipml5-read-only/src/tinyMEDIA/src/tmedia_sessio... 
>       at tmedia_session_mgr.set_ro 
>       
(http://184.73.86.199/sipml5-read-only/src/tinyMEDIA/src/tmedia_sessio... 
>       at tsip_dialog_invite.process_ro 
>       
(http://184.73.86.199/sipml5-read-only/src/tinySIP/src/dialogs/tsip_di... 
>       at tsk_fsm_entry.__tsip_dialog_invite_cond_is_bad_content [as 
fn_condition] 
>       
(http://184.73.86.199/sipml5-read-only/src/tinySIP/src/dialogs/tsip_di... 
>       at tsk_fsm.act 
>       
(http://184.73.86.199/sipml5-read-only/src/tinySAK/src/tsk_fsm.js?svn=... 
>       at tsip_dialog.fsm_act 
>       
(http://184.73.86.199/sipml5-read-only/src/tinySIP/src/dialogs/tsip_di... 
>       at tsip_dialog.__tsip_dialog_invite_event_callback [as fn_callback] 
>       
(http://184.73.86.199/sipml5-read-only/src/tinySIP/src/dialogs/tsip_di...)" 
>       5. __proto__: DOMException 
>    
tsk_utils.js:67<http://184.73.86.199/sipml5-read-only/src/tinySAK/src/tsk_utils.
js?svn=9> 
>       1. 
tsk_utils_log_errortsk_utils.js:67<http://184.73.86.199/sipml5-read-only/src/tin
ySAK/src/tsk_utils.js?svn=9> 
>       2. 
tmedia_session_jsep.__set_rotmedia_session_jsep.js:231<http://184.73.86.199/sipm
l5-read-only/src/tinyMEDIA/src/tmedia_sessio...> 
>       3. 
tmedia_session_jsep.__get_lotmedia_session_jsep.js:127<http://184.73.86.199/sipm
l5-read-only/src/tinyMEDIA/src/tmedia_sessio...> 
>       4. 
tmedia_session.get_lotmedia_session.js:563<http://184.73.86.199/sipml5-read-only
/src/tinyMEDIA/src/tmedia_sessio...> 
>       5. 
tmedia_session_mgr.get_lotmedia_session.js:293<http://184.73.86.199/sipml5-read-
only/src/tinyMEDIA/src/tmedia_sessio...> 
>       6. 
tmedia_session_mgr.set_rotmedia_session.js:397<http://184.73.86.199/sipml5-read-
only/src/tinyMEDIA/src/tmedia_sessio...> 
>       7. 
tsip_dialog_invite.process_rotsip_dialog_invite.js:730<http://184.73.86.199/sipm
l5-read-only/src/tinySIP/src/dialogs/tsip_di...> 
>       8. __tsip_dialog_invite_cond_is_bad_content 
>       
tsip_dialog_invite__server.js:96<http://184.73.86.199/sipml5-read-only/src/tinyS
IP/src/dialogs/tsip_di...> 
>       9. 
tsk_fsm.acttsk_fsm.js:82<http://184.73.86.199/sipml5-read-only/src/tinySAK/src/t
sk_fsm.js?svn=9> 
>       10. 
tsip_dialog.fsm_acttsip_dialog.js:724<http://184.73.86.199/sipml5-read-only/src/
tinySIP/src/dialogs/tsip_di...> 
>       11. 
__tsip_dialog_invite_event_callbacktsip_dialog_invite.js:894<http://184.73.86.19
9/sipml5-read-only/src/tinySIP/src/dialogs/tsip_di...> 
>       12. 
tsip_dialog.callbacktsip_dialog.js:709<http://184.73.86.199/sipml5-read-only/src
/tinySIP/src/dialogs/tsip_di...> 
>       13. __tsip_transac_ist_Started_2_Proceeding_X_INVITE 
>       
tsip_transac_ist.js:184<http://184.73.86.199/sipml5-read-only/src/tinySIP/src/tr
ansactions/ts...> 
>       14. 
tsk_fsm.acttsk_fsm.js:91<http://184.73.86.199/sipml5-read-only/src/tinySAK/src/t
sk_fsm.js?svn=9> 
>       15. 
tsip_transac.fsm_acttsip_transac.js:123<http://184.73.86.199/sipml5-read-only/sr
c/tinySIP/src/transactions/ts...> 
>       16. 
tsip_transac_ist.starttsip_transac_ist.js:142<http://184.73.86.199/sipml5-read-o
nly/src/tinySIP/src/transactions/ts...> 
>       17. 
tsip_dialog_layer.handle_incoming_messagetsip_dialog_layer.js:238<http://184.73.
86.199/sipml5-read-only/src/tinySIP/src/dialogs/tsip_di...> 
>       18. tsip_transport_layer.handle_incoming_message 
>       
tsip_transport_layer.js:231<http://184.73.86.199/sipml5-read-only/src/tinySIP/sr
c/transports/tsip...> 
>       19. 
__tsip_transport_ws_onmessagetsip_transport.js:425<http://184.73.86.199/sipml5-r
ead-only/src/tinySIP/src/transports/tsip...> 
> 
>    1. 
>    TypeError 
>    1. arguments: Array[2] 
>       2. get message: function () { [native code] } 
>       3. get stack: function () { [native code] } 
>       4. set message: function () { [native code] } 
>       5. set stack: function () { [native code] } 
>       6. type: "non_object_property_call" 
>       7. __proto__: Error 
>    
tsk_utils.js:67<http://184.73.86.199/sipml5-read-only/src/tinySAK/src/tsk_utils.
js?svn=9> 
>       1. 
tsk_utils_log_errortsk_utils.js:67<http://184.73.86.199/sipml5-read-only/src/tin
ySAK/src/tsk_utils.js?svn=9> 
>       2. 
tsk_fsm.acttsk_fsm.js:96<http://184.73.86.199/sipml5-read-only/src/tinySAK/src/t
sk_fsm.js?svn=9> 
>       3. 
tsip_transac.fsm_acttsip_transac.js:123<http://184.73.86.199/sipml5-read-only/sr
c/tinySIP/src/transactions/ts...> 
>       4. 
tsip_transac_ist.starttsip_transac_ist.js:142<http://184.73.86.199/sipml5-read-o
nly/src/tinySIP/src/transactions/ts...> 
>       5. 
tsip_dialog_layer.handle_incoming_messagetsip_dialog_layer.js:238<http://184.73.
86.199/sipml5-read-only/src/tinySIP/src/dialogs/tsip_di...> 
>       6. tsip_transport_layer.handle_incoming_message 
>       
tsip_transport_layer.js:231<http://184.73.86.199/sipml5-read-only/src/tinySIP/sr
c/transports/tsip...> 
>       7. 
__tsip_transport_ws_onmessagetsip_transport.js:425<http://184.73.86.199/sipml5-r
ead-only/src/tinySIP/src/transports/tsip...> 
> 
> Media Added 
tsk_utils.js:55<http://184.73.86.199/sipml5-read-only/src/tinySAK/src/tsk_utils.
js?svn=9> 
> __tsip_transport_ws_onmessage 
tsk_utils.js:55<http://184.73.86.199/sipml5-read-only/src/tinySAK/src/tsk_utils.
js?svn=9> 
> recv=CANCEL sip:[email protected]:63789;transport=WS SIP/2.0 Via: 
> SIP/2.0/WS 
> 87.106.69.240:5062;rport;branch=z9hG4bK-524287-1---d605ab0781319063 From: 
> "chaofeng"<sip:[email protected]>;tag=Fjv70F751c33m To: 
> 
<sip:[email protected]:5060;transport=udp;ws-src-ip=173.73.161.29;ws-src-
port=63789 > 
> Call-ID: 088b9c8e-7666-1230-a78e-12313907bd07 CSeq: 33331514 CANCEL 
> Content-Length: 0 Max-Forwards: 70 
tsk_utils.js:55<http://184.73.86.199/sipml5-read-only/src/tinySAK/src/tsk_utils.
js?svn=9> 
> State machine: tsip_dialog_invite_Started_2_Started_X_any 
>

Original issue reported on code.google.com by [email protected] on 14 Sep 2012 at 2:46

send 200 OK to NOTIFY(Event: keep-alive)

What steps will reproduce the problem?
1.
2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 4 Apr 2012 at 12:37

DomEX when a non ip-phone number call me

I used the demo to connect with my phone-ip number. i can call/receive a call 
from some sip provider as ekiga. But when a non ip call me, the windows of 
video show up and i god a dom exeption, so no sound no video.
when i use my number with a ip-phone i can receive the call without any problem.

i'm using chrome with opensuse 12.2


in the console i got: 
/src/tinySAK/src/tsk_utils.js?svn=10:55
DOMException
code: 12
message: "SYNTAX_ERR: DOM Exception 12"
name: "SYNTAX_ERR"
stack: "Error: An invalid or illegal string was specified.↵    at 
tmedia_session_jsep.__set_ro 
(http://sipml5.org/src/tinyMEDIA/src/tmedia_session_jsep.js?svn=10:222:23)↵   
 at tmedia_session_jsep.__get_lo 
(http://sipml5.org/src/tinyMEDIA/src/tmedia_session_jsep.js?svn=10:127:18)↵   
 at tmedia_session.get_lo 
(http://sipml5.org/src/tinyMEDIA/src/tmedia_session.js?svn=10:563:17)↵    at 
tmedia_session_mgr.get_lo 
(http://sipml5.org/src/tinyMEDIA/src/tmedia_session.js?svn=10:293:45)↵    at 
tmedia_session_mgr.set_ro 
(http://sipml5.org/src/tinyMEDIA/src/tmedia_session.js?svn=10:397:11)↵    at 
tsip_dialog_invite.process_ro 
(http://sipml5.org/src/tinySIP/src/dialogs/tsip_dialog_invite.js?svn=10:730:39)�
��    at tsk_fsm_entry.__tsip_dialog_invite_cond_is_bad_content [as 
fn_condition] 
(http://sipml5.org/src/tinySIP/src/dialogs/tsip_dialog_invite__server.js?svn=10:
96:27)↵    at tsk_fsm.act 
(http://sipml5.org/src/tinySAK/src/tsk_fsm.js?svn=10:82:46)↵    at 
tsip_dialog.fsm_act 
(http://sipml5.org/src/tinySIP/src/dialogs/tsip_dialog.js?svn=10:726:23)↵    
at tsip_dialog.__tsip_dialog_invite_event_callback [as fn_callback] 
(http://sipml5.org/src/tinySIP/src/dialogs/tsip_dialog_invite.js?svn=10:894:32)"
__proto__: DOMException
 /src/tinySAK/src/tsk_utils.js?svn=10:67
tsk_utils_log_error /src/tinySAK/src/tsk_utils.js?svn=10:67
tmedia_session_jsep.__set_ro 
/src/tinyMEDIA/src/tmedia_session_jsep.js?svn=10:230
tmedia_session_jsep.__get_lo 
/src/tinyMEDIA/src/tmedia_session_jsep.js?svn=10:127
tmedia_session.get_lo /src/tinyMEDIA/src/tmedia_session.js?svn=10:563
tmedia_session_mgr.get_lo /src/tinyMEDIA/src/tmedia_session.js?svn=10:293
tmedia_session_mgr.set_ro /src/tinyMEDIA/src/tmedia_session.js?svn=10:397
tsip_dialog_invite.process_ro 
/src/tinySIP/src/dialogs/tsip_dialog_invite.js?svn=10:730
__tsip_dialog_invite_cond_is_bad_content 
/src/tinySIP/src/dialogs/tsip_dialog_invite__server.js?svn=10:96
tsk_fsm.act /src/tinySAK/src/tsk_fsm.js?svn=10:82
tsip_dialog.fsm_act /src/tinySIP/src/dialogs/tsip_dialog.js?svn=10:726
__tsip_dialog_invite_event_callback 
/src/tinySIP/src/dialogs/tsip_dialog_invite.js?svn=10:894
tsip_dialog.callback /src/tinySIP/src/dialogs/tsip_dialog.js?svn=10:711
__tsip_transac_ist_Started_2_Proceeding_X_INVITE 
/src/tinySIP/src/transactions/tsip_transac_ist.js?svn=10:184
tsk_fsm.act /src/tinySAK/src/tsk_fsm.js?svn=10:91
tsip_transac.fsm_act /src/tinySIP/src/transactions/tsip_transac.js?svn=10:123
tsip_transac_ist.start 
/src/tinySIP/src/transactions/tsip_transac_ist.js?svn=10:142
tsip_dialog_layer.handle_incoming_message 
/src/tinySIP/src/dialogs/tsip_dialog_layer.js?svn=10:238
tsip_transport_layer.handle_incoming_message 
/src/tinySIP/src/transports/tsip_transport_layer.js?svn=10:231
__tsip_transport_ws_onmessage 
/src/tinySIP/src/transports/tsip_transport.js?svn=10:425
TypeError
arguments: Array[2]
get message: function () { [native code] }
get stack: function () { [native code] }
set message: function () { [native code] }
set stack: function () { [native code] }
type: "non_object_property_call"
__proto__: Error
 /src/tinySAK/src/tsk_utils.js?svn=10:67
tsk_utils_log_error /src/tinySAK/src/tsk_utils.js?svn=10:67
tsk_fsm.act /src/tinySAK/src/tsk_fsm.js?svn=10:96
tsip_transac.fsm_act /src/tinySIP/src/transactions/tsip_transac.js?svn=10:123
tsip_transac_ist.start 
/src/tinySIP/src/transactions/tsip_transac_ist.js?svn=10:142
tsip_dialog_layer.handle_incoming_message 
/src/tinySIP/src/dialogs/tsip_dialog_layer.js?svn=10:238
tsip_transport_layer.handle_incoming_message 
/src/tinySIP/src/transports/tsip_transport_layer.js?svn=10:231
__tsip_transport_ws_onmessage

thank you for any help

Original issue reported on code.google.com by [email protected] on 29 Sep 2012 at 7:26

Adds support for MWI

What steps will reproduce the problem?
1.
2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 29 Mar 2012 at 5:55

How to use video call from chrome to chrome and chrome to eyebeam or xlite5


hi all,
How to use video call from chrome to chrome and chrome to eyebeam or xlite5?
Can you tell me?

Thanks!
Ronal.

Original issue reported on code.google.com by [email protected] on 10 Sep 2012 at 7:31

INVITE server transaction OK retransmissions don't stop after ACK

First of all, thank you so much for sipml.

When sipml is a browser UAS, in an initial INVITE server transaction, after 
sipml sends OK and receives the ACK it doesn't stop sending OK retransmissions. 
To me it looks like the stx state machine doesn't connect the Accepted state to 
Confirmed/Terminated/Completed.

Attached is a quick patch for my testing that shortcuts the states. It works 
for me, but I am not sure about the overall purpose of your extended FSM, so it 
might be missing something.

Thanks for any advice.

Original issue reported on code.google.com by [email protected] on 1 Sep 2012 at 9:56

Attachments:

sipml_ACK_fsm_loophole_fix.txt

Rewrite the INVITE's AoR

What steps will reproduce the problem?
1.
2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 29 Mar 2012 at 5:55

Uncaught ReferenceError: tsip_action_type_e is not defined

What steps will reproduce the problem?
1.
2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 4 Apr 2012 at 12:25

Installation of source

What steps will reproduce the problem?
1. Install Linux (Centos)
2. Enable HTTPD
3. Copy source code to root web directory

What is the expected output? What do you see instead?
That the connection is made with SIP

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

Please provide any additional information below.
I go to call.htm (corrected the "sipml5.org" in tsip_stact to hostname of the 
server with source files of sipml5) - Try to connect with Asteris SIP settings 
Results: Nothing happens.

Additional question: Is there a debug for sipml5?

Original issue reported on code.google.com by [email protected] on 24 May 2012 at 10:15

iLBC

What steps will reproduce the problem?
1.
2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 30 Mar 2012 at 11:57

No Audio at all after updating to latest Asterisk Patch

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

What steps will reproduce the problem?
1. Installed Asterisk from scratch from the Doubango page, and downloaded 
latest patch and installed, completed all steps to get Asterisk running.

2. Copied the sip.conf, extensions.conf, http.conf, and users.conf from the 
repository.

3. Added avpf=yes and encryption=yes to user 1061 to make call with Chrome.

4. Registered both users, one on Mac chrome and one on Ubuntu Chrome. (Same 
behavior on Windows XP as well)

5. I can call and connect, but I get no audio at all on either end, and I see 
the following logs in Asterisk. Note the error, which I've added whitespace 
around:

  == Using SIP RTP CoS mark 5
    -- Executing [101@default:1] Dial("SIP/1060-00000002", "SIP/1061") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/1061
    -- SIP/1061-00000003 is ringing

[Sep 14 00:20:49] ERROR[7196][C-00000002]: netsock2.c:269 ast_sockaddr_resolve: 
getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known

[Sep 14 00:20:49] WARNING[7196][C-00000002]: chan_sip.c:15200 
__set_address_from_contact: Invalid host name in Contact: (can't resolve in 
DNS) : 'df7jal23ls0d.invalid'
    -- SIP/1061-00000003 answered SIP/1060-00000002
       > User Agent transport = WS�       > User Agent transport = WS
�  == Spawn extension (default, 101, 1) exited non-zero on 'SIP/1060-00000002'
       > User Agent transport = WS�       > User Agent transport = WS�       > User Agent transport = WS�       > User Agent transport = WS�       > User Agent transport = WS�       > User Agent transport = WS�       > User Agent transport = WS�       > User Agent transport = WS�       > User Agent transport = WS�       > User Agent transport = WS�       > User Agent transport = WS
Disconnected from Asterisk server
Executing last minute cleanups
Asterisk cleanly ending (0).


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

The expected output is to hear audio. But there is no audio.

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

I'm on Asterisk r372699M. I checked out fresh at the advice of Mamadou. I 
downloaded the latest patch as of the time of this post and applied it. The 
Asterisk server is on Ubuntu 11.04 64 bit, Chrome 23.0.1266.0 is on Ubuntu 
12.04 64 bit, and I have a Chrome Canary 23.0.1265.0 on both Mac 10.7 and 
Windows XP. 

Please provide any additional information below.

I tested with 2 Chrome's because the call won't even ring to XLite on Mac 10.7. 
 Before the update, I had one way audio. from Chrome to XLite only.

Again, I'm using all of the defaults from the repository and followed the docs 
exactly. Please let me know if you need more info.

JavaScript from Chrome:

State machine: c0000_Started_2_Outgoing_X_oINVITE tsk_utils.js:55
__on_state_change tsk_utils.js:55
Media Added tsk_utils.js:55
Call in progress... tsk_utils.js:55
SEND: INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKAWSBi1XkVwt90MvI0pZV29kR5xFCEfaz;rport
From: <sip:[email protected]>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:[email protected]>
Contact: "1060"<sip:[email protected];transport=ws>;+sip.ice
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55418 INVITE
Content-Type: application/sdp
Content-Length: 1282
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/
Organization: Doubango Telecom

v=0
o=- 126270660 1 IN IP4 127.0.0.1
s=webrtc (chrome 22.0.1189.0) - Doubango Telecom (sipML5 r000)
t=0 0
a=group:BUNDLE audio video
m=audio 32828 RTP/SAVPF 103 104 0 8 106 105 13 126
c=IN IP4 50.43.1.8
a=rtcp:32828 IN IP4 50.43.1.8
a=candidate:4086812097 1 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:4086812097 2 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:3172217137 1 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=candidate:3172217137 2 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=ice-ufrag:LLwIc5JWkKcHQGiA
a=ice-pwd:Ejj4ueILoSFLKIPhnU+blZ6R
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:C7nOKxVL0L/pK8ZYIbtH2mTE0GdlXUSmzdeT5iHp
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=ssrc:2823364993 cname:LvYPKr+O3jVL3995
a=ssrc:2823364993 mslabel:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS
a=ssrc:2823364993 label:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS00
 tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport;received=50.43.1.8;branch=z9hG4bKAWSBi1XkVwt90MvI0pZV
29kR5xFCEfaz
From: <sip:[email protected]>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:[email protected]>;tag=as2b85ef53
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55418 INVITE
Content-Length: 0
Server: Asterisk PBX SVN-trunk-r372699M
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest 
realm="doubango.org",nonce="4e408110",stale=FALSE,algorithm=MD5

 tsk_utils.js:55
SEND: ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKAWSBi1XkVwt90MvI0pZV29kR5xFCEfaz;rport
From: <sip:[email protected]>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:[email protected]>;tag=as2b85ef53
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55418 ACK
Content-Length: 0
Max-Forwards: 70

 tsk_utils.js:55
State machine: x0000_Any_2_Any_X_i401_407_INVITE tsk_utils.js:55
SEND: INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKgwSmiD1S3ffO84uAIxAatvAFk9cAdJFt;rport
From: <sip:[email protected]>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:[email protected]>
Contact: "1060"<sip:[email protected];transport=ws>;+sip.ice
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55419 INVITE
Content-Type: application/sdp
Content-Length: 1282
Max-Forwards: 70
Authorization: Digest 
username="1060",realm="doubango.org",nonce="4e408110",uri="sip:[email protected]",re
sponse="da920b16ba2d972a4b955f64d6be5b72",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/
Organization: Doubango Telecom

v=0
o=- 126270660 1 IN IP4 127.0.0.1
s=webrtc (chrome 22.0.1189.0) - Doubango Telecom (sipML5 r000)
t=0 0
a=group:BUNDLE audio video
m=audio 32828 RTP/SAVPF 103 104 0 8 106 105 13 126
c=IN IP4 50.43.1.8
a=rtcp:32828 IN IP4 50.43.1.8
a=candidate:4086812097 1 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:4086812097 2 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:3172217137 1 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=candidate:3172217137 2 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=ice-ufrag:LLwIc5JWkKcHQGiA
a=ice-pwd:Ejj4ueILoSFLKIPhnU+blZ6R
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:C7nOKxVL0L/pK8ZYIbtH2mTE0GdlXUSmzdeT5iHp
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=ssrc:2823364993 cname:LvYPKr+O3jVL3995
a=ssrc:2823364993 mslabel:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS
a=ssrc:2823364993 label:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS00
 tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 100 Trying
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport;received=50.43.1.8;branch=z9hG4bKgwSmiD1S3ffO84uAIxAa
tvAFk9cAdJFt
From: <sip:[email protected]>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060;transport=WS>
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55419 INVITE
Content-Length: 0
Server: Asterisk PBX SVN-trunk-r372699M
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer

 tsk_utils.js:55
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js:55
Trying tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport;received=50.43.1.8;branch=z9hG4bKgwSmiD1S3ffO84uAIxAa
tvAFk9cAdJFt
From: <sip:[email protected]>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:[email protected]>;tag=as7e61952f
Contact: <sip:[email protected]:5060;transport=WS>
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55419 INVITE
Content-Length: 0
Server: Asterisk PBX SVN-trunk-r372699M
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer

 tsk_utils.js:55
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js:55
Ringing tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport;received=50.43.1.8;branch=z9hG4bKgwSmiD1S3ffO84uAIxAa
tvAFk9cAdJFt
From: <sip:[email protected]>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:[email protected]>;tag=as7e61952f
Contact: <sip:[email protected]:5060;transport=WS>
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55419 INVITE
Content-Type: application/sdp
Content-Length: 650
Server: Asterisk PBX SVN-trunk-r372699M
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer

v=0
o=root 426452035 426452035 IN IP4 10.168.1.6
s=Asterisk PBX SVN-trunk-r372699M
c=IN IP4 10.168.1.6
t=0 0
m=audio 10650 RTP/SAVPF 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=silenceSupp:off - - - -
a=ptime:20
a=ice-ufrag:27b11bac1e57e2242d92cb8b2e65b422
a=ice-pwd:21b0dc964ac3faf11d5e9ee05b4feff0
a=candidate:Haa89741 1 udp 2130706431 10.168.1.6 10650 typ host generation 0 
svn 10
a=candidate:Haa89741 2 udp 2130706430 10.168.1.6 10651 typ host generation 0 
svn 10
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:gFgZbuFJl3XlTE8xviOwzMWnGGy4qaNMMQYZ45GR
 tsk_utils.js:55
State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE tsk_utils.js:55
__on_open tsk_utils.js:55
__on_state_change tsk_utils.js:55
SEND: ACK sip:[email protected]:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKB7KDOPzXz6iPArF1flC7;rport
From: <sip:[email protected]>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:[email protected]>;tag=as7e61952f
Contact: "1060"<sip:[email protected];transport=ws>;+sip.ice
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55419 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest 
username="1060",realm="doubango.org",nonce="4e408110",uri="sip:[email protected]:50
60;transport=WS",response="d4455e5ab99c3f7a961d67c5f6d464ea",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/
Organization: Doubango Telecom

 tsk_utils.js:55
OK tsk_utils.js:55
In Call tsk_utils.js:55
SEND: INVITE sip:[email protected]:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKJnaoITOlWLPF5lCKomEKz68BJemWAsCl;rport
From: <sip:[email protected]>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:[email protected]>;tag=as7e61952f
Contact: "1060"<sip:[email protected];transport=ws>;+sip.ice
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55420 INVITE
Content-Type: application/sdp
Content-Length: 3104
Max-Forwards: 70
Authorization: Digest 
username="1060",realm="doubango.org",nonce="4e408110",uri="sip:[email protected]:50
60;transport=WS",response="6d618b54128182514ba7259239d7416a",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/
Organization: Doubango Telecom

v=0
o=- 126270660 2 IN IP4 127.0.0.1
s=webrtc (chrome 22.0.1189.0) - Doubango Telecom (sipML5 r000)
t=0 0
a=group:BUNDLE audio video
m=audio 32828 RTP/SAVPF 103 104 0 8 106 105 13 126
c=IN IP4 50.43.1.8
a=rtcp:32828 IN IP4 50.43.1.8
a=candidate:4086812097 1 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:4086812097 2 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:3172217137 1 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=candidate:3172217137 2 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=candidate:4086812097 1 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:3172217137 1 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=candidate:4086812097 2 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:3172217137 2 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=ice-ufrag:LLwIc5JWkKcHQGiA
a=ice-pwd:Ejj4ueILoSFLKIPhnU+blZ6R
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:C7nOKxVL0L/pK8ZYIbtH2mTE0GdlXUSmzdeT5iHp
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=ssrc:2823364993 cname:LvYPKr+O3jVL3995
a=ssrc:2823364993 mslabel:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS
a=ssrc:2823364993 label:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS00
m=video 32828 RTP/SAVPF 100 101 102
c=IN IP4 50.43.1.8
a=rtcp:32828 IN IP4 50.43.1.8
a=candidate:4086812097 1 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:4086812097 2 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:3172217137 1 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=candidate:3172217137 2 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=ice-ufrag:LLwIc5JWkKcHQGiA
a=ice-pwd:Ejj4ueILoSFLKIPhnU+blZ6R
a=sendrecv
a=mid:video
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:C7nOKxVL0L/pK8ZYIbtH2mTE0GdlXUSmzdeT5iHp
a=rtpmap:100 VP8/90000
a=rtpmap:101 red/90000
a=rtpmap:102 ulpfec/90000
a=ssrc:3925540917 cname:LvYPKr+O3jVL3995
a=ssrc:3925540917 mslabel:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS
a=ssrc:3925540917 label:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS10
 tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 488 Not acceptable here
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport;received=50.43.1.8;branch=z9hG4bKJnaoITOlWLPF5lCKomEK
z68BJemWAsCl
From: <sip:[email protected]>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:[email protected]>;tag=as7e61952f
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55420 INVITE
Content-Length: 0
Server: Asterisk PBX SVN-trunk-r372699M
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16

 tsk_utils.js:55
SEND: ACK sip:[email protected]:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKJnaoITOlWLPF5lCKomEKz68BJemWAsCl;rport
From: <sip:[email protected]>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:[email protected]>;tag=as7e61952f
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55420 ACK
Content-Length: 0
Max-Forwards: 70

 tsk_utils.js:55
SEND: INVITE sip:[email protected]:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKy7uaxTPmuiFOT1S4hMFg3Nt5b6nLkxXp;rport
From: <sip:[email protected]>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:[email protected]>;tag=as7e61952f
Contact: "1060"<sip:[email protected];transport=ws>;+sip.ice
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55421 INVITE
Content-Type: application/sdp
Content-Length: 3104
Max-Forwards: 70
Authorization: Digest 
username="1060",realm="doubango.org",nonce="4e408110",uri="sip:[email protected]:50
60;transport=WS",response="6d618b54128182514ba7259239d7416a",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/
Organization: Doubango Telecom

v=0
o=- 126270660 3 IN IP4 127.0.0.1
s=webrtc (chrome 22.0.1189.0) - Doubango Telecom (sipML5 r000)
t=0 0
a=group:BUNDLE audio video
m=audio 32828 RTP/SAVPF 103 104 0 8 106 105 13 126
c=IN IP4 50.43.1.8
a=rtcp:32828 IN IP4 50.43.1.8
a=candidate:4086812097 1 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:4086812097 2 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:3172217137 1 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=candidate:3172217137 2 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=candidate:4086812097 1 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:3172217137 1 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=candidate:4086812097 2 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:3172217137 2 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=ice-ufrag:LLwIc5JWkKcHQGiA
a=ice-pwd:Ejj4ueILoSFLKIPhnU+blZ6R
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:C7nOKxVL0L/pK8ZYIbtH2mTE0GdlXUSmzdeT5iHp
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=ssrc:2823364993 cname:LvYPKr+O3jVL3995
a=ssrc:2823364993 mslabel:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS
a=ssrc:2823364993 label:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS00
m=video 32828 RTP/SAVPF 100 101 102
c=IN IP4 50.43.1.8
a=rtcp:32828 IN IP4 50.43.1.8
a=candidate:4086812097 1 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:4086812097 2 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:3172217137 1 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=candidate:3172217137 2 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=ice-ufrag:LLwIc5JWkKcHQGiA
a=ice-pwd:Ejj4ueILoSFLKIPhnU+blZ6R
a=sendrecv
a=mid:video
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:C7nOKxVL0L/pK8ZYIbtH2mTE0GdlXUSmzdeT5iHp
a=rtpmap:100 VP8/90000
a=rtpmap:101 red/90000
a=rtpmap:102 ulpfec/90000
a=ssrc:3925540917 cname:LvYPKr+O3jVL3995
a=ssrc:3925540917 mslabel:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS
a=ssrc:3925540917 label:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS10
 tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 488 Not acceptable here
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport;received=50.43.1.8;branch=z9hG4bKy7uaxTPmuiFOT1S4hMFg
3Nt5b6nLkxXp
From: <sip:[email protected]>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:[email protected]>;tag=as7e61952f
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55421 INVITE
Content-Length: 0
Server: Asterisk PBX SVN-trunk-r372699M
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16

 tsk_utils.js:55
SEND: ACK sip:[email protected]:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKy7uaxTPmuiFOT1S4hMFg3Nt5b6nLkxXp;rport
From: <sip:[email protected]>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:[email protected]>;tag=as7e61952f
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55421 ACK
Content-Length: 0
Max-Forwards: 70

 tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=BYE sip:[email protected];transport=ws SIP/2.0
Via: SIP/2.0/WS 10.168.1.6:5060;branch=z9hG4bK31bf63fb
From: <sip:[email protected]>;tag=as7e61952f
To: <sip:[email protected]>;tag=ACODOuXTJzNKEwrv9ip8
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 102 BYE
Content-Length: 0
Max-Forwards: 70
User-Agent: Asterisk PBX SVN-trunk-r372699M
Proxy-Authorization: Digest 
username="1060",realm="doubango.org",nonce="4e408110",uri="sip:james.org",respon
se="9f3bb2fa621b20d320034f888e4580c6",algorithm=MD5
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16

 tsk_utils.js:55
State machine: x0000_Any_2_Terminated_X_iBYE tsk_utils.js:55
=== INVITE Dialog terminated === tsk_utils.js:55
__on_state_change tsk_utils.js:55
PeerConnection::stop() tsk_utils.js:55
SEND: SIP/2.0 200 OK
Via: SIP/2.0/WS 10.168.1.6:5060;branch=z9hG4bK31bf63fb
From: <sip:[email protected]>;tag=as7e61952f
To: <sip:[email protected]>;tag=ACODOuXTJzNKEwrv9ip8
Contact: <sip:[email protected];transport=ws>
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 102 BYE
Content-Length: 0

 tsk_utils.js:55
Call terminated tsk_utils.js:55
2Media Removed tsk_utils.js:55
State machine: tsip_dialog_register_Connected_2_InProgress_X_oRegister 
tsk_utils.js:55
SEND: REGISTER sip:james.org SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKa5GXC5sZp7be9ufmtyK1DVCnbJtakeDV;rport
From: <sip:[email protected]>;tag=9ZR1Q4e8UHtJg5X1NcfX
To: <sip:[email protected]>
Contact: 
"1060"<sip:[email protected];transport=ws>;expires=200;+g.oma.sip-im;+au
dio;language="en,fr"
Call-ID: ff548533-c2de-a693-bcd2-b99766c615c5
CSeq: 52213 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest 
username="1060",realm="doubango.org",nonce="55a3ba48",uri="sip:james.org",respon
se="45f027d273101c3f3d0e8e775d2020ce",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/
Organization: Doubango Telecom

 tsk_utils.js:55
REGISTER request successfully sent tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport;received=50.43.1.8;branch=z9hG4bKa5GXC5sZp7be9ufmtyK1
DVCnbJtakeDV
From: <sip:[email protected]>;tag=9ZR1Q4e8UHtJg5X1NcfX
To: <sip:[email protected]>;tag=as029056b1
Call-ID: ff548533-c2de-a693-bcd2-b99766c615c5
CSeq: 52213 REGISTER
Content-Length: 0
Server: Asterisk PBX SVN-trunk-r372699M
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest 
realm="doubango.org",nonce="129f1225",stale=FALSE,algorithm=MD5

 tsk_utils.js:55
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494 
tsk_utils.js:55
SEND: REGISTER sip:james.org SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKtciI7O8LAEn5LUX97Nut4Z5SG3SkhuDy;rport
From: <sip:[email protected]>;tag=9ZR1Q4e8UHtJg5X1NcfX
To: <sip:[email protected]>
Contact: 
"1060"<sip:[email protected];transport=ws>;expires=200;+g.oma.sip-im;+au
dio;language="en,fr"
Call-ID: ff548533-c2de-a693-bcd2-b99766c615c5
CSeq: 52214 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest 
username="1060",realm="doubango.org",nonce="129f1225",uri="sip:james.org",respon
se="13c2126a4e68e9282761a9b171edc1bc",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/
Organization: Doubango Telecom

 tsk_utils.js:55
REGISTER request successfully sent tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport;received=50.43.1.8;branch=z9hG4bKtciI7O8LAEn5LUX97Nut
4Z5SG3SkhuDy
From: <sip:[email protected]>;tag=9ZR1Q4e8UHtJg5X1NcfX
To: <sip:[email protected]>;tag=as029056b1
Contact: <sip:[email protected];transport=ws>;expires=200
Call-ID: ff548533-c2de-a693-bcd2-b99766c615c5
CSeq: 52214 REGISTER
Expires: 200
Content-Length: 0
Server: Asterisk PBX SVN-trunk-r372699M
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Date: Fri, 14 Sep 2012 0:40:53 GMT

 tsk_utils.js:55
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=NOTIFY sip:[email protected];transport=ws SIP/2.0
Via: SIP/2.0/WS 10.168.1.6:5060;branch=z9hG4bK7b067980
From: "asterisk"<sip:[email protected]>;tag=as0660ad45
To: <sip:[email protected];transport=ws>
Contact: <sip:[email protected]:5060;transport=WS>
Call-ID: [email protected]:5060
CSeq: 102 NOTIFY
Content-Type: application/simple-message-summary
Content-Length: 106
Max-Forwards: 70
User-Agent: Asterisk PBX SVN-trunk-r372699M
Event: message-summary

Messages-Waiting: no
Message-Account: sip:[email protected];transport=WS
Voice-Message: 0/0 (0/0)
 tsk_utils.js:55
SEND: SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/WS 10.168.1.6:5060;branch=z9hG4bK7b067980
From: "asterisk"<sip:[email protected]>;tag=as0660ad45
To: <sip:[email protected];transport=ws>
Call-ID: [email protected]:5060
CSeq: 102 NOTIFY
Content-Length:

Original issue reported on code.google.com by [email protected] on 14 Sep 2012 at 12:51

Add support for GWT

It is not issue, just a feature request.

If you add a wrapper for GWT, it make more websites avaible to use your sip 
library.

Original issue reported on code.google.com by [email protected] on 28 Sep 2012 at 1:29

group chat

What steps will reproduce the problem?
1.
2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 24 Apr 2012 at 11:08

tsip_message_parser do not parse message with non english content


To reproduce: try to send MESSAGE with non-english content, parser do not see 
message body in that case.

Is tsip_parser_message.js whole generated file by ragel or it has some custom 
content?

Original issue reported on code.google.com by [email protected] on 29 May 2012 at 1:45

DTMF using SIP INFO

What steps will reproduce the problem?
1.
2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 5 Apr 2012 at 1:44

onclose event not raised when websocket is closed

According to the standard, a browser must send 0xff00 to signal to the remote 
peer that it wants to close the socket. Chrome don't send this chucnk (buggy).
This is why we get 'Disconnected' message from sipML5 when the pending (ghost) 
connection is really closed by the server (e.g. activity) while the current is 
active.
Workaround: check if the closed event is from the active stack

Original issue reported on code.google.com by [email protected] on 29 May 2012 at 5:26

SIPML5 + Asterisk - No audio after 30 seconds

More info at: 
https://groups.google.com/group/doubango/browse_thread/thread/be712f8908b80f55

Original issue reported on code.google.com by [email protected] on 21 Sep 2012 at 6:29

call tsip_dialog_set_curr_action() in tsip_dialog_fsm_act()

What steps will reproduce the problem?
1.
2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 8 Apr 2012 at 9:23

Failed to parse (remote) sdp message


What steps will reproduce the problem?
1. Start calling somebody
2. Error message shown at receiver console: failed to parse remote or jut sdp 
message

What is the expected output? What do you see instead?
Video should be transfered.
I see the following in console:
Failed to parse sdp message: v=0
o=- 27545299 1 IN IP4 127.0.0.1
s=webrtc (chrome 22.0.1189.0) - Doubango Telecom (sipML5 r000)
t=0 0
a=group:BUNDLE audio video
m=audio 35551 RTP/SAVPF 103 104 0 8 106 105 13 126
c=IN IP4 217.92.224.18
a=rtcp:35551 IN IP4 217.92.224.18
a=candidate:3581128468 1 udp 2130714367 172.23.0.86 50481 typ host generation 0
a=candidate:3581128468 2 udp 2130714367 172.23.0.86 50481 typ host generation 0
a=candidate:2083266753 1 udp 1912610559 217.92.224.18 35551 typ srflx 
generation 0
a=candidate:2083266753 2 udp 1912610559 217.92.224.18 35551 typ srflx 
generation 0
a=candidate:2616218596 1 tcp 1694506751 172.23.0.86 54750 typ host generation 0
a=candidate:2616218596 2 tcp 1694506751 172.23.0.86 54750 typ host generation 0
a=ice-ufrag:KVSuUjn2k7Frf/XU
a=ice-pwd:7coqlgdFfMDVurLSziT2YHop
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:GoD�큈Nꌖ���׊Ⱦ,̌ꒆ�聰�ɿⳞD͍ϔމ刏Jﰬ8얎Ot嘂�ˉ�
�σY檾:ʍIsބ͍ϔމ剟)촪V牏IⳞD͍ϔމ偟)촾V牏IⳞD͍ϔމ刏OﺱV��
�Iﴵ�⋋ ҘσY췐H鉏IⳞD͍ϔމ刌Y췐At嘂�ˉҘσY˜ӜϑЗڔڏڗ˖��
�IⳞD͚̊勆NJ茊Yܗޔڃˈ,Ԋ݌钵�䖪�ⳞD͚̊勆NJ茊YҊӘݜӃ⪋)璌    �
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What version of the product are you using? On what operating system?
sipML5-v1.0.89.0 on OSX 10.6.8 with Chrome 22

Original issue reported on code.google.com by [email protected] on 28 Sep 2012 at 1:00

send_BYE() if set_ro() fails

What steps will reproduce the problem?
1.
2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 9 Apr 2012 at 7:52

Uncaught ReferenceError: tsip_action_type_e is not defined (tsip_dialog_invite.js:30)

What steps will reproduce the problem?
1.
2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 31 Mar 2012 at 10:16

iSAC

What steps will reproduce the problem?
1.
2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 30 Mar 2012 at 11:57

chrome to Asterisk call: SYNTAX_ERR: DOM Exception 12 upon 200 OK from Asterisk

Hello,

I followed the instructions from the following page:

http://code.google.com/p/sipml5/wiki/Asterisk

and I am able to establish a call from Google Chrome to an Asterisk server onto 
the same LAN, but the audio transmission in both directions does not seem to 
work at all.

The SYNTAX_ERR: DOM Exception 12 happens upon receiving the 200 OK response and 
I think it happens in native code called within this.o_pc.setRemoteDescription, 
which in turn is called from tmedia_session_jsep.prototype.__set_ro, so there's 
probably something wrong with the SDP received from Asterisk.

What steps will reproduce the problem?

1. Open ML5 page call.htm
2. Set the SIP account parameters and WebSocket server (I am using the 
repro-based one downloadable from your site, not Asterisk itself) 
3. Place a call

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

I expect to be able to hear/send audio but instead it doesn't work.

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

Chrome 24.0.1297.0 dev-m on Win 7 Professional SP1 x64

Please provide any additional information below.

This is the Chrome-side log:

 tsk_utils.js:55
State machine: x0000_Any_2_Any_X_i401_407_INVITE tsk_utils.js:55
SEND: INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKkDFyAEJ8yxyFs8XKzEzZPmaXbTbrGpTT;rport
From: <sip:[email protected]>;tag=fTAsAMFb3z3fgPP4CIkJ
To: <sip:[email protected]>
Contact: "fabry"<sip:[email protected];transport=ws>;+sip.ice
Call-ID: 87d06336-ca33-f248-d7d6-a2ec7b07ea4f
CSeq: 31932 INVITE
Content-Type: application/sdp
Content-Length: 1146
Max-Forwards: 70
Authorization: Digest 
username="8600",realm="asterisk",nonce="00112101",uri="sip:[email protected].
12",response="04fc599e2a496a1b71f881c1f67527b2",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/
Organization: Doubango Telecom

v=0
o=- 3617092930 1 IN IP4 127.0.0.1
s=webrtc (chrome 22.0.1189.0) - Doubango Telecom (sipML5 r000)
t=0 0
a=group:BUNDLE audio video
m=audio 60443 RTP/SAVP 103 104 0 8 106 105 13 126
c=IN IP4 10.77.35.15
a=rtcp:60443 IN IP4 10.77.35.15
a=candidate:1697405919 1 udp 2113937151 10.77.35.15 60443 typ host generation 0
a=candidate:1697405919 2 udp 2113937151 10.77.35.15 60443 typ host generation 0
a=candidate:732931887 1 tcp 1509957375 10.77.35.15 53043 typ host generation 0
a=candidate:732931887 2 tcp 1509957375 10.77.35.15 53043 typ host generation 0
a=ice-ufrag:lfzNvrLdcOa18IiY
a=ice-pwd:dhwmGn1IroVQKLxYBh8aR64N
a=ice-options:google-ice
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:Q4unJxQVjw+oh5PvWvk3UophXn2LQAPHL2F7A8MK
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=ssrc:3374699521 cname:k+3tRzqrO9GHIppN
a=ssrc:3374699521 mslabel:IDB9Uz1ISXFcDZhBc8ZktMV31LWwrQlaNpN4
a=ssrc:3374699521 label:IDB9Uz1ISXFcDZhBc8ZktMV31LWwrQlaNpN400
 tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 100 Trying
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport=53036;received=10.77.35.15;branch=z9hG4bKkDFyAEJ8yxyF
s8XKzEzZPmaXbTbrGpTT
From: <sip:[email protected]>;tag=fTAsAMFb3z3fgPP4CIkJ
To: <sip:[email protected]>
Call-ID: 87d06336-ca33-f248-d7d6-a2ec7b07ea4f
CSeq: 31932 INVITE
Content-Length: 0

 tsk_utils.js:55
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js:55
Trying tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP 
10.77.35.15:53036;rport=53036;received=10.77.35.15;branch=z9hG4bKkDFyAEJ8yxyFs8X
KzEzZPmaXbTbrGpTT
From: <sip:[email protected]>;tag=fTAsAMFb3z3fgPP4CIkJ
To: <sip:[email protected]>;tag=as028f7b42
Contact: <sip:[email protected]:5060>
Call-ID: 87d06336-ca33-f248-d7d6-a2ec7b07ea4f
CSeq: 31932 INVITE
Content-Type: application/sdp
Content-Length: 576
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Server: Asterisk PBX SVN-trunk-r373330M
Supported: replaces,timer

v=0
o=root 1547035746 1547035746 IN IP4 10.77.36.12
s=Asterisk PBX SVN-trunk-r373330M
c=IN IP4 10.77.36.12
t=0 0
m=audio 12778 RTP/SAVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:0a35d5ab2e4d42df3fc1262a56cbe223
a=ice-pwd:2786b50c4df31cbc3705554b5b6481a8
a=candidate:Ha4d240c 1 udp 2130706431 10.77.36.12 12778 typ host
a=candidate:Ha4d240c 2 udp 2130706430 10.77.36.12 12779 typ host
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:AsXbqqQscS5j6Fq044NwY6b/0HumDyWpPOz/VwtR
 tsk_utils.js:55
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js:55
__on_open tsk_utils.js:55
__on_state_change tsk_utils.js:55
2Session Progress tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 
10.77.35.15:53036;rport=53036;received=10.77.35.15;branch=z9hG4bKkDFyAEJ8yxyFs8X
KzEzZPmaXbTbrGpTT
From: <sip:[email protected]>;tag=fTAsAMFb3z3fgPP4CIkJ
To: <sip:[email protected]>;tag=as028f7b42
Contact: <sip:[email protected]:5060>
Call-ID: 87d06336-ca33-f248-d7d6-a2ec7b07ea4f
CSeq: 31932 INVITE
Content-Length: 0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Server: Asterisk PBX SVN-trunk-r373330M
Supported: replaces,timer

 tsk_utils.js:55
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js:55
Ringing tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 200 OK
Via: SIP/2.0/TCP 
10.77.35.15:53036;rport=53036;received=10.77.35.15;branch=z9hG4bKkDFyAEJ8yxyFs8X
KzEzZPmaXbTbrGpTT
From: <sip:[email protected]>;tag=fTAsAMFb3z3fgPP4CIkJ
To: <sip:[email protected]>;tag=as028f7b42
Contact: <sip:[email protected]:5060>
Call-ID: 87d06336-ca33-f248-d7d6-a2ec7b07ea4f
CSeq: 31932 INVITE
Content-Type: application/sdp
Content-Length: 576
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Server: Asterisk PBX SVN-trunk-r373330M
Supported: replaces,timer

v=0
o=root 1547035746 1547035747 IN IP4 10.77.36.12
s=Asterisk PBX SVN-trunk-r373330M
c=IN IP4 10.77.36.12
t=0 0
m=audio 12778 RTP/SAVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:0a35d5ab2e4d42df3fc1262a56cbe223
a=ice-pwd:2786b50c4df31cbc3705554b5b6481a8
a=candidate:Ha4d240c 1 udp 2130706431 10.77.36.12 12778 typ host
a=candidate:Ha4d240c 2 udp 2130706430 10.77.36.12 12779 typ host
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:AsXbqqQscS5j6Fq044NwY6b/0HumDyWpPOz/VwtR
 tsk_utils.js:55
State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE tsk_utils.js:55
DOMException {message: "SYNTAX_ERR: DOM Exception 12", name: "SYNTAX_ERR", 
code: 12, stack: "Error: An invalid or illegal string was 
specified.…c/transactions/tsip_transac_ict.js?svn=10:454:32)"}
 tsk_utils.js:67
SEND: ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKcDlNGy6AXElWlUWib27B;rport
From: <sip:[email protected]>;tag=fTAsAMFb3z3fgPP4CIkJ
To: <sip:[email protected]>;tag=as028f7b42
Contact: "fabry"<sip:[email protected];transport=ws>;+sip.ice
Call-ID: 87d06336-ca33-f248-d7d6-a2ec7b07ea4f
CSeq: 31932 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest 
username="8600",realm="asterisk",nonce="00112101",uri="sip:[email protected].
12:5060",response="826f96616377356598a830f046abdc01",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/
Organization: Doubango Telecom

On the Asterisk side, there are no warnings or errors.
I noticed that the SDP returned by Asterisk has no a=rtcp line, could that be 
the problem?

Thanks in advance.
Best regards,
Fabrizio Ammollo

Original issue reported on code.google.com by [email protected] on 18 Oct 2012 at 9:42

Uncaught ReferenceError: tmedia_type_e is not defined

What steps will reproduce the problem?
1.
2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 29 Mar 2012 at 8:13

send_response(force_sdp) -> start media session if not done

What steps will reproduce the problem?
1.
2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 1 Apr 2012 at 1:01

Support direct SIP calls

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

What steps will reproduce the problem?
1. Enter number to call without logging in
2. Click Call
3.

What is the expected output? What do you see instead?
Expected: Direct SIP call without proxy
Actual: Call button disabled

What version of the product are you using? On what operating system?
Live demo on Chrome 21.0.1144.0 canary

Please provide any additional information below.

Original issue reported on code.google.com by [email protected] on 21 May 2012 at 3:38

Adds support for "webkitRTCPeerConnection" on chrome

https://groups.google.com/forum/#!topic/discuss-webrtc/0E7xnLHqMYA

Original issue reported on code.google.com by [email protected] on 19 Sep 2012 at 8:24

Rewrite BYE Request Uri

What steps will reproduce the problem?
1.
2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 31 Mar 2012 at 3:28

ArrayBufferView size is not a small enough positive integer.


Canary: 21.0.1137.1 Windows Vista
Version: r49 from svn

Call Stack:
Uncaught RangeError: ArrayBufferView size is not a small enough positive 
integer. tsk_buff.js:38
tsk_buff_ab2str tsk_buff.js:38
tsk_ragel_state_init_ai tsk_ragel.js:39
__tsip_transport_ws_onmessage tsip_transport.js:379

buff in tsk_buff_ab2str:
"SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKv6Ac1PwQppgZOcqxawts6aha5GrW5Ikk;rport
From:  <sip:[email protected]>;tag=ODbKBbENOA2LAoGxLabZ
To:  <sip:[email protected]>;tag=2b249c78777846509921a74dbdb6530e
Call-ID: 52159381-973f-3120-c659-2293baeab1cb
CSeq: 61812 REGISTER
WWW-Authenticate: Digest 
realm="officesip.local",nonce="26cb330709fa2b663a488b0977873241",qop="auth",algo
rithm=MD5,stale=false,opaque="00000006"
x-Error-Details: No auth header
Content-Length: 0

"

Original issue reported on code.google.com by [email protected] on 16 May 2012 at 2:58

Call from chrome to softphone through asterisk cause only noise

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

What steps will reproduce the problem?
1. Setup asterisk according to https://code.google.com/p/sipml5/wiki/Asterisk 
on wirtual machine
2. Register and call from sipml5 (my revision 109) to softphone
3. Answer to call in softphone


What is the expected output? What do you see instead?
Expected normall call, but there is only noise.

What version of the product are you using? On what operating system?
sipml5 rev 109, chrome 21.0.1180.89, asterisk os - debian 6, chrome and 
softphone os windows7-64

Please provide any additional information below.
As asked Mamadou chrome console from page open to call end is attached.
192.168.225.204 asterisk url, 192.168.225.183 chrome and softphone url.
Any other info will be provided if needed.

Original issue reported on code.google.com by [email protected] on 26 Sep 2012 at 12:17

Attachments:

chrome_console.txt

PRACK retransmission with different Call-ID

What steps will reproduce the problem?
1.
2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 30 Mar 2012 at 9:43

Hangup does not work if I dial

Cannot cancel dialing out from browser. Only calls that get in. And the server 
keeps ringing.

What steps will reproduce the problem?
1. Connect to server
2. Dial out from browser any number/extension
3. Try to HangUp before the other party answers, it keeps ringing.

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

It should hangup the call. The browser shows Call terminating, but the server 
keeps ringing. It sends CANCEL instead of DECLINE or BYE
because "this.o_session.b_server" is False and "this.e_state" is "Early" on if 
conditions in t_sip_dialog.js line 731 where is decided what action to take.

What version of the product are you using? On what operating system?
10. Windows 8. (tested on Chrome ws and also on Firefox udp)

Original issue reported on code.google.com by [email protected] on 22 Sep 2012 at 1:51

DNS NAPTR+SRV on the cloud

What steps will reproduce the problem?
1.
2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 29 Mar 2012 at 8:30

VNC Viewer

What steps will reproduce the problem?
1.
2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 29 Mar 2012 at 8:49

1024+ WS connections

What steps will reproduce the problem?
1.
2.
3.

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


Please use labels and text to provide additional information.

Original issue reported on code.google.com by [email protected] on 31 Mar 2012 at 10:16

Bad quality of video when implement "sipML5 solution architecture (2)"

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

What steps will reproduce the problem?
1.Following all steps of guides to implement "sipML5 solution architecture (2)" 
- link http://code.google.com/p/sipml5/wiki/Asterisk

2. Private Asterisk Server -  revision 372699 
3. patch asterisk_372699.patch
4. install Asterisk from source was patched
5. File "sip.conf" include "videosupport=yes", realm and domainsasrealm are 
local IP of Asterisk server ( 192.168.212.59 )
6. File "http.conf", "users.conf", "extensions.conf" are the same as examples.
7. In expert mode : no check Disable Video and Disable AVPF
              WebSocket Server URL: ws://192.168.212.59:8088/ws
              SIP outbound Proxy URL: udp://192.168.212.59:5060
8. register with Asterisk and make a video call between 2 sipml5 client in 
seperate machine ( same local network ).

9. make a video conference with 2 sipml5 client and 1 xlite client.



What is the expected output? 

- good quality of communication with voice and video.
- good quality of video call.
- show on attendees video of a conference in sipml5 client windows. 

What do you see instead?

- video call between 2 sipml5 client have bad quality. The voice is pretty good 
but the video is not clearly.( i capture the scene and attach a photo to this 
post ).
- I only get black window when I implement conferenc with 2 other clients.

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

- Source Sipml5 newest update.
- Firefox 15
- webrtc4all_windows_1_11_745.sfx
- Astersik revision 372699 and Centos 6

Please provide any additional information below.

- Javascript console log.
- Photo of scenes show bad quality of video.


Please help me to solve theses problem. 

Best regards,

ducdung0909

Original issue reported on code.google.com by [email protected] on 20 Sep 2012 at 4:23

Attachments:

[bad video - 2 client.png](https://storage.googleapis.com/google-code-attachments/sipml5/issue-41/comment-0/bad video - 2 client.png)
[bad video - conference.png](https://storage.googleapis.com/google-code-attachments/sipml5/issue-41/comment-0/bad video - conference.png)
error-xlite-try-connect-conference.txt
log-Asterisk.txt
[log-call between 2 client.txt](https://storage.googleapis.com/google-code-attachments/sipml5/issue-41/comment-0/log-call between 2 client.txt)

Ericsson browser: "Failed to parse remote sdp"

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

What steps will reproduce the problem?
1. Start calling somebody.
2. Error message shown at receiver console: "Failed to parse remote sdp".
message". when received 200 OK

What is the expected output? What do you see instead?
I want see "In call" status, but got "Call terminated" status. 

What version of the product are you using? On what operating system?
sipml5(r121), Ericsson browser (iOS6, iPad3), Asterisk 1.11(r373330)

Please provide any additional information below.
SDP which cannot be parsed by sipml5:

v=0
o=root 836352876 836352876 IN IP4 11.111.11.11
s=Asterisk PBX
c=IN IP4 11.111.11.11
t=0 0
m=audio 24722 RTP/AVPF 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=ice-ufrag:41b1e21f7e29d20258bb5c382bfa844c
a=ice-pwd:31332d4a7bdf2e7922b75b8e3446614e
a=ice-options:google-ice
a=candidate:H5b79513e 1 udp 2130706431 11.111.11.11 24722 typ host
generation 0 svn 16
a=candidate:H5b79513e 2 udp 2130706430 11.111.11.11 24723 typ host
generation 0 svn 16
a=sendrecv
m=video 0 RTP/AVPF 103

In desktop browsers (Chrome, IE+webrtc4all) works good!

Original issue reported on code.google.com by [email protected] on 31 Oct 2012 at 12:00

smallsquirrel / sipml5 Goto Github PK

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