Comments (37)
Some additional info can be found in
https://groups.google.com/forum/?hl=en&fromgroups=#!topic/doubango/1Kd74RzdlmY
Original comment by [email protected]
on 26 Sep 2012 at 12:37
from sipml5.
[deleted comment]
from sipml5.
@ln.nima.nl
"22.0.1229.79" is not STABLE chrome but beta. STABLE is at 21....
Original comment by [email protected]
on 26 Sep 2012 at 1:59
from sipml5.
I have the similar problem when I connect from chrome on Ubuntu 12.04 -64
bit(same pc I am running Asterisk on ).
incoming calls are ok , but calls from chrome are connected successfully but
audio does not work. I see this error in Asterisk console:
[Sep 26 15:18:09] WARNING[6679][C-00000018]: res_srtp.c:395 ast_srtp_unprotect:
SRTP unprotect failed with: authentication failure 110
[Sep 26 15:18:11] WARNING[6679][C-00000018]: res_srtp.c:395 ast_srtp_unprotect:
SRTP unprotect failed with: authentication failure 110
wireshark shows packets are being transmitted between chrome and asterisk and
packets are coming from softphone but no packet is being sent from asterisk to
the 2nd client ( xlite or any other sip , in my case a pstn gateway ).
I am hearing either noise or silence in chrome.
but running chrome and xlite on another pc works fine for me and I have two way
audio for incoming and outgoing.
for the record:
server Ip : 192.168.2.105
chrome version : stable chrome , 22.0.1229.79
I am calling extension 101 which refers to xlite account.
chrome console log is attached .
thanks
Original comment by [email protected]
on 26 Sep 2012 at 2:00
Attachments:
from sipml5.
comment #3 no longer make sense because "ln.nima.nl" removed it's comment (#2).
Original comment by [email protected]
on 26 Sep 2012 at 2:01
from sipml5.
dear bossiel, it is the latest update (today morning ).
it is stable, am I wrong ?
chrome updated automatically on all my platforms (win7 , ubuntu , win xp )
anyways I had the similar problem with version 21 on previous days.
Original comment by [email protected]
on 26 Sep 2012 at 2:02
from sipml5.
sorry , I decided to edit the text to avoid misunderstanding.
my mistake :)
Original comment by [email protected]
on 26 Sep 2012 at 2:06
from sipml5.
Previous days are previous days
Original comment by [email protected]
on 26 Sep 2012 at 2:06
from sipml5.
@teihrib
I'm not surprised that there is no audio because there is a big javacript
parsing error ("SYNTAX_ERR: DOM Exception 12") in your logs.
I'm using same chrome version (21.0.1180.89) on windows Vista and OSX but I
have no js errors.
Do you have a Vista or OS X for tests?
What if chrome is the called party?
Original comment by [email protected]
on 26 Sep 2012 at 2:15
- Changed state: Accepted
from sipml5.
Tested with "22.0.1229.79 m" and it's working as expected
Original comment by [email protected]
on 26 Sep 2012 at 2:23
from sipml5.
There is an Asterisk server running (will be shutdown on Friday 28, 2012) on
sipml5.org for tests.
4 accounts: 1060, 1061, 1062, 1063. Even numbers for chrome.
WebSocket connection: ws://sipml5.org:8088/ws
UDP connection: udp://sipml5.org:3060
Original comment by [email protected]
on 26 Sep 2012 at 2:28
from sipml5.
Until now I was only thinking that it is a bug in Asterisk but since you
implied that it has something to do with browser , I feel that something is
wrong with chrome in 64 bit systems , because "teihrib" is using 64 bit and I
only have this problem on 64 bit system and I remember another user on doubango
forum (James) who has similar problem and his system is 64 bit as far as I
remember.
Original comment by [email protected]
on 26 Sep 2012 at 2:28
from sipml5.
I tested with the online server.
The problem is the same and I still get DOM exception and noise when I call
from chrome on Ubuntu 64 bit to any other client (softphone or chrome on
win7-32).
From chrome on win7-32 to chrome on ubuntu-64 works fine and I have two way
audio.
I will be very thankful if you can try it yourself on some 64 bit system to see
if it works for you.
Thanks for the support
Original comment by [email protected]
on 26 Sep 2012 at 6:56
from sipml5.
The exception is caused by the SRTP crypto type.
Your chrome version try to use "AES_CM_128_HMAC_SHA1_32" while only
"AES_CM_128_HMAC_SHA1_80" is supported by chrome.
Open javascript console, type navigator.userAgent and tell us what is the
result.
Original comment by [email protected]
on 26 Sep 2012 at 8:40
from sipml5.
oops... I meant "...is supported by webrtc"
Original comment by [email protected]
on 26 Sep 2012 at 8:42
from sipml5.
[deleted comment]
from sipml5.
I'm here now.
navigator.userAgent
"Mozilla/5.0 (Windows NT 6.1; WOW64) AppleWebKit/537.4 (KHTML, like Gecko)
Chrome/22.0.1229.79 Safari/537.4" I'll try sipml5.org asterisk now. So I get
these results. When I call from softphone to chrome everything is ok. When I
call from one chrome to another I get the same noise. When I call from chrome
to softphone, softphone doesn't rrecieve this call, inspite of in chrome the
call is answered (but there is only silence) I attached my chrome console logs
for these calls.
Original comment by [email protected]
on 27 Sep 2012 at 5:33
Attachments:
- chrome_console_chrome2chrome_through_sipml5.org.txt
- chrome_console_chrome2softphone_through_sipml5.org.txt
- chrome_console_softphone2chrome_through_sipml5.org.txt
from sipml5.
Another info. Folowing Mamadou advice I tryed to use public sip services
(ekiga.net in my case). I registered two accounts. Then I called from one
chrome to another and everything is fine (there is no DOM exception). Then I
called from softphone to chrome and get dom exception. I think it's because of
INVITE message from softphone, that comes with RTP/AVP, not RTP/SAVPF.
Original comment by [email protected]
on 27 Sep 2012 at 6:11
from sipml5.
Here is chrome console (caller side) for ekiga.net. Hope this helps
Original comment by [email protected]
on 27 Sep 2012 at 6:24
Attachments:
from sipml5.
@bossiel : I see : "Mozilla/5.0 (X11; Linux x86_64) AppleWebKit/537.4 (KHTML,
like Gecko) Chrome/22.0.1229.79 Safari/537.4" . Do you think there is any
solution for crypto problem from client and server side ? or it should be
resolved by google chrome WebRTC team ?
@teihrib : When you try public sip accounts on sipml5.org , webrtc2sip
websocket server is used and it does not have anything to do with media, it
just proxies the sip messages and it's a totally different scenario compared to
Asterisk. so chrome to chrome must be fine , and softphone to chrome does not
work , as you said , because of profile mismatch .
Original comment by [email protected]
on 27 Sep 2012 at 8:30
from sipml5.
I forgot to say that you can use doubango softphone clients and make direct
calls from softphone to chrome and vice versa .
have a look at here :
http://code.google.com/p/sipml5/wiki/Calling_SIP_clients
Original comment by [email protected]
on 27 Sep 2012 at 8:33
from sipml5.
@ln.nima.nl it's just strange for me, because it looks like chrome is ok in
this case? or I missed something.
Original comment by [email protected]
on 27 Sep 2012 at 10:17
from sipml5.
I verified the outgoing SDP and it's really strange .You are right, when I use
chrome on win7-32 , there is only one crypto line (AES_CM_128_HMAC_SHA1_80) but
using chrome un ubuntu-64 , SDP has two crypto lines , first is 32 and second
is 80 , and apparently Asterisk only handles the first crypto line which is 32
and that's the source of this issue. (although both pcs load the same webpage
and same problem when I tried with online server)
I got the Idea from this post
:https://groups.google.com/d/msg/discuss-webrtc/YhI5fUpx1jc/DOJv90wSZaIJ
that removing the first line is gonna solve the problem .
Can you point me to some direction on how to do it ?
which Java script source file should I modify? or is it possible to teach
Asterisk to handle both crypto lines and only accept "AES_CM_128_HMAC_SHA1_80" ?
Thank you
Original comment by [email protected]
on 27 Sep 2012 at 11:49
from sipml5.
I modified tmedia_session_jsep.js function decorate_lo to remove crypto string
with _32. Then the call established and there is no domexception anymore, but I
hear only silence. And in asterisk log there again
[Sep 28 01:42:31] WARNING[22993][C-00000015] res_srtp.c: SRTP unprotect failed
with: authentication failure 10
[Sep 28 01:42:33] WARNING[22993][C-00000015] res_srtp.c: SRTP unprotect failed
with: authentication failure 110
Also there is new message
[Sep 28 01:35:46] WARNING[22925][C-00000012] app.c: No audio available on
SIP/1060-00000024??
Any thoughts?
Original comment by [email protected]
on 28 Sep 2012 at 6:15
from sipml5.
OMG, it's working! I don't know what was wrong (possible browser cache), but
now it's working! Here is modified code (just for test, not for regular use).
When run this code there is no more message app.c: No audio available on
SIP/1060, but "SRTP unprotect failed with:" still appears.
Original comment by [email protected]
on 28 Sep 2012 at 6:28
Attachments:
from sipml5.
Thank you very much Anton. You did a great job .
My problem is solved now.
Now I can call out from my chrome on ubuntu-64 and everything works perfect and
as expected . I don't see any errors not in chrome , not in Asterisk . (Unlike
you I don't see: SRTP unprotect failed). Incoming calls are also fine.
And I need to add that it did not affect other clients that were working
correctly before(my chrome on win7-32).
Try to restart everything (web and asterisk server and your pcs) and also clear
all the cache on browser, and try again and see if it works for you.
I don't know if Mamadou will accept this hacked code and add it to the source
officially and change the status of this issue to FIXED . but it will be great
if he accepts this .
looking forward to that .
Regards
Original comment by [email protected]
on 28 Sep 2012 at 9:26
from sipml5.
No, this code isn't good enough to be added to main sipml5 code, it's just a
proof of hypothesis about AES :)
Original comment by [email protected]
on 28 Sep 2012 at 9:51
from sipml5.
Further testing gives me another result. When I call through asterisk to real
voip provider, I get the same DOM exception, but in chrome log there are
another messages:
[6604:7880:1541542568:INFO:CONSOLE(55)] "State machine:
c0000_Outgoing_2_Connected_X_i2xxINVITE", source:
http://192.168.224.226/src/tinySAK/src/tsk_utils.js?svn=10 (55)
[3844:8012:1541542584:VERBOSE3:webrtcsdp.cc(1253)] Ignored line: a=fmtp:101 0-16
[3844:8012:1541542584:VERBOSE3:webrtcsdp.cc(1253)] Ignored line:
a=silenceSupp:off - - - -
[3844:8012:1541542584:VERBOSE3:webrtcsdp.cc(1253)] Ignored line: a=ptime:20
[3844:8012:1541542584:VERBOSE3:webrtcsdp.cc(1253)] Ignored line:
a=ice-options:google-ice
[3844:8012:1541542584:VERBOSE1:webrtcsession.cc(372)] SetRemoteDescription
called with action in wrong state, action: 2 state: 6
And the main inconvinience is that in case of local voip in lan (ip 192.168...)
there is all ok (bidirectional sound even with this DOMException), but when use
public voip provider there is sound only in "from browser" direction and no
sound coming to browser. What may be wrong?
Original comment by [email protected]
on 1 Oct 2012 at 8:50
from sipml5.
Try to put on hold and then resume. (I experienced also what you encountered)
Original comment by [email protected]
on 4 Oct 2012 at 10:09
from sipml5.
I've tryed it, nothing changed after resuming...
Original comment by [email protected]
on 5 Oct 2012 at 6:04
from sipml5.
A quick update:
With chrome stable (23.0.1271.64 m) and sipml5 r131 , I don't have this problem
any more and I don't need a patch.
Looks like the issue is removed and sipml5 is working on my ubuntu 12.04-64
with the default source.
Now I only see one line of crypto (80) in SDP for my outgoing calls.
Original comment by [email protected]
on 14 Nov 2012 at 9:46
from sipml5.
Confirm only one line with crypto 80 in chrome 23.0.1271.64 m. Issue should be
closed:)
Original comment by [email protected]
on 15 Nov 2012 at 4:22
from sipml5.
Original comment by [email protected]
on 15 Nov 2012 at 1:43
- Changed state: Done
from sipml5.
i have the same issue on the newest chrome. I tried this guys patch but now I
have no audio..
Original comment by [email protected]
on 31 Jul 2013 at 10:47
from sipml5.
The same problem exists on Chrome 32.0.1700.102 :(
Original comment by [email protected]
on 29 Jan 2014 at 11:38
from sipml5.
same problem here too. sipml5 sends AES 32 + 80.
static sound in receiving end, but sipml5 client can hear audio from called
client.
Original comment by [email protected]
on 15 May 2014 at 5:58
from sipml5.
also have the same issue with chrome Version 40.0.2214.115 (64-bit), in
asterisk res_srtp.c:407 ast_srtp_unprotect: SRTP unprotect failed with:
authentication failure 10
Original comment by [email protected]
on 13 Mar 2015 at 1:57
from sipml5.
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