Call from chrome to softphone through asterisk cause only noise about sipml5 HOT 40 CLOSED

Some additional info can be found in 
https://groups.google.com/forum/?hl=en&fromgroups=#!topic/doubango/1Kd74RzdlmY

[deleted comment]

@ln.nima.nl
"22.0.1229.79" is not STABLE chrome but beta. STABLE is at 21....

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

comment #3 no longer make sense because "ln.nima.nl" removed it's comment (#2).

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.

sorry , I decided to edit the text to avoid misunderstanding.
my mistake :)

Previous days are previous days

@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?

Tested with "22.0.1229.79 m" and it's working as expected

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

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.

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

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.

oops... I meant "...is supported by webrtc"

[deleted comment]

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.

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. 

Here is chrome console (caller side) for ekiga.net. Hope this helps

@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 .

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

@ln.nima.nl it's just strange for me, because it looks like chrome is ok in 
this case? or I missed something.

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

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?

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.

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

No, this code isn't good enough to be added to main sipml5 code, it's just a 
proof of hypothesis about AES :)

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?

Try to put on hold and then resume. (I experienced also what you encountered)

I've tryed it, nothing changed after resuming... 

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.

Confirm only one line with crypto 80 in chrome 23.0.1271.64 m. Issue should be 
closed:)

i have the same issue on the newest chrome. I tried this guys patch but now I 
have no audio.. 

The same problem exists on Chrome 32.0.1700.102 :(

same problem here too. sipml5 sends AES 32 + 80.

static sound in receiving end, but sipml5 client can hear audio from called 
client.

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 

The same problem, call is ok with audio but after about 1 minute get silence
Tryed on 2 Asterisk servers
Chrome Version 41.0.2272.101
In tcpdump packets coming but can't see in Asterisk rtp debug 
[Apr  1 11:27:50] WARNING[9154][C-00000000]: res_srtp.c:406 ast_srtp_unprotect: 
SRTP unprotect failed with: authen                                              
       tication failure 10

It's very strange, because we can't reproduce this anymore. Also you should 
avoid using SDES-SRTP (with "crypto" line), because only DTLS-SRTP is mandatory 
(for example Firefox not supports SDES at all).
Also I think that you faced with other kind of problem and you should fill 
separate issue or fill request in doubango google group.

Fixed now
Increasing dtlsrekey parameter solved my problem