GoogleCodeExporter
commented
8 years ago Same here, on asterisk 11.6. Exactly same picture, onew-way RTP, no RTP from
Chrome.Original comment by zhega...@enaza.ru on 8 Nov 2013 at 10:37
Open GoogleCodeExporter opened 8 years ago
GoogleCodeExporter
commented
8 years ago Same here, on asterisk 11.6. Exactly same picture, onew-way RTP, no RTP from
Chrome.Original comment by zhega...@enaza.ru on 8 Nov 2013 at 10:37
GoogleCodeExporter
commented
8 years ago I have tested your script in this scenario, just replacing account and server
info:
Chrome<--wss-->webrtc2sip<-->OpenSIPs<-->Asterisk<-->Voip_Provider<-->PSTN
and it doesn't work. Conclusion: Your script is totally broken!
1)callSession = stack.newSession(,,,) is wrong. It has some missing elements
like "sip_caps".
2) what is audio-remote?
3) <audio id="audio_remote" autoplay="autoplay"> Is missing.
4) Add it and rename 'audio-remote' by 'audio_remote in 2'
5) Your 'stack = new SIPml.Stack (,,) looks weird too. Don't assume anything,
just provide all the parameters according to examples and documentation.
Because of that, this event 'm_stream_audio_local_added' never takes place.
After fixed, your script worked for me. Registration, Invite, bi-directional
audio and BYE.
Cheers!
Miguel Oyarzo
Melbourne
Original comment by miguel.o...@dubber.net on 8 Nov 2013 at 2:22
GoogleCodeExporter
commented
8 years ago Miguel Oyarzo,
Thank you very much.
(I was away for a week ...)
What was missing is "<audio id="audio_remote" autoplay="autoplay">" line
in the body .. and the audio_remote in the script!!!
AFTER THIS IT WORKS!
Thanks!!
(I just took the example script form the sipml5 test and tried to test it
.. I didn't look into the html lines .....)
(I removed some parameters like sip_caps , for testing purposes .. anyway
they are not making any impact on simple calls, at-least on my case ...)
I was testing with sipml5 test page as well (http://sipml5.org/call.htm#)
but no audio ..
Finally audio worked, after patching the Asterisk 11.5 code with the
latest patch ASTERISK-21383
(https://issues.asterisk.org/jira/browse/ASTERISK-21383)
Can you tell me, if Asterisk 11.5 alone worked for you or with the patch
ASTERISK-21383?
(And also, I've made the wss call working with asterisk, as well ( without
webrtc2sip), after modifying asterisk code ....)
Thanks &
Rgds
ThavaOriginal comment by ath...@gmail.com on 13 Nov 2013 at 1:05
GoogleCodeExporter
commented
8 years ago This is my new scenario:
1) SipMl5 -> Asteriskx.x
2) SipMl5 -> Oversip--> Asteriskx.x
I have tested Asterisk 11.5, 11.6 and 11.7. In both scenarios I got those
errors:
WARNING[17942]: sip/sdp_crypto.c:173 sdp_crypto_activate: Could not set SRTP
policies
WARNING[17942]: sip/sdp_crypto.c:173 sdp_crypto_activate: Could not set SRTP
policies
WARNING[17942]: SRTP audio stream rejected, 'Could not set SRTP policies'
(res_srtp.so is loaded)
I'm passing this parameter to ./configure: --with-crypto --with-ssl=ssl
--with-srtp
Is something missing?
Thanks,
Miguel
Melbourne
Original comment by miguel.o...@dubber.net on 28 Nov 2013 at 4:27
GoogleCodeExporter
commented
8 years ago Hello, I apply patch 21383 on asterisk 11.6 but no changes, no sound but call
is established.
Use:
http://sipml5.org/call.htm#)
Disable Video: yes
WS ws://XX.XX.XX.XX:8088/ws
ICE servers [{ url: 'stun:stun.l.google.com:19302'}]
Disable 3GPP Early IMS:yes
Cache the media stream:yes
Disable Call button options:yes
Chrome Vesrison:31.0.1650.57 m
In the asterisk log
Stopping retransmission on
of Request 125: Match Found
In Chrome Console I saw:
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS
df7jal23ls0d.invalid;rport=55704;received=XX.XX.XX.XX;branch=z9hG4bKmNBSsCX42HK6
ZIWgugFApKDGT1HBgNPy
SEND: SIP/2.0 481 Dialog/Transaction Does Not Exist
Original comment by Sitnik...@gmail.com on 28 Nov 2013 at 9:17
GoogleCodeExporter
commented
8 years ago This error usually means your SipML5 client is receiving SIP messages from a
differnt path than ws/wss. I would think in SIP/routing issue from the end
point.Original comment by miguel.o...@dubber.net on 30 Nov 2013 at 12:19
GoogleCodeExporter
commented
8 years ago Same issue , not able hear voice from both endpoints
SIP user as follow :
[8001]
type=friend
host=dynamic
secret=XXXXXXXXXXXXXXX
context=default
icesupport = yes
insecure=port,invite
allow = h264,h263p,mpeg4
Original comment by ni...@unitedinfotech.in on 4 Jan 2014 at 12:28
GoogleCodeExporter
commented
8 years ago I've fixed the issue on my side. If you give more details on the setup,
sip messages, colsole-outs.. I may be able to help ..Original comment by ath...@gmail.com on 6 Jan 2014 at 3:43
GoogleCodeExporter
commented
8 years ago What asterisk version have you tested it on?
We are using 11.7 with patches from:
https://issues.asterisk.org/jira/browse/ASTERISK-22961
https://issues.asterisk.org/jira/browse/ASTERISK-21930
https://issues.asterisk.org/jira/browse/ASTERISK-21383
And have no sound with packets flowing one way.Original comment by Jidel...@gmail.com on 28 Jan 2014 at 10:21
GoogleCodeExporter
commented
8 years ago The issue has been fixed.Original comment by Jidel...@gmail.com on 28 Jan 2014 at 11:59
GoogleCodeExporter
commented
8 years ago How you fixed this?Original comment by hks1...@gmail.com on 6 Feb 2014 at 2:55
GoogleCodeExporter
commented
8 years ago Sorry, it wasn't fixed per se, but it works whenever there is no NAT. The
question that remains, however, is how to make sure the signaling works (in a
NAT situation). This seems to be a separate issue, which we are now trying to
resolve. I will update the status of this ticket once we find a solution to
that.Original comment by Jidel...@gmail.com on 6 Feb 2014 at 3:01
GoogleCodeExporter
commented
8 years ago Why NAT is an issue for you? It is supposed that STUN provides visibility
behind NAT. SIPml has default hard-code STUN servers, if you don't provide any
in your javascript file (*stun.google* for example). At the other end in
asterisk you have to enable stun server too. Original comment by miguel.o...@dubber.net on 6 Feb 2014 at 3:15
GoogleCodeExporter
commented
8 years ago That's what we're trying to figure out. The STUN is added (google public) and
that doesn't help.
My rtp.conf contains:
icesupport=true
stunaddr=stun.l.google.com:19302
P.S. I'm not an expert so I may be mistaken in my conclusions. What I did
notice though, is that one of my PCs (not behind nat) is able to make calls
from both Firefox and Chrome over wss (with dtlsenable=yes), while for another
pc it the calls are only possible over Chrome (with dtlsenable=no) or from
Firefox (with dtlsenable=yes). So effectively that makes it a Chrome NAT
problem...Original comment by Jidel...@gmail.com on 6 Feb 2014 at 3:34
GoogleCodeExporter
commented
8 years ago I have here this error from Google Chrome 32 (Asterisk 11.6-cert:
tRemoteDescription failed: Called with an SDP without ice-ufrag and ice-pwd. Original comment by sir.lo...@gmail.com on 17 Feb 2014 at 11:54
GoogleCodeExporter
commented
8 years ago Ok I've found the problem
If you compile Asterisk with OPUS support then there will be a conflict with
ICE for some reason.
I should checkout the code of the OPUS Patch because it fails in Asterisk >11.6
(i have it in Asterisk 11.5 and is fine)Original comment by sir.lo...@gmail.com on 18 Feb 2014 at 12:47
GoogleCodeExporter
commented
8 years ago hello everyone,
i am using asterisk 11.7.0, sipml5 (www.sipml5.org/call.htm)
i am facing audio problems in (asterisk + sipml5 + chrome Version 27)
when i make calls from chrome browser to a sip phone connected to asterisk, i
can see rtp packets flowing from sip phone to chrome but no rtp packets from
chrome....
here is the log when i called from chrome to a sip phone, rtp is flowing in
only one direction
-- SIP/701-00000005 answered SIP/704-00000004
> 0xef3150 -- Probation passed - setting RTP source address to 203.199.110.73:21592
Got RTP packet from 203.199.110.73:21592 (type 00, seq 017916, ts 6646527,
len 000160)
Sent RTP packet to 203.199.110.89:39240 (via ICE) (type 00, seq 007119, ts
6646520, len 4294967284)
Got RTP packet from 203.199.110.73:21592 (type 00, seq 017917, ts 6646767,
len 000160)
Sent RTP packet to 203.199.110.89:39240 (via ICE) (type 00, seq 007120, ts
6646760, len 4294967284)
Got RTP packet from 203.199.110.73:21592 (type 00, seq 017918, ts 6646927,
len 000160)
Sent RTP packet to 203.199.110.89:39240 (via ICE) (type 00, seq 007121, ts
6646920, len 4294967284)
Got RTP packet from 203.199.110.73:21592 (type 00, seq 017919, ts 6647087,
len 000160)
Sent RTP packet to 203.199.110.89:39240 (via ICE) (type 00, seq 007122, ts
6647080, len 4294967284)
Got RTP packet from 203.199.110.73:21592 (type 00, seq 017920, ts 6647247,
len 000160)
Sent RTP packet to 203.199.110.89:39240 (via ICE) (type 00, seq 007123, ts
6647240, len 4294967284)
can anyone please help me with the solution...
thanx in advanceOriginal comment by shad.a.s...@gmail.com on 25 Feb 2014 at 6:38
GoogleCodeExporter
commented
8 years ago [deleted comment]I'm having a similar issue: established call but no audio on both ends, but I'm
using freeswitch intead of asteriks.Original comment by vasco...@gmail.com on 29 Apr 2014 at 10:08
GoogleCodeExporter
commented
8 years ago I've fixed the issue on my side. If you give more details on the setup,
sip messages, colsole-outs.. I may be able to help ..Original comment by ath...@gmail.com on 30 Apr 2014 at 5:57
GoogleCodeExporter
commented
8 years ago Ok, I'm posting the console outputsOriginal comment by vasco...@gmail.com on 30 Apr 2014 at 2:19
GoogleCodeExporter
commented
8 years ago Ok, I am attaching the console output ...
Is probably something missing in my implementation!
To receive a call my code is like that:
e.newSession.accept();
To make a call:
var makeCall = function(number){
callSession = sipStack.newSession('call-audiovideo', {
video_remote: document.getElementById('video-remote'),
events_listener: { events: '*', listener: eventsListener } // optional: '*' means all events
});
callSession.call(number);
}
And I have a tag with the midia resource at the html file ...
Original comment by vasco...@gmail.com on 30 Apr 2014 at 3:53
Attachments:
GoogleCodeExporter
commented
8 years ago HI,
I am facing similar kind of problem. I am not able to receive the remote audio.
can anybody help me out.
As per my understanding some how I am unable to redirect my remote audio to my
speaker.
Can any body please check my js code
Original comment by yogi....@gmail.com on 14 May 2014 at 1:04
Attachments:
Original issue reported on code.google.com by
ath...@gmail.comon 25 Oct 2013 at 5:51Attachments: