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Summary:ASTERISK-23425: No sound when i make call from chrome via webrtc (sipml5) to asterisk extension. Asterisk return answer without ice-ufrag and ice-pwd.
Reporter:Kirill Ushakov (Kirus)Labels:asterisk sdp
Date Opened:2014-03-06 02:00:57.000-0600Date Closed:2014-04-08 12:58:53
Priority:MajorRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/SRTP Channels/chan_sip/WebSocket Resources/res_http_websocket Resources/res_srtp
Versions:11.4.0 11.7.0 11.8.0 11.8.1 Frequency of
Occurrence		Constant
Related
Issues:
is related toASTERISK-23515 ICE info missing ice-ufrag and ice-pwd in INVITE received by caller
Environment:********** OS *********** Distributor ID: Ubuntu Description: Ubuntu 12.04.3 LTS Release: 12.04 Codename: precise **********kernel************* Linux version 3.8.0-29-generic (buildd@panlong) (gcc version 4.6.3 (Ubuntu/Linaro 4.6.3-1ubuntu5) ) #42~precise1-Ubuntu SMP Wed Aug 14 16:19:23 UTC 2013 x86_64 GNU/Linux **********Chrome************ Chrome 33.0.1750.146 m **********Asterisk************ Astersik version 11.8.0 with SRTP ( ./configure CFLAGS=-fPIC --prefix=/usr ) configured. ***Asterisk config users.conf*** [4343] canreinvite = no type = peer host = dynamic context = mycontext hassip = yes hasiax = no nat = force_rport,comedia qualify = no encryption = yes avpf = yes ;savpf = yes language = ru videosupport = no directmedia = no disallow = all allow = alaw secret = mysecret transport = ws,udp icesupport = yes ***called extention in extensions.ael*** 7 => { Answer(); Playback(hello-world); MusicOnHold(default,300); Hangup(); }Attachments:( 0) asterisk_output.txt
( 1) chrome_output.txt
( 2) extensions.ael.txt
( 3) jira-jssip.txt
( 4) jira-sipdebug.txt
( 5) jira-sipml5.txt
( 6) os_kernel_asterisk_chrome.conf
( 7) rtp.conf.txt
( 8) sip.conf.txt
( 9) sipconf.txt
Description:No sound when i make call from chrome via webrtc (sipml5) to asterisk extension. Asterisk return answer without ice-ufrag and ice-pwd.

In attachments i change my "External IP" and "Domain" to "195.195.195.195"  and "my.domain.com"

for reproduce the problem, your server must be behind the NAT.
Comments:By: Rusty Newton (rnewton) 2014-03-06 16:15:20.545-0600

There are a few ICE relevant patches recently and I'm not sure if this issue is affected by them or not.

* Don't use users.conf, it can be weird
* Try following https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
** You might try with your own SIPML5 client as well as the demo to compare
* Try with the latest SVN of Asterisk 11

This sounds like ASTERISK-23176, but no one has provided enough detail to reproduce. What compiler flags are you using on build?
By: Kirill Ushakov (Kirus) 2014-03-11 02:26:17.442-0500

Hello Rusty.
As you say, i moved user "4343" from user.conf to sip.conf
I'm carefully read instruction, checked configuration and didn't found something new.
I'm already use four asterisk servers with sipml5 but they have real IP address and all works fine. This is first server behind the NAT.
Latest (410106) snv build of "Asterisk11" doesn't help me.
Using standart build options: loadable_modules, build_native.
I seen this issue before and tryed with "don't_optimize" and "better_backtraces" too.
i'm using latest version sipml5 and tryed to use demo version sipml5.
Nothing helped me.
By: Rusty Newton (rnewton) 2014-03-13 17:07:37.646-0500

Please attach your complete sip.conf, rtp.conf and extensions.conf (all with a .txt format for ease of viewing in the browser)

Are both the Asterisk server and the WebRTC client behind separate NATted networks?
By: Kirill Ushakov (Kirus) 2014-03-17 04:25:47.574-0500

my configs attached.
in sip.conf.txt
udp port changed.  
domain name changed to my.domain.com

Yes, both the Asterisk server and the WebRTC client behind separate NATted networks.
All used udp/tcp ports opened.

By: Kirill Ushakov (Kirus) 2014-03-25 10:19:41.478-0500

Hello.
i'm tryed to use Kamailio websocket instead asterisk websocket and received same error.
Chrome Sipml5 SetRemoteDescription failed: Called with an SDP without ice-ufrag and ice-pwd.
By: Kirill Ushakov (Kirus) 2014-03-26 03:45:11.843-0500

Tryed to use patched asterisk 11.4 and it works on 50% and have ice-ufrag, ice-pwd.
call from pstn to sipml5 - one way audio.
call from sipml5 to sipml5 normal calls.

By: Kirill Ushakov (Kirus) 2014-03-26 07:16:46.809-0500

Making more tests with asterisk 11.4 and if i'm behind the NAT, i still no have ice-ufrag and ice-pwd.
By: Rusty Newton (rnewton) 2014-03-31 17:21:24.407-0500

Please test with Asterisk 11.9.0-rc1 or above (or latest SVN) as we have a report that the issue no longer occurs in this version.
By: Kirill Ushakov (Kirus) 2014-04-01 10:09:08.584-0500

Tried asterisk versions: 11.9.0-rc1 and Asterisk SVN-branch-11-r411531M
Same error no ice-ufrag ice-pwd.

And now added new error in both versions:
Call from chrome to chrome browser:
[Apr  1 22:54:53] WARNING[16443][C-00000000]: chan_sip.c:10512 process_sdp: Rejecting secure audio stream without encryption details: audio 51345 RTP/SAVPF 111 103 104 0 8 106 105 13 126


By: Max E. Reyes Vera J. (navaismo) 2014-04-02 19:02:40.923-0500

I have tested with asterisk 11.9.0-rc1 with sipml5 and jssip to a playback extension and the issue with ufrag gone, now asterisk and the browser have the audio, also with an echo test.

I'm attching my logs and confs.

Please share your setup to reproduce it.
By: Kirill Ushakov (Kirus) 2014-04-03 05:08:48.537-0500

Hello Max,Rusty.
I'm sorry, after installation forgot to enable icesupport in sip.conf.
Tested again Asterisk SVN-branch-11-r411531M
For now i can call from:
* chrome (33.0.1750.154 m) to asterisk ( SVN-branch-11-r411531M)  extension  (sound in both sides)
* chrome (33.0.1750.154 m) to chrome (33.0.1750.154 m)  (sound in both sides)
* I made ten calls from "pstn" to "queue" to "chrome (33.0.1750.154 m)"  sound in both sides 90% / one way sound 10%
Also asterisk works with latest Opera (20.0.1387.91)

Other browsers: Chrome canary (36.0.1923.2 latest), FireFox (28 latest), ie10 with wertc4all not work, i receive this error:

[Apr 1 22:54:53] WARNING[16443][C-00000000]: chan_sip.c:10512 process_sdp: Rejecting secure audio stream without encryption details: audio 51345 RTP/SAVPF 111 103 104 0 8 106 105 13 126

This error worried me because "chrome canary" in this list.

Best regards, Kirill



By: Max E. Reyes Vera J. (navaismo) 2014-04-03 12:38:34.199-0500

Canary use the lates changes so prepare yourself for issues with futures releases of chrome. WebRTC doesn't have a RFC and many changes are coming, here you can see why is failing with canary version: https://code.google.com/p/webrtc/issues/detail?id=2774

So in the future all APIs must be updated and asterisk also or use a mediagateway between webrtc and your pbx.
By: Kirill Ushakov (Kirus) 2014-04-07 06:36:44.556-0500

Hello MAX.
Thank you for your answer.

Webrtc implemented in browser and if there some changes, asterisk or media gateway must implement them too.
Media gateway (if you mean webrtc2sip) developers update his own software faster than asterisk developers?  
I'm little don't understand where is profit here.

I think much better if asterisk developers watch your link and prepare asterisk for this changes :)

Best regards, Kirill
By: Rusty Newton (rnewton) 2014-04-08 12:58:35.237-0500

Closing this out since things appear to be working now.

{quote}
Other browsers: Chrome canary (36.0.1923.2 latest), FireFox (28 latest), ie10 with wertc4all not work, i receive this error:

[Apr 1 22:54:53] WARNING[16443][C-00000000]: chan_sip.c:10512 process_sdp: Rejecting secure audio stream without encryption details: audio 51345 RTP/SAVPF 111 103 104 0 8 106 105 13 126
{quote}

For the above issue, you may file a new issue if you like. However as Max said, there will be more changes coming, WebRTC is a moving target and the community of developers is doing their best to keep up with things as time allows.

As always, patches are accepted, you are welcome to contribute with code where you can! :)  Thanks for the report!