Audio not working when using Chrome

[Aura ITC]

Hi Everyone,

I am a newbie to BBB, but honestly want to thank the developers for such a great product! 

I have setup BBB and everything is working as expected. I have also setup https (SSL) and this one issue is driving me a little crazy.

When I try to join the conference using Chrome, I get the following:

"Detected the following WebRTC issue: Error 1004: Failure on call (reason=Request Timeout). Do you want to try Flash instead?"

If I click "yes" to try flash, everything works as expected. I get audio and video.

I tried the troubleshooting steps listed on: (http://docs.bigbluebutton.org/install/install.html#audio-not-working) & have changed settings as mentioned on the troubleshooting steps to my server's external IP address, but Chrome still fails with the above message.

IE, Firefox work in default flash mode (I think it is expected?) 

I would deeply appreciate any assistance in this regard.

Thank you!

[Chad Pilkey]

Did the WebRTC audio work before you went through the SSL steps? Can the same computer connect to demo.bigbluebutton.org and join the audio bridge successfully?

Firefox should be trying to connect though WebRTC first instead of Flash too.

[Aura ITC]

Dear Chad,

Thank you for your response.

The audio worked before SSL, but again on flash only as I believe for Chrome to do non-flash SSL was needed? I think I read that somewhere based on the error I saw on Chrome before SSL. Audio works on flash version on Chrome too now (SSL or non SSL)

This may be a newbie question, but I do not have all UDP ports 16384 - 32768 accessible yet. I just have UDP 16384 & UDP 16385 accessible.  

Would this cause the issue?

Thank you!

[Chad Pilkey]

Ports not being open could definitely cause it to fail, but I would normally expect a different error message. I don't think FreeSWITCH chooses ports in sequence so any of the ~16,000 ports could be selected for use.

You might be able to limit the ports used by editing /opt/freeswitch/conf/autoload_configs/switch.conf.xml and then look for the following lines.
    <!-- RTP port range -->
    <!-- <param name="rtp-start-port" value="16384"/> -->
    <!-- <param name="rtp-end-port" value="32768"/> -->
You should be able to uncomment the parameters and modify the range. I would suggest keeping the range above 10 for testing purposes.

About SSL and Chrome, only the beta versions of Chrome put an HTTPS restriction on WebRTC right now and Firefox has no such restriction. I would suggest testing with Chrome 46 or Firefox to reduce your potential causes of error.

[Aura ITC]

Dear Chad,

Thanks again for your time & response.

I will test your recommendations & report back with the results.

Kind Regards,

---

From: Chad Pilkey
Sent: ‎11/‎2/‎2015 11:49 PM
To: BigBlueButton-Setup
Subject: [bigbluebutton-setup] Re: Audio not working when using Chrome

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[Aura ITC]

Dear Chad,

I tested, but the results are still the same. Chrome (46) works in flash only mode. Firefox does not even try and directly fails (as flash is blocked) and fallback to HTML5.

From the same computer, I can join the demo sessions from chrome on the BBB demo meeting without any issue. 

Is there anything else I could try please?

Thank you!

[Fred Dixon]

Hi Aura,

> This may be a newbie question, but I do not have all UDP ports 16384 - 32768 accessible yet. I just have UDP 16384 & UDP 16385 accessible.  

We recommend you open the range of ports required by the installation.  If you don't, WebRTC audio likely won't work.

Regards,... Fred

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[Aura ITC]

Hi Fred,

Thanks for the response.

As my deployment is on Azure, I got a Instance Level Public IP (ILPP) today which has all endpoints open, but I still have the same issue.

I see Chrome shows "calling" and then fails with the initial error I posted. (time out) & falls back to flash.

What exactly is it "calling"?

Kind Regards,

---

From: Fred Dixon
Sent: ‎11/‎3/‎2015 6:59 PM
To: bigbluebu...@googlegroups.com
Subject: Re: [bigbluebutton-setup] Re: Audio not working when using Chrome

[Chad Pilkey]

"Calling" means that the user's browser is trying to make a call to the server to connect the audio. There are a couple of steps involved in the procedure, but we don't put messages for all of the steps.

Since the timeout message is coming from the FreeSWITCH server my googling points to the issue likely being a misconfigured server. People had double quotes (""value" instead of "value") or empty values ("" instead of "value"). Did you make any changes to the FreeSWITCH configuration files other than the ones in our documentation? Also, double check the lines mentioned in our documentation to rule out formatting errors.

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[Fred Dixon]

Hi Aura,

I did a google search on

  bigbluebutton azure

You should check these previous threads:

  https://groups.google.com/d/msg/bigbluebutton-setup/ANsfxK6DqjM/wqFdENo6aBgJ

 and

  https://groups.google.com/d/msg/bigbluebutton-setup/moLuqehefBc/-PjzlApeIakJ

Regards,... Fred

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[Prakash]

We are having similar issue with Azure ie  WebRTC over https. Enabled ILPP and only http works fine.

Noticed that proxy_pass IP specified below is Internal IP ( though it didn't work with public ip as well )

In /etc/bigbluebutton/nginx/sip.nginx file

location /ws {

        proxy_pass https://10.78.146.158:7443;

        proxy_http_version 1.1;

        proxy_set_header Upgrade $http_upgrade;

        proxy_set_header Connection "Upgrade";

        proxy_read_timeout 6h;

        proxy_send_timeout 6h;

        client_body_timeout 6h;

        send_timeout 6h;

}

Rest all configuration are as mentioned in documents.

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[Aura ITC]

Hi Fred & Chad,

I found one problem and am ahead of that error.

Basically, the documentation to setup SSL states the following point:

+++

Now the websocket forwarding address in nginx must be updated. Edit the file /etc/bigbluebutton/nginx/sip.nginx and change the protocol and port on the proxy_pass line as shown:

location /ws {
  proxy_pass https://203.0.113.3:7443;

  proxy_http_version 1.1;
  proxy_set_header Upgrade $http_upgrade;
  proxy_set_header Connection "Upgrade";
  proxy_read_timeout 6h;
  proxy_send_timeout 6h;
  client_body_timeout 6h;
  send_timeout 6h;

}

+++

But then, under the section audio not working, it states:

+++

Edit /etc/bigbluebutton/nginx/sip.nginx to
location /ws {
        proxy_pass http://EXTERNAL_IP_ADDRESS:5066;

        proxy_http_version 1.1;
        proxy_set_header Upgrade $http_upgrade;
        proxy_set_header Connection "Upgrade";
        proxy_read_timeout 6h;
        proxy_send_timeout 6h;
        client_body_timeout 6h;
        send_timeout 6h;
}

+++
So when I was troubleshooting, I had set it to:
proxy_pass http://INTERNAL_IP_ADDRESS:5066; (I Don't remember why!) :( and it seems this was leading to the timeout error

Now, I have set this to proxy_pass https://INTERNAL_IP_ADDRESS:7443 and I see "calling" and then "connecting" and then it fails with this error:
Detected the following WebRTC issue: Error 1007: ICE negotiation failed. Do you want to try Flash instead?

I can confirm the ports (UDP) required are open. I have reduced the range as per Chad's earlier email. 

Can you let me know which IP ADDRESS needs to be used for proxy_pass. Is it internal or external? The documentation for setting up SSL has an example (I think) IP as 203.0.113.3

Thank you!

[Chad Pilkey]

The SSL and "Audio not working" sections change different parts of the proxy_pass. The former changes the protocol and port and the latter changes the IP address. If you need to follow both steps then you will end up with "proxy_pass https://<external_address>:7443".

As for the ICE negotiation error it could either be caused by FreeSWITCH offering an internal IP address or the UDP ports aren't being forwarded to your server correctly. To diagnose, connect to a meeting, open the browser console (F12), try to connect to the audio. Once the error message pops up search through the browser console (ctrl+F) for "a=candidate". There should be two groupings of candidates. The first grouping is your possible candidates to be sent to the server and the second is the candidate offered by the server. Check the IP address and port in the candidate offered by the server. The IP address should be your public IP and the port should be within the range that you have forwarded.

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[Aura ITC]

Thanks Chad.

I looked at the browser console and it led me to change the proxy_pass to https://EXTERNAL_IP:7443

Now it fails with this error:

Detected the following WebRTC issue: Error 1004: Failure on call (reason=Unavailable). Do you want to try Flash instead?

It seems more people have been having the same error (new one as above) with SSL

1) https://groups.google.com/forum/#!topic/bigbluebutton-setup/EkICpUv73Mk 

2) https://groups.google.com/forum/#!msg/bigbluebutton-setup/pWPRdCKZ0dU/Icu6ojALAwAJ

Both the above are exact same issue with SSL

I am not sure if it is an Azure Specific issue, but any insights will be appreciated. 

Thank you!

[Aura ITC]

Hi Fred & Chad,

As other people are experiencing the same issue with SSL (Detected the following WebRTC issue: Error 1004: Failure on call (reason=Unavailable). Do you want to try Flash instead?) - I wanted to check if anyone has SSL setup running outside of your testing with the current BBB version?

Do you want me to try anything else? 

I need to get this running so if commercial support is the way to go, let me know. I feel this could be either a bug or a doc config bug issue as other ppl are hitting this too.

Thanks!

[Marc Schipperheyn]

As for the ICE negotiation error it could either be caused by FreeSWITCH offering an internal IP address or the UDP ports aren't being forwarded to your server correctly. To diagnose, connect to a meeting, open the browser console (F12), try to connect to the audio. Once the error message pops up search through the browser console (ctrl+F) for "a=candidate". There should be two groupings of candidates. The first grouping is your possible candidates to be sent to the server and the second is the candidate offered by the server. Check the IP address and port in the candidate offered by the server. The IP address should be your public IP and the port should be within the range that you have forwarded.

What I'm seeing is this. Which looks like the internal ip address

a=candidate:1202022811 1 udp 2122265343 fd9a:ca93:c921:1:c01b:8a10:17b0:e66e 54948 typ host generation 0

a=candidate:2718281765 1 udp 2122197247 2804:431:b718:a3c6:1d64:15b4:7c6:f753 54949 typ host generation 0

a=candidate:2311777672 1 udp 2122129151 192.168.1.30 64330 typ host generation 0

a=candidate:1202022811 2 udp 2122265342 fd9a:ca93:c921:1:c01b:8a10:17b0:e66e 64331 typ host generation 0

a=candidate:2718281765 2 udp 2122197246 2804:431:b718:a3c6:1d64:15b4:7c6:f753 64332 typ host generation 0

a=candidate:2311777672 2 udp 2122129150 192.168.1.30 51497 typ host generation 0

a=candidate:153523563 1 tcp 1518285567 fd9a:ca93:c921:1:c01b:8a10:17b0:e66e 0 typ host tcptype active generation 0

a=candidate:3968224469 1 tcp 1518217471 2804:431:b718:a3c6:1d64:15b4:7c6:f753 0 typ host tcptype active generation 0

a=candidate:3343454584 1 tcp 1518149375 192.168.1.30 0 typ host tcptype active generation 0

a=candidate:153523563 2 tcp 1518285566 fd9a:ca93:c921:1:c01b:8a10:17b0:e66e 0 typ host tcptype active generation 0

a=candidate:3968224469 2 tcp 1518217470 2804:431:b718:a3c6:1d64:15b4:7c6:f753 0 typ host tcptype active generation 0

a=candidate:3343454584 2 tcp 1518149374 192.168.1.30 0 typ host tcptype active generation 0

a=candidate:2145031260 2 udp 1685921534 201.69.186.254 51497 typ srflx raddr 192.168.1.30 rport 51497 generation 0

a=candidate:2145031260 1 udp 1685921535 201.69.186.254 64330 typ srflx raddr 192.168.1.30 rport 64330 generation 0

 

201.69.186.254 is the external ip address of my personal router

So, I need to solve the bbb server serving the internal in stead of the external address

[Marc Schipperheyn]

BTW, not trying to highjack this thread. But we seem to have the same issue.

I also tried manually setting 

<X-PRE-PROCESS cmd="set" data="local_ip_v4=OUT_EXTERNAL_IP"/>

as suggested in another thread. 

But this lead to another error

WebSocket connection to 'wss://videomeet.ology.com.br/ws' failed: Error during WebSocket handshake: Unexpected response code: 404

Anyway, it seems to me that the root of the issue is related to the internal ip-address being used in stead of the external in some cloud hosted scenarios.

[Aura ITC]

Marc:  What exact error are you seeing for Chrome?

1) Detected the following WebRTC issue: Error 1007: ICE negotiation failed. Do you want to try Flash instead?

OR

2) (Detected the following WebRTC issue: Error 1004: Failure on call (reason=Unavailable). Do you want to try Flash instead?) 

IF # 1, then you need to edit the following:

proxy_pass to https://EXTERNAL_IP:7443

HTH

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[Marc Schipperheyn]

I get #2. I followed the https guide, so I should be ok in terms of the supposed configuration

[Aura ITC]

ok, thanks. # 2 is where we all are stuck! 

On Thu, Nov 5, 2015 at 7:29 PM, Marc Schipperheyn <m.schip...@gmail.com> wrote:

I get #2. I followed the https guide, so I should be ok in terms of the supposed configuration

--

[Marc Schipperheyn]

Yeah, it's time for an expert to wave his magic wand!

[HostBBB.com]

SSL configuration as documented in wiki works with .90 and 1.0beta configurations pretty much as documented from my experience. we have it deployed in both versions for some time now.

I would get the base non SSL  http BigblueButton server working first and fully tested before apply the SSL stuff.

I would suspect a UDP port issue.  Should be able to do some simple tests on random udp port in correct range to verify.

regards,

Stephen

[Marc Schipperheyn]

My money is on some weirdness leading to the internal ip being used in stead of the external

[Prakash]

For me it did work with http. Hence doesn't look like internal IP or UDP port open issue.

[HostBBB.com]

http://docs.bigbluebutton.org/install/install.html#audio-not-working

This usually solves the issue on server providers that don't have the external ip appearing in ifconfig  i.e aws

Also the first entry in instructions.

Remove this line

<X-PRE-PROCESS cmd="set" data="local_ip_v4=xxx.yyy.zzz.qqq"/>

make sure you remove and just don't comment out.  I had issues with freewitch xml parser picking up this even though it is commented out.

regards,

Stephen

hostbbb.com

On Thursday, November 5, 2015 at 9:38:34 AM UTC-5, Marc Schipperheyn wrote:

[Aura ITC]

Hi Stephen,

That troubleshooter has been followed exactly as it states. Still the same issue only with SSL

Thx!

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[HostBBB.com]

Nov 2 you had posted   "Chrome (46) works in flash only mode"

Has Chrome ever connected WEBRTC with the standard HTTP setup, no SSL?

On Thursday, November 5, 2015 at 9:54:38 AM UTC-5, Aura ITC wrote:

Hi Stephen,

That troubleshooter has been followed exactly as it states. Still the same issue only with SSL

Thx!

On Thu, Nov 5, 2015 at 8:23 PM, HostBBB.com <207...@gmail.com> wrote:

http://docs.bigbluebutton.org/install/install.html#audio-not-working

This usually solves the issue on server providers that don't have the external ip appearing in ifconfig  i.e aws

Also the first entry in instructions.

Remove this line

<X-PRE-PROCESS cmd="set" data="local_ip_v4=xxx.yyy.zzz.qqq"/>

make sure you remove and just don't comment out.  I had issues with freewitch xml parser picking up this even though it is commented out.

regards,

Stephen

hostbbb.com

On Thursday, November 5, 2015 at 9:38:34 AM UTC-5, Marc Schipperheyn wrote:

My money is on some weirdness leading to the internal ip being used in stead of the external

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[Prakash]

For me WebRTC works with Chrome 64 in http mode ( without SSL configuration changes)

I compared browser console log between http and https scenarios.  And noticed that in https, SIP Invite has below error response.

SIP/2.0 480 Temporarily Unavailable

And in  http case,  

SIP/2.0 200 OK

[Prakash]

Sorry typo, I meant Chrome (46).

[Aura ITC]

I get the same with SSL "Temporarily Unavailable" :(

SIP/2.0 200 OK

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[Marc Schipperheyn]

Interesting suggestion! However, it did not solve the issue

[Chad Pilkey]

Marc, Aura ITC, Prakash,

We think we tracked the issue down. The problem is caused because of the interaction of the external IP and HTTPS steps. There are two regular expressions in /opt/freeswitch/conf/dialplan/public that both end up failing if you follow both steps. The first, bbb_sip.xml, fails if you follow the external IP steps and the second, bbb_webrtc.xml, fails if you follow the HTTPS steps. Could one of you try editing /opt/freeswitch/conf/dialplan/public/bbb_webrtc.xml and change the following line:
<condition field="${sip_via_protocol}" expression="^ws$" break="on-false">
To
<condition field="${sip_via_protocol}" expression="^wss?$" break="on-false">

The modified regular expression should allow the call to proceed when it comes from a secure websocket.

If the change works we'll make sure to update the documentation and fix the problem in the packaging as well.

Chad

SIP/2.0 200 OK

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[Marc Schipperheyn]

Hi Chad,

Thanks for that. It did the trick!

Tested on Version 48.0.2555.0 canary (64-bit)

Cheers,

Marc

[Chad Pilkey]

I'm glad to hear that it worked. I'll update the documentation to include this step.

[Aura ITC]

Hi Chad,

This worked for me as well. Thank you for your time :)

If we ever meet, drinks are on me..

Cheers,

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[Prakash]

It worked for me too.  Thank you Chad.

On Friday, November 6, 2015 at 12:04:10 AM UTC+5:30, Aura ITC wrote:

Hi Chad,

This worked for me as well. Thank you for your time :)

If we ever meet, drinks are on me..

Cheers,

On Thu, Nov 5, 2015 at 11:27 PM, Chad Pilkey <capi...@gmail.com> wrote:

I'm glad to hear that it worked. I'll update the documentation to include this step.

On Thursday, November 5, 2015 at 12:50:31 PM UTC-5, Marc Schipperheyn wrote:

Hi Chad,

Thanks for that. It did the trick!

Tested on Version 48.0.2555.0 canary (64-bit)

Cheers,

Marc

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