Posts in Category "UV"

Error creating or editing a UV profile

This article is applicable only to customers using Universal Voice telephony profiles and running into a very specific error in the user interface.  If you are not experiencing the exact problem below, you do not need to worry about this article.

Update: 5/23 – We are fixing this issue in our hosted environment as part of the 9.5.3 upgrade which is currently being rolled out across the various clusters  over the next month.

For licensed (on premise) customers (this includes customers in the ACMS program and customers hosted by other Connect hosting partners other than Adobe), we will be releasing a patch in early June, 2016 for the Connect 8.x and Connect 9.x environments.

 

The error: ‘This field only accepts valid DTMF commands and phone numbers (+,0-9,*,#,-,space,A,B,C,D).

uv.jpg

Issue: Recently an issue has cropped up with editing existing (or creating new) Universal Voice (UV) profiles.  This is directly related to a required change that was introduced in the latest version of the Adobe Flash Player.  The behavior is that if you try to go into My ProfileMy Audio Profiles > and click the ‘New Profile’ button  or  My Profile > My Audio Profiles > and select an existing UV profile and click the ‘Edit‘ button, you will get this error (above) when trying to SAVE the profile.

This issue is resolved in the upcoming Adobe Connect release (9.5.3).  In the interim (for Adobe Hosted customers while they wait to be upgraded) or for Licensed (on-premise) customers who may be on an older version of the application and not able to upgrade to 9.5.3 right away when it comes out, there are basically 3 ways around this problem…

    • [1] Downgrade Flash player to a version prior to version 21 or use a browser or machine that has an older version of the Flash Player (prior to 21).  This would require you to uninstall the Adobe Flash player in a browser where it is not embedded as part of the actual browser, and install an older version from the archived versions here.
    • [2] Create (or just edit an existing) mms.cfg file on the system and add the following flag (line) in the file (this will ONLY work for the next 3 months approximately).  This is only a temporary solution and the Flash Player team plans to remove support for this flag in upcoming updates but for now, it can be used until we deliver possible patches for older Connect versions and fix this in the upcoming 9.5.3 release.


      enablePCRE2=0

This mms.cfg file can be created on your system in a simple text editor if it doesn’t already exist on your computer.  First do a search for it.  If it doesn’t exist, simply create a new file in Textpad or Notepad or Textedit or whatever you use for a simple text file editor.  Add the one line above and save the new or existing mms.cfg file (with that line as either the only line in the file or added to the bottom of whatever else is in that file).

On Windows systems, the mms.cfg file would be (or needs to be) placed in:

C:/Windows/system32/Macromed/Flash (32-bit Windows) and C:/Windows/SysWOW64/Macromed/Flash (64-bit Windows).

On Mac systems, the mms.cfg file would be (or needs to be) placed in:

 MainDisk:Library:Application Support:Macromedia

    • [3] Use the XML API.  As this is only a Flash Player issue and only affects users who are using the UI in Adobe Connect, if you are familiar with the XML API, you can use the ‘telephony-profile-update‘ API as a workaround. If you are not familiar with the XML API, the documentation for the API is here.  If this is still confusing for you and you are not quite sure how to execute XML API calls, the steps below will not be for you.  If you need clarity on the below, please contact Adobe Connect support as you normally would.

The workflow is as follows for the API (again if you are not familiar with the API, you need to either familiarize yourself with the product documentation, ask and admin for assistance, or use one of the other workflows above to work around this issue):

For existing profile updates:

1) Find the profile-id by making this API call in the browser once you’ve logged into your Adobe Connect account: https://{yourConnectURL}/api/xml?action=telephony-profile-list

It will list out all your profiles.  Search for the one you want to edit and make note of the ‘profile-id‘ value (numeric) and the ‘provider-id‘ value (also numeric).

2) Get the field-ids by running this API: https://{yourConnectURL}/api/xml?action=telephony-profile-info&profile-id=xxxxxxxx  (where the profile-id = the numeric value of the profile from call 1 above).

This will show you an XML response with telephony-profile-fields listed.  You may have a field for a certain moderator pin or other value you need to edit.  The field will be listed as an ‘x-tel‘ field.  It will be in a format that looks like this: ‘x-tel-123456789‘ or something similar with numbers after the ‘x-tel’.

You can also get the fields for the provider by taking the provider-id and running this API: https://{yourConnectURL}/api/xml?action=telephony-provider-field-list&provider-id=xxxxxxxx

This will show you all the ‘x-tel-xxxxxxx’ values you need to use.

3) To update that field, you would make this API call:  https://{yourConnectURL}/api/xml?action=telephony-profile-update&profile-id=xxxxxxxx&provider-id=xxxxxxxx&field-id=x-tel-123456789&value=xxxxxxx&profile-status=enabled   (where profile-id, provider-id both equal their corresponding numeric values from the first API call above…and where the field-id= the x-tel value from call 2…. and where value= the new value you need to edit)…

For creating NEW profiles:

1) Find the provider ID you want to create a new UV profile from with this API call: https://{yourConnectURL}/api/xml?action=telephony-provider-list .   The provider-id value gets returned in this result.

2) Obtain a list of all the applicable provider fields by running this API call: https://{yourConnectURL}/api/xml?action=telephony-provider-field-list&provider-id=xxxxxxx (where the provider-id = the numeric provider-id for that UV provider).

3) Run this API to create a new UV profile: https://{yourConnectURL}/api/xml?action=telephony-profile-update&provider-id=xxxxxxxx&profile-name=xxxxxxx&field-id=x-tel-xxxxxxxxx&value=xxxxxxx&profile-status=enabled  (where the provider-id= the numeric provider-id for that UV provider… field-id= the x-tel field you want to populate …. value = the value (numeric) of that field…. profile-name = the name of your new profile …. profile-status=enabled ).

Of course, you could have more than one field to populate as everyone’s UV providers/profiles could be custom and unique to a requirement.

If you are unsure, you need to talk to your Connect admin or contact Adobe support of course for any questions regarding the above profile creation or update steps.  A support agent can assist you with identifying values and creating / updating profiles if needed.

Connect Meeting and Client-side Speaker Audio Output Control

There is a Connect feature request from various customers in place asking for the Connect Meeting GUI to offer an option to choose audio output devices. The request is a complex one because the audio output control options are opaque to Flash; the settings for audio output are in the various operating systems (OS) of the many possible clients. Connect uses what is chosen as the OS default as depicted in our help documents:

Set up audio broadcasting

The feature request number is: CONN-4082570; one customer recently suggested that we add expanded functionality for client speaker audio output control roughly similar to what we already have in Adobe Connect for Microphone and Webcam selection.  A speaker drop-down menu for sound output is desirable for obvious reasons.

There is no set date for implementation of this enhancement in Connect and I will update this blog entry if that changes. In the meantime, if the default client OS audio output option is not the option desired for use with Connect Meeting, the following example may help: I will show how to add a Bluetooth speaker to a Windows client and toggle the audio output in Connect from the built-in laptop Realtek speaker to a new iHome Bluetooth speaker. While audio output options may vary, by showing how it is done with this common example of a Bluetooth output device, it will hopefully help to show how other optional client-side speaker output devices may also be managed in kind.

To see the enabled audio output options on a Windows client, look at the Device Manager under the Control Panel:

sound-dev-mgr.fw

Here we see a Realtek device and this corresponds with the option in the lower right of the desktop tray:

audiowzd-bt5b.fw

Opening the mixer shows more detail:

realtek.fw

If I play music by invoking the Audio Setup Wizard in Connect Meeting, the Realtek speaker will play:

asw.fw

conn-soundoutput.fw

Since our example will be to switch to a Bluetooth speaker, the first step will be to make certain that Bluetooth is enabled. On my Lenovo, that is done by pressing the keys FN>F5 simultaneously. Here we see Bluetooth is enabled:

sound-bt.fw

The next step is to follow the device instructions to pair the Bluetooth speaker with the client computer; these will vary.

See the Bluetooth icon enabled and  highlighted in my system tray:

sound-bt1.fw

After putting the iHome speaker in pairing mode, I am able to search for it from the client:

sound-bt2.fw

sound-bt2a.fw

sound-bt3a.fw

Now we have more than one speaker option to toggle as the Device Manager and the system tray attest:

sound-dev-mgr1.fw

sound-bt5.fw

In Connect we now see the option to use the new audio output device:

sound-fin.fw

Note: The iHome Bluetooth speaker also has a built-in Microphone so the Connect Audio Setup Wizard will see it in the Microphone drop down menu.

Without audio output controls built into Connect, adding and/or changing the default audio output device in the client OS is the way to toggle the audio output option in Connect. The key thing to be aware us is the danger of audio feedback loops. When separate speakers feed back into a microphone and cause echoing in a Connect Meeting. On a mobile device such as an iPAD, without a headset the speaker audio will feed right back in the microphone; it is best practice to use a headset with iPad to prevent audio loop/echos.

New Adobe Connect Support Blog Subscription Option

Now you can stay on top of the new articles and posts by subscribing to the Adobe Connect Support Blog. Simply go to the Adobe Connect Support Blog home page and enter your email address and check off the categories about which you would like to be notified. Click “Subscribe me” and you will begin receiving  regular updates:

subscribe.fw

 

 

Troubleshooting FMG On-premise: “Error in joining Audio Conference”

If Flash Media Gateway cannot establish a connection, you may see this pop up message upon invoking Unified Voice: Error in joining Audio Conference:

fmg.fw uvtestfail.fw

 There are a number of potential causes; let’s consider some approaches to troubleshooting:

Restart the FMG service. Check the task manager to make sure that there is not a hanging FMG process. If fmgmain is hanging, terminate it manually through the task manager and then restart the FMG services. Here are the relevant services highlighted:

rel-svcs.fw

Manually check the FMG status from the Connect server  via a browser and telnet:

  • http://fmgIPAddress/2222/admin/getFMGStatus?auser=sa&apswd=fmgpassword

fmgcheck.fw

  • telnet from fmg to the Connect server on 8506
  • telnet from the Connect server to fmg on 2222
  • telnet from fmg to sip on 5060
  • Check to see if ports are listening:
    • netstat -an|find “8506”
    • netstat -an|find “2222”

Here is the flow illustrated with an eye toward the Intercall implementation:

Connect_FMG_Flow

These are the relevant logs for troubleshooting telephony issues; some Connect telephony adaptors leverage FMG:

  • \Flash Media Gateway\2.0.1.19_8x8\log\core.xx.log and master.xx.log, sip.xxxx.log

fmglogdir.fw

  • \Connect\logs\telephony\-relevant-adaptor.xx.log

logstelephony.fw

  • \Connect\logs\support\debug.xx.log

debuglog.fw

  • \Connect\logs\support\apps\_defaultHost_\telephonyProviderTesterApp\instances\7\####\application.xx.log

teltestapplog.fw

Note: When sending logs to Adobe Connect Support for scrutiny of a telephony (or any server-side) issue, it is generally prudent to send all logs if allowable by truncating them to focus on the issue at hand:

  • Stop all services: FMS, FMG, Connect & Telephony
  • Rename or delete all log directories
  • Restart all services and recreate the problem being scrutinized
  • Stop all services and zip complete log directories focused on the issue at hand and provide them to the support consultant

And along those same lines, the entire /conf/ directories, all four zipped will help us locate errors as well:

  • C:\Connect\9.3.1\TelephonyService\conf
  • C:\Flash Media Gateway\2.0.1.19\conf
  • C:\Connect\9.3.1\comserv\win32\conf
  • C:\Connect\9.3.1\comserv\conf

Relevant configuration files for troubleshooting may include:

  • \Flash Media Gateway\2.0.1.19_8x8\conf\sip.xml, rtmp.xml, http.xml, workflow.xml

fmgconfdir.fw

  • \Connect\9.3.1\TelephonyService\conf\telephony-settings.xml, server.xml

telconf.fw

  • \Connect\9.3.1\custom.ini

customdir.fw

FMG connection settings are configurable in the console on port 8510 on the Connect server. While it is recommended that FMG be installed and run on a separate server, in some cases, it is collocated with Connect. Check to make sure there are not two instances of FMG where only one is needed. This is possible as when distributing FMG onto a separate server as appropriate in a robust clustered environment, an oversight may be to leave it also installed locally on the Connect server as well:

fmg8510.fw

Delete superfluous instances and disable any local unneeded FMG service.

fmg85101.fw

  fmgsvc.fw

VoIP Bandwidth and Microphones

When using the Nellymoser codec, one microphone might produce more bandwidth over against another. Nellymoser accepts five different microphone rates values: 5, 8, 11, 22 and 44. Each of these rate values consumes bandwidth of roughly double its value. For example the 8kHz setting consumes roughly 16kbps and 11kHz setting consumes roughly 22 kbps. The default rate value is 8kHz. This may vary depending on the sound capture device in use.

The Flash Player microphone reference documentation reads:

“The default value is 8 kHz if your sound capture device supports this value. Otherwise, the default value is the next available capture level above 8 kHz that your sound capture device supports, usually 11 kHz.”

It is possible for two different microphones to consume different amounts of bandwidth.

In case of the Speex codec, it has a fixed sampling rate of 16kHz. Speex allows control of the quality by offering us 11 different encoding quality options. See the following reference table :

 

Speex NellyMoser
Quality (EncodeQuality)Mic rate is fixed at 16 Bandwidth(kbps) Quality (Mic Rate) Bandwidth (kbps)
0 3.95 5 11.025
1 5.75 8 16
2 7.75 11 22.05
3 9.80 22 44.1
4 12.8 44 88.2
5 16.8
6 20.6
7 23.8
8 27.8
9 34.2
10 42.2

 

Adobe Connect supports three Speex settings:

  • Fast : Encode Quality 4
  • Good : Encode Quality 6
  • Best : Encode Quality 8

Increasing the encoding quality will increase the quality of the stream but will also require greater bandwidth. The Speex codec is optimized for voice and also includes voice activity detection that allows it to reduce bandwidth.

Updating audio profile in a meeting

We can update the audio profile without interrupting the running meeting.  Below are the steps to update the Audio Profile.

Step.1

Open that meeting room and click on “Meeting” Button

image1

Step.2

Click on “Preferences” button in the drop down menu.

image2

Step.3

Click on “Audio Conference” and select the desired audio profile from the drop down menu under “Audio Profile Settings”  

image3

image4

Then we get the screen with blue bar which says “Updating Audio Profile…”

Step.4

Click on “Start Meeting Audio” under the audio menu in meeting.

image5

 

 

Arkadin Audio Profile Conference Numbers

For Arkadin customers who integrate Arkadin audio profiles into Adobe Connect Meeting rooms, they need to be very careful in what numbers they are inputting into the profile fields when they are creating the telephony profiles.  Also, if these profiles are being provisioned automatically by an application utilizing the API, developers need to make sure the values they are passing in via the web services are also correct.

Arkadin has 3 phone numbers that are required when building an audio profile (UI pictured below).

arkprof

  1. Toll Access Number‘ (in the API this is: ‘x-tel-arkadin-conference-number‘)
  2. Toll Free Access Number‘ (in the API this is ‘x-tel-arkadin-conference-number-free‘)
  3. SIP Access Number‘ (in the API this is ‘x-tel-arkadin-conference-number-uvline‘)

It is crucial that you do NOT inadvertently put the Toll number (a non ‘1-8xx’ number) in for the Toll Free value and vice versa.  If you put a toll number in for the toll free number, the audio profile will save correctly, HOWEVER the UV line (Universal Voice) will not be able to connect to your meeting room when you try to start the audio (for Audio Broadcast and for Meeting Recording with Arkadin).  Universal Voice can only call out to a toll FREE number.  So if you are seeing your Arkadin audio conference not connecting correctly in the Adobe Connect Meeting room, please make sure to check your Arkadin profile that is assigned to the meeting, to make sure the toll free number is actually a toll free number, and the toll number is also correct.  The SIP access number should be set to the toll FREE number as well.

 

Connect Console Values Populate to Wrong Profile in the sip.xml File

Issue: When installing FMG as part of an on-premise Connect deployment, the fresh installation of FMG includes many default profiles in the sip. xml file. When you enter values from Connect console (port 8510 locally on the server), those values are populated to the first profile that is listed in sip.xml (sipPhone) and not to the correct sipGateway profile which is called from the workflow.xml file.

The expected behavior is that the values from Connect console should update the sipGateway profile rather than the first profile in sip.xml

Workaround: The Adobe Connect Support team is using currently approaching this problem from one of two possible ways:

  • You may copy a sipGateway profile from a vertsion of FMG prior to version 2.x and paste that into FMG 2.x.
  • You may call sipPhone profile from workflow.xml

Note: In FMG prior to version 2.x, sipGateway was the first profile listed in sip.xml.  The workflow.xml file checks the input  number and on the basis of that chooses the profile from sip.xml. With default FMG settings it will be using ‘SipGateway’ most of the time. This is scheduled to be fixed in Connect 9.3.

Demystifying Mixing VoIP and Telephony in a Meeting

With Unified Voice (UV) enabled and selected from within a meeting room:

The host may choose: Meeting>Preferences>Audio Conference>Allow participants to use Microphones: When Allow participants to use Microphones is checked, participants have power to enable their own microphones within the meeting. When it is unchecked, the host must enable microphones first and then the participant can enable the microphone with host permission within the meeting:

uv-1.fw

In either case, checked or unchecked, the host needs to first start the audio conference within any meeting:

uv-1b.fw

And the participant needs to choose the microphone option and enable it (even though the host has enabled it manually within the meeting or set it as permanently enabled within the room):

uv-1a.fw

Note that by default, when UV is in use, the telephony option is checked for the participant:

UV2.fw

The participant must select the microphone option in order to use the microphone instead of the phone; this will allow the microphone to broadcast to the users in the meeting using UV telephony:

UV3.fw

With all these settings in place, VoIP microphones can talk to telephony and telephony to VoIP and both will be audible in an archive recording for playback on demand.

Linux Sound Card Drivers cause a Crash during a Connect VoIP Session

To avoid problems with select Linux sound card drivers turn off Enhanced Audio in a Connect Meeting:

Meeting->Preferences>Microphone:

mic-enh.fw

Some crashes on Linux are avoided by turning off Enhanced Audio and canceling Acoustic Echo Cancellation Mode.