Posts in Category "UV"

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.

For Success with Unified Voice for On-premise Connect Deployments be kind to SIP Traffic

Issue: Avoid latency caused by packet inspection of SIP traffic for Unified Voice (UV)

To avoid latency caused by packet inspection of SIP traffic UV, simply be sure to disable SIP packet inspection on an application-aware firewall. The best-practice it to implement a global address tag in the sip.xml file on the Flash Media Gateway (FMG) server.

FMG and SIP works best with Connect when there is an absence of superfluous speed-bumps; click on this diagram thumbnail to view the traffic flow among the servers:

Connect_FMG_Flow

Connect Meeting Bandwidth Utilization using Multiple Interactive Collaborating Video Feeds

Usage question: How many Video feeds can I have running in my meeting room at once?

Answer: Let’s consider a working example around the bandwidth utilization of six Video cameras in a single meeting room consisting of one host and five participants. This working example may be that a of an interactive management meeting or of a college classroom where multiple students interact in a small group session using their webcams. From an examination of this example, you will be able to calculate video camera utilization parameters for other meetings whether they be larger or smaller ones.

To help illustrate what I mean, see this picture from our Connect 9.1 Release Notes

six-cams.fw

Each of our six webcam-wielding clients is connected to the server and will receive five video streams from the server (N-1).

Lets calculate first the number of streams outbound: 6 x 5 = 30

Lets also consider the 6 publishing streams from each client to the Connect server for a total of 36  total streams to support the Connect Video pods.

Now lets calculate the amount of bandwidth used by each stream; here you have power to decide how much bandwidth is to be used by each stream as there are many variables that Adobe Connect puts in your control:

In your meeting room, as a host, click Meeting > Preferences:

meeting-preferences.fw

Under Meeting > Preferences, there are two important options that you are going to adjust – Bandwidth and Video:

meeting-preferences-room-bandwidth.fw

meeting-preferences-video.fw

 

The size of the video streams commensurate with each webcam instance will depend on how you configure these settings.

webcam.fw

Following our example, if you go with the settings that I have depicted in the screen captures above to support the 6 Video feeds in a single meeting: DSL Bandwith and Standard Video quality = 213 kbps per stream:

36 streams x 213 = 7668 kbps or 8 Mbps for the 6 separate cameras.

There are other variables to consider as well. Building on our example, let’s say you also want to use VoIP:

VoIP.fw

DSL VoIP = 22 kbs x 36 = 792 kbs or 1Mbps (rounded up) additional bandwidth needed.

There are other ways to optimize: the video streams are always larger when clients use the Flashplayer in a browser rather than using the Connect Meeting addin. The Connect addin uses the ON2 codec and is far more economical when it comes to bandwidth utilization. For each client running without the Connect addin it is prudent to plan for an additional 50% for each of their Video streams. To avoid this additional bandwidth consumption, send out a link with the Adobe Connect Addin prior to your meeting and encourage clients to install it. It is a small modified version of the Flashplayer:

Another variable to consider that when the Video instance sizes are smaller, Connect adjusts to a lower publishing resolution to save some bandwidth. Unless you are sure the clients have the addin, the final planning number for our 12 webcam meeting is:

300 kbps for each stream (assuming that the addin will not be ubiquitous) x 36 stream = 11 Mbps + 1 Mbps for VoIP = 12Mbps.

Presenting a PowerPoint or a PDF that is uploaded to the Meeting room does not add to the overhead. Chat, Notes and Whiteboards are also insignificant with reference to bandwidth impact.

To drill home the point and procedure, let’s try the same exercise with 12 concurrent interactive collaborating Video feeds:

  • Each of our 12 clients is connected to the server and will receive 11 video streams from the server (N-1).
  • Lets calculate first the number of streams outbound: 12 x 11 = 132
  • Lets also consider the 12 publishing streams from each client to the Connect server for a total of 144  total streams to support the Connect Video feeds.
  • Following this larger example, if you go with the settings that I depicted previously in the screen captures above  to support the 12 Video feeds in a single meeting: DSL Bandwith and Standard Video quality = 213 kbps per stream:
  • 144 streams x 213 = 30672 kbps or 31 Mbps (rounded up) for the 12 separate cameras.
  • DSL VoIP = 22 kbps x 144 = 3168 kbs or 3Mbps additional bandwidth needed.
  • 300 kbps for each stream (assuming that the addin will not be ubiquitous) x 144 stream = 43Mbps + 3 Mbps for VoIP = 46Mbps.

Hopefully these exercises help with your planning for large successful meetings. There are other variables to consider such as Screen-Sharing and we will touch on those in a subsequent blog article. Consider, for example that when you are pushing the limits of your network, audio is usually given QoS priority over video.

Note: These examples assume that each client has a separate connection with the server and the Connect Edge servers are not remote to consolidate streams; they are not geographically distributed; they are collocated with the origin servers as is commonly the case so that each of the 12 attendees are receiving 11 subscribed streams from the data center (N-1).

Connect & Unified Voice (UV) Traffic Flow Diagram

Issue: Plan for the flow of traffic to enable UV among the various components in any Connect deployment: Connect, Flash Media Gateway (FMG), Session Initiation Protocol (SIP)

There are numerous documents on the topic of Unified Voice (UV) with Connect:

This diagram shows the flow of traffic and the protocols used for UV with Connect and is offered as a planning and a troubleshooting tool; click on the diagram to expand it for viewing:

 

Connect_FMG_Flow

Adobe Connect SIP and NAT issues

PROBLEM:-  I have configured the SIP settings with a new SIP provider but the call is not getting established when dialing onto the audio conference bridge, Audio conference bridge provider claims that they have received your call.

REASON:- Your Adobe connect server/ Flash Media Gateway(FMG) is deployed behind NAT and its domain /IP is not routable on internet, When you dial into the audio bridge through SIP , SIP header carries the local/internal IP address of FMG/Connect server and remote SIP provider does not have any idea how to reach to the local/internal IP address.

SOLUTION :- There are two ways to fix this

1. Adding Global Address tag in sip.xml file.

Add/Replace the below tag on sip.xml file on profileID sipGateway

<globalAddress>your-connect-domain</globalAddress>

your-connect-domain could be a domain name , IP address which should be routable on internet.

Example

<Profile>
<profileID>sipGateway</profileID>
<globalAddress>your-connect-domain</globalAddress>
<userName>101</userName>
<password>101</password>
<displayName>sipGateway</displayName>
<registrarAddress>10.10.10.10</registrarAddress>
<doRegister>true</doRegister>
<defaulthost>10.10.10.10</defaulthost>
<hostPort>0</hostPort>
<context>sipGatewayContext</context>
<supportedCodecs>
<codecID>G711u</codecID>
<codecID>speex</codecID>
</supportedCodecs>
</Profile>

 

2. Deploy FMG on another server/box having a public domain/ IP access.

 

UV Call drops automatically in Adobe Connect

PROBLEM :- I have configured SIP account on my connect server and the connection to audio bridge drops automatically.

 

REASON :- This could be a issue with the silence level detection on the remote SIP provider where the call is automatically disconnected when no RTP media / SIP signals are been sent by Adobe connect to remote SIP provider.

Flash Media Gateway (FMG) core.00.log indicates the following lines -:

2011-03-22::09:42:51.339 DEBUG SIP 1908 Received Call msg type : -1
2011-03-22::09:42:51.339 DEBUG SIPLEG 1908 [LEG ID:106] – **callLegStateChangedEvHandler** State=RVSIP_CALL_LEG_STATE_DISCONNECTED Reason=RVSIP_CALL_LEG_REASON_REMOTE_DISCONNECTED Dir=RVSIP_CALL_LEG_DIRECTION_OUTGOING
2011-03-22::09:42:51.339 DEBUG SIP 1908 [LEG ID:106] – Callleg disconnect cause was undefined, now set to 200 for reason RVSIP_CALL_LEG_REASON_REMOTE_DISCONNECTED
2011-03-22::09:42:51.339 DEBUG CALLLEG 1908 [LEG ID:106] – State Change SENDRECV -> HANGUP
2011-03-22::09:42:51.339 DEBUG SIPLEG 1908 [LEG ID:0] – **callLegStateChangedEvHandler** State=RVSIP_CALL_LEG_STATE_TERMINATED Reason=RVSIP_CALL_LEG_REASON_CALL_TERMINATED Dir=RVSIP_CALL_LEG_DIRECTION_OUTGOING
2011-03-22::09:42:51.339 DEBUG SIP 1908 [LEG ID:0] – Callleg disconnect cause was undefined, now set to 500 for reason RVSIP_CALL_LEG_REASON_CALL_TERMINATED
2011-03-22::09:42:51.355 DEBUG CALLLEG 4772 [LEG ID:106] – Hangup Call (cause 200), from FMSMGAppNodesHelper.cpp:1153
2011-03-22::09:42:51.355 DEBUG CALLLEG 4772 [LEG ID:105] – Bridging Completed for
2011-03-22::09:42:51.355 DEBUG CALLLEG 4772 [LEG ID:105] – Hangup Call (cause 200), from FMSMGAppNodesHelper.cpp:1251
2011-03-22::09:42:51.355 INFO CALLLEG 4772 [LEG ID:105] – Hangup [EXEC] [OK]
2011-03-22::09:42:51.355 DEBUG CALLLEG 4772 [LEG ID:105] – App bridge(sip|18665463377@sipGateway) Returns 0 (next ID:1)
2011-03-22::09:42:51.355 DEBUG CALLLEG 4772 [LEG ID:105] – Going For State 7
2011-03-22::09:42:51.355 DEBUG CALLLEG 4772 [LEG ID:105] – Call Leg HANGUP, cause: OK
2011-03-22::09:42:51.355 INFO CALLLEG 4772 [LEG ID:105] – CallLeg Ended
2011-03-22::09:42:51.355 DEBUG CALLLEG 4772 [LEG ID:105] – Cleaning up Leg [HANGUP]
2011-03-22::09:42:51.355 INFO RTMP 4328 Received onStatus <NetStream.Unpublish.Success> code <status> classType <fmg/fmg/1 is now unpublished.> description <(null)> details
2011-03-22::09:42:52.339 DEBUG CALLLEG 4480 [LEG ID:105] – Hangup Call (cause 200), from FMSMGAppNodesHelper.cpp:1153
2011-03-22::09:42:52.339 DEBUG CALLLEG 4480 [LEG ID:106] – Bridging Completed for
2011-03-22::09:42:52.339 DEBUG CALLLEG 4480 [LEG ID:106] – Hangup Call (cause 200), from FMSMGAppNodesHelper.cpp:960
2011-03-22::09:42:52.339 DEBUG CALLLEG 4480 [LEG ID:106] – Going For State 7
2011-03-22::09:42:52.339 DEBUG SIPLEG 4480 [LEG ID:106] – hangupHandler called…sipGateway
2011-03-22::09:42:52.339 DEBUG SIP 4480 Closing audio RTP session
2011-03-22::09:42:52.339 DEBUG SIP 4480 closed audio socket
 
 

SOLUTION -:

1. Do not leave the audio conference bridge idle for silence level detection algorithim

2. Login onto the connect server , open the console page > Flash Media Gateway Settings > Server Configurations.

3. Set the “Registration” value as “Required”.

4. Decrease the value of Registration Expiration in steps of 60

5. Save the configuration and re-test the issue.

6. Repeat step 4 and 5 until the issue is fixed.