To avoid problems with select Linux sound card drivers turn off Enhanced Audio in a Connect Meeting:
Some crashes on Linux are avoided by turning off Enhanced Audio and canceling Acoustic Echo Cancellation Mode.
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:
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
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:
Under Meeting > Preferences, there are two important options that you are going to adjust – Bandwidth and Video:
The size of the video streams commensurate with each webcam instance will depend on how you configure these settings.
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:
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:
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).
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:
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
your-connect-domain could be a domain name , IP address which should be routable on internet.
2. Deploy FMG on another server/box having a public domain/ IP access.
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
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.
PROBLEM -: I have enabled Universal voice feature on my Adobe connect server , when i dial into the audio conference bridge (using test dial-in steps), pass the audio conference code , the IVR prompt doesnt realize the conference codes and prompts to re-enter the conference code.
When i dial the same conference bridge using my base phone/mobile and pass the audio conference code , IVR accepts it.
SOLUTION -: Try to replace the SIP gateway/provider/vendor with another vendor which supports DTMF mode RFC 2833.
When you pass the conference code Adobe connect FMG (Flash Media Gateway) sends out DTMF tones to the the remote SIP gateway.
Adobe connect supports only RFC 2833 standard mode of DTMF communication. It seems that the remote SIP provider is not supporting this mode and the conference code send out as DTMF tones by FMG to the audio conference bridge is failing due to non supportive DTMF media (RFC 2833 standards) by the SIP gateway/provider.
PROBLEM -: I want to enable Universal Voice feature on my on-premise Adobe connect server using our internal Cisco call manager server.
There are some basic network configurations needed for this integration as -:
1. SIP port 5060 should be opened between Adobe connect and cisco call manager for SIP communications
2. Port range 5000 – 6000 should be opened between Adobe connect and cisco call manager for transversing voice packets via RTP.
3. DTMF mode RFC 2833 must be supported by cisco call manager as well as its carrier.
Integration steps -:
1. I need to create username/password based SIP account on cisco call manager
2. Login into cisco call manager and Navigate to Device -> Phone -> Add new ->
Refer to the screenshot below
Select Third party Basic device
Select Sip profile as standard sip profile
Select Digest user as LDAP ID of a user x
Check mark Media Termination Point Protocol.
3. Navigate to User Management > End User .
Search for LDAP ID of user x
The user-id of the LDAP user x would be the username of SIP account
The pin would be the password of the SIP account.
4. Configure your adobe connect SIP account settings with username of SIP account , password of the SIP account as per step 3.