as far as i know there is a possibility to integrate 3rd Party VoIP into Connect 8. Which finally should mean default Adobe VoIP channel is replaced by the service from the 3rd Party VoIP provider so that when one clicks the microphone button in the meeting room, this is the 3rd Party VoIP channel which is activated, not the default VoIP channel of Connect 8.
If i am right, could someone please kindly provide with some manual, which describes technical details of how this should be done and what are the technical requirements to the VoIP provider, so that we can discuss these details with our current VoIP partner. Unfortunately the standard Help gives only the overview of this functionality, no technical details and the only document which i got from the local Adobe office is this one (and it is also just a general information)..
Yes, Adobe Connect can integrate with any SIP provider:
Adobe Connect Universal Voice uses a component called Flash Media Gateway to send and receive audio from a SIP
server. Install Flash Media Gateway and configure it to communicate with a SIP server. The SIP server can be hosted
by a third-party or part of your company’s infrastructure. (SIP providers are also called VoIP providers.)
See the installation documentation: http://help.adobe.com/en_US/connect/8.0/installconfigure/connect_8_ins tall.pdf
thank you for your reply, but there is one very important difference in between what you propose and what i am looking for. Universal Voice (UV) supposes dialling somewhere, i.e. using your landline/mobile as a part of the process of participating in the conference. Moreover, when you're in, you gotta hold your handset all the meeting long, so UV does not give you the hands-free mode vs. VoIP when you just plug headset and mic and here you are - convenient and easy. This is where the question comes from - to integrate VoIP (SIP) provider's service so that the standard audio channel of Connect is replaced with the SIP provider's one, when you click the microphone icon in the menu. No dialling, no headset - same algorithm as before, but audio of higher quality at the end of the day. Possible?
That is truly available today in Connect!
When the HOST begins the meeting, they start the audio session via the menu. They have three options:
1. Use the computer (VOIP...hands-free....speakers/mike from PC/Mac...or use PC/Mac headset like Skype would)
2. Dial into the audio conference provider
3. Receive a call from the audio provider
Once they choose their option, the audio is begun.
If they want to start broadcasting VOIP then they go back to the audio menu and use the start broadcasting option which then allows attendees to use VOIP.
So, it is ultimately in the hands of the HOST to use VOIP or not. Some hosts choose not to use VOIP for obvious reasons:
1. bandwidth may be very low and using VOIP can increase the need for bandwidth
2. past network troubles may indicate an unstable or poorly created network infrastructure which has caused poor VOIP experiences
3. Desire to use telephonic voice grade audio provider rather than an unpredictable VOIP quality audio option
I frequently do not turn on VOIP if I do not know what the network condition is with attendees or if a large number of people will be on the call. More people on the call using VOIP will increase bandwidth requirements and some people may not have the bandwidth available to have a good experience.
The use of VOIP versus an audio provider depends on the country one resides in and the audio providers chosen. In some countries VOIP is better than some audio providers but in others it is the opposite. Your mileage may vary depending on so many factors.
yes, you are absolutely right in the above-said, i agree. However - again - this is smth different.
Let me explain in other words: when the host or attendee or whoever clicks the mic icon, the VoIP is activated, fine. The quality of built-in VoIP in AC8 is far from being ideal, saying the truth. For this reason i just wanted to ask if there is possibility to leave the same functionality in terms of VoIP usage (= to use VoIP you click the mic button), but the VoIP channel which is activated in this case will be the channel of the 3rd Party VoIP provider, not the default VoIP channel of AC8. Possible?
Understand, that the default audio used in Adobe Connect uses whatever third party SIP provider the installer used when they installed Adobe Connect Server. There is no VOIP in C8 unless you provide a third party SIP server address when installing Connect. AC does not provide SIP or VOIP unless an outside provider is integrated following the install guide.
See this on page 39-40 of the install guide:
"Adobe Connect Universal Voice uses a component called Flash Media Gateway to send and receive audio from a SIP server. Install Flash Media Gateway and configure it to communicate with a SIP server. The SIP server can be hosted by a third-party or part of your company’s infrastructure. (SIP providers are also called VoIP providers.)"
Understand, I have always been talking about Adobe Connect On-Premise or Licensed as we call it where you install your own server yourself and you maintain and control it completely. We have many customers that have their own servers so they can control everything to fit their needs.
If what you are really asking is can a customer of an Adobe Hosted server replace the provided SIP provider and replace that with one of their choosing, then the answer is no. Why? Because all customers on the one server cluster are using the server together and there is no way to separate the SIP/FMG server. Each Connect server is using one FMG and one SIP interface and that is all included inside the Connect Server. You cannot have any more than one FMG server with Connect. When you opt for using any hosted type service you give up a lot in the options for customization as well as flexibility of when you are upgraded to the next release or what features you want turned off or on. You are sharing that server with many other customers.
I hope that clarifies the issue.
firstly - yes, we are talking of the licensed version only, i.e. the one installed on our server and being part of our infrastructure. As for the 3rd Party VoIP integration, i examined p.39-40 and the picture below (taken from the guide) explains how this integration works:
Red comments are mine. So i take the 3rd Party VoIP integration as an addition to those pre-defined telephony adaptors (Avaya, Intercall, etc.), which finally means that the process of deploying either solution out of those two is the same: you set up the dialling sequence options, passwords, etc. and (the main inconvenience!) one has to use phone, i.e. this is not a hands-free option for the attendee.
What i want to find out is: can we integrate 3rd Party VoIP provider to the infrastructure so that the dialling process itself as well as phones are eliminated and the attendee only has to click microphone icon at the top menu of the conference room to use audio from the 3rd Party VoIP provider - not the default audio channel which is activated when the mic icon is clicked (= AC8 default audio channel).
In terms of data flow (see p.6 of the Guide you referred to) the solution i am looking for supposes a direct connection between parts of the infrastructure highlighted with blue rectangles:
In this case the attendee clicks the microphone icon in the meeting room and the 3rd Party VoIP channel is activated instead of the default AC8 audio. No headphones, no dialling.
Yes, you can use any VOIP phone device or use a headset or use the PC/Mac speakers/microphone in Connect easily. I do this all the time. You have the option to use any SIP provider you want when you install Connect.
The picture is using a phone picture but it can be ANY audio device connected to your computer.
If you have further questions on this topic, please let's take this offline and trade emails directly. I would be happy to help you or further explain options. My direct email is: firstname.lastname@example.org