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March 15, 2011
Question

Adobe Connect 8: 3rd Party VoIP integration - how?

  • March 15, 2011
  • 1 reply
  • 3172 views

Hello all,

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)..

Thanx, folks!

    This topic has been closed for replies.

    1 reply

    March 22, 2011

    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_install.pdf

    March 22, 2011

    Hi Heyward,

    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?

    March 22, 2011

    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.