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LanScape VOIP Media Engine™
 LanScape Support Forum -> LanScape VOIP Media Engine™
Subject Topic: Bridging calls Post ReplyPost New Topic
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peter.conklin
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Joined: May 01 2008
Location: Australia
Posts: 9
Posted: May 01 2008 at 10:58pm | IP Logged Quote peter.conklin

Hi,

Is there a way to "Bridge" or "Trombone" 2 seperately dialled and connected SWITCH type calls, similar to what would be done on a MVIP/H.100 Bus?

The reason a transfer will not work is because once the call is transferred we will lose control/visibility of the call and unable to "take the call back" (They would both be PSTN call legs).

The only way I can see in the documentation is to use the IVRTX/RX functions to send the RTP to each other.

Would be nice if there was a BridgeCalls( Call1 , Call2 ) and Unbridge( Call1 ) , Unbridge (Call2) type of functions.

Regards
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support
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Joined: January 26 2005
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Posted: May 02 2008 at 12:27pm | IP Logged Quote support

Hi Peter,

You have asked a good question. You can “bridge” two calls together… here’s how…

The old way:
Before we added the enhanced conferencing capability to the media engine, developers would have to “cross wire” the phone lines using the Rx IVR and Tx IVR capabilities of the phone lines. Basically the app would logically tie the lines together and exchange the media between the bridged phone lines using the IVR inputs and outputs. This worked good but was kind of clunky.

The new (and simpler) way:
The media engine now supports the ability to define individual conference session ID for each phone line. Now if you want to bridge 2 lines, the media engine will do it all for you. In other words, your app simply has to place each line into the same conference session and then place the lines into conference mode. This is kind of cool because not only is it extremely easy to bridge two lines together, you can bridge (conference) any number of lines together. Pretty powerful. Here is the basic pseudo code for the VOIP app:

1)
Answer a call on one of the phone lines and assign it a conference ID.

2)
Answer a second call on another phone line and assign it the same conference ID that was assigned to the phone call that was answered in step 1.

3)
Once both calls are in the media engine SipInCall state, the voip app places both calls into conference mode (using the ConferenceLine() API procedure).

4)
While in conference mode, the media engine automatically “bridges” the call media between the phone lines.

5)
The voip app can then do whatever it needs to with the calls after they are in conference/bridged mode.

Hope this info helps.


Support


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peter.conklin
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Joined: May 01 2008
Location: Australia
Posts: 9
Posted: May 04 2008 at 5:00am | IP Logged Quote peter.conklin

The only concern I had was possible licensing implications of using conference over the Rx IVR/Tx IVR methods.

If we were to develop say something that needed to cater for 120 calls (240 bridged calls), would that mean we need to purchase 240 conference licenses? Or is there just a single conference license that allows any number of conferences?

I assume with the RxIVR/Tx IVR these licenses would not be applicable.

As our needs only needs 2 parties per conference, I thought maybe there is a 'cut down' license to allow only up to 2 conference per session?

Technically both methods might be the same but cost wise the conference method could become very expensive in a high density system.

Regards
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peter.conklin
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Joined: May 01 2008
Location: Australia
Posts: 9
Posted: May 04 2008 at 5:03am | IP Logged Quote peter.conklin

Sorry, typo in the above I mean we would need to puchase 120 conference licenses, not 240.
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support
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Posts: 1666
Posted: May 05 2008 at 1:23pm | IP Logged Quote support

Hi Peter,

When we license the media engine, conferencing is always enabled. Very rarely have we restricted in any way or completely disabled conferencing support.

For licensing, the only parameter you would have to consider is how many total lines you will need. In this case 240 lines. We don’t license a 240 line media engine so your next best selection would be to go with the 256 line beast found here:

LanScape VOIP Media Engine™ - Supports up to 256 concurrent network phone lines:
http://www.lanscapecorp.com/Store/pd1002651502.asp?categoryI d=0

Eventually our two groups will have to determine your exact media engine usage and deployment requirements to finalize appropriate licensing.

If you have a spare moment, we would like to hear more details regarding what you are planning to develop and deploy. Don’t hesitate to be discreet or to mask your product trade name or your company name. We are only interested in the deployment model and end result functionality required in the final VOIP product you intend to develop.


Support

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