Manually dialing out to a participant from a conference

The Pexip Infinity platform allows you to manually dial out to participants from a conference, on an ad-hoc basis. When you dial out to a participant in this way, a call is placed to their endpoint from the Virtual Meeting Room or Virtual Auditorium. If and when they answer the call, they will join the conference as either a Host or Guest, depending on the option that you selected. Participants joining the conference in this way will not go through the IVR screen and will not have to enter a PIN.

The participant could also take the form of a dedicated multimedia stream to enterprise CDN (Content Delivery Network) streaming and recording services such as Wowza, Adobe, VBrick, Quickchannel, Qumu and Azure Media Services, and to public streaming services such as YouTube, Facebook and Periscope. Any Pexip conference can be streamed as a live event to an unlimited number of viewers, and can automatically be recorded and stored for later consumption. For more information, see Streaming and recording a conference.

Call Routing Rules may be applied when dialing out from a conference to a new participant:

  • Call Routing Rules are applied if automatic routing is selected when placing the call — this means that the dialed alias must match an outgoing Call Routing Rule for the call to be placed (using the protocols and call control systems etc. as configured for that rule).
  • For manual routing, Pexip Infinity always attempts to route the call according to the requested protocol and media capabilities.

The ability to manually dial out to a participant is enabled by default. You can disable this setting by going to the Global settings page (Platform configuration > Global settings) and clearing the Enable outbound calls checkbox (note that this will also disable calls made via the Pexip Distributed Gateway).

Both Administrators and conference hosts can manually add a participant to a conference. Administrators can do so Using the Administrator interface, and hosts can do so Using Infinity Connect.

You can also configure a Virtual Meeting Room or Virtual Auditorium so that one or more participants are dialed out to automatically whenever a conference starts. For more information, see Automatically dialing out to a participant from a conference.

If your environment includes a PSTN gateway or uses an ITSP (Internet telephony service provider), consider the potential for toll fraud if you have Call Routing Rules that can route calls to the PSTN gateway or ITSP, or if you allow conference participants to dial out to other participants via the PSTN gateway or ITSP. See PSTN gateways and toll fraud for more information.

When dialing out from a conference, the outbound call bandwidth limit is inherited from the VMR's bandwidth settings. (If a Call Routing Rule is applied, the rule's bandwidth settings are not used.)

Using the Administrator interface

You can use the Pexip Infinity Administrator interface to dial out to a participant from a Virtual Meeting Room or Virtual Auditorium. If and when the call is answered, that endpoint will join the conference.

To dial out to a participant using the Administrator interface:

  1. Select the Virtual Meeting Room or Virtual Auditorium to dial the participant from. You can do this in any of the following ways:

    • Go to Service configuration > Virtual Meeting Rooms and select the name of the Virtual Meeting Room.
    • Go to Service configuration > Virtual Auditoriums and select the name of the Virtual Auditorium.
    • If the conference that you wish to dial out from is already in progress, go to Status > Conferences and select the name of the Virtual Meeting Room or Virtual Auditorium being used.
  2. At the bottom left of the screen, select Dial out to participant.
  3. Complete the following fields:

    Field Description
    System location

    Select the system location from which the call will be placed. If there is more than one Conferencing Node in that location, Pexip Infinity will choose the most appropriate.

    The system location determines which H.323 gatekeeper or SIP proxy to use to route the call.

    Service alias This lists all the aliases that have been configured for the selected Virtual Meeting Room or Virtual Auditorium. The participant will see the incoming call as coming from the selected alias.
    Participant alias The alias of the endpoint/participant that you want to dial.
    Route this call

    Select how to route the call:

    • Manually: uses the requested Protocol and the defaults for the specified System location.
    • Automatically: routes the call according to the configured Call Routing Rules— this means that the dialed alias must match an outgoing Call Routing Rule for the call to be placed (using the protocols and call control systems etc. as configured for that rule).

    Default: Manually.

    Participant display name

    An optional user-facing display name for this participant, which may be used in participant lists and as the overlaid participant name (if enabled).

    If this name is not specified then the Participant alias is used as the display name instead.

    Protocol

    The signaling protocol to use when dialing the participant. Select either SIP, H.323, or if the endpoint is a Skype for Business / Lync client, select Lync / Skype for Business (MS-SIP). The RTMP protocol is typically used when adding a streaming participant. Note that if the call is to a registered device, Pexip Infinity will instead use the protocol that the device used to make the registration.

    This field only applies when Route this call is set to Manually.

    Call capability

    Allows you to limit the media content of the call. The participant being called will not be able to escalate beyond the selected capability. For more information, see Controlling media capability.

    Default: Main video + presentation.

    This field only applies when Route this call is set to Manually.

    Role Select whether you want the participant to join the conference as a Host or Guest.
    DTMF sequence

    An optional DTMF sequence to be transmitted after the call to the dialed participant starts.

    A DTMF sequence can include:

    • the digits 0-9
    • * (asterisk)
    • # (hash)
    • , (comma).

    The DTMF tones will be sent 3 seconds after the call connects, one at a time, every 0.5 seconds. A comma is a special digit that represents a 2 second pause (multiple commas can be used if a longer pause is needed).

    For example, if you are dialing an audio bridge and want to enter conference number 777 followed by #, pause for six seconds and then supply conference PIN 1234 followed by #, you would configure 777#,,,1234# as your DTMF sequence.

    Streaming

    Identifies the dialed participant as a streaming or recording device. See Streaming and recording a conference for more information.

    Dual stream (presentation) URL

    When adding a dual streaming RTMP participant, this specifies the RTMP URL for the second (presentation) stream.

    Leave this field blank when adding a single streaming participant.

    Keep conference alive

    Determines whether the conference will continue when all other participants have disconnected:

    • Yes: the conference will continue to run until this participant has disconnected (applies to Hosts only).
    • If multiple: the conference will continue to run as long as there are two or more If multiple participants and at least one of them is a Host.
    • No: the conference will be terminated automatically if this is the only remaining participant.

    Default: Yes.

    For streaming participants, we recommend that this option is set to No.

    For more information, see Automatically ending a conference.

  4. Select Dial out to participant.

A message Initiated dial out to participant will appear at the top of the screen.

To confirm whether the participant has joined the conference:

  1. Go to Status > Conferences.
  2. Select the name of the Virtual Meeting Room or Virtual Auditorium.
  3. Select the Participants tab.

The participant will appear in the list.

Using Infinity Connect

If you have Host privileges, you can use the Infinity Connect client to dial out to participants from the Virtual Meeting Room or Virtual Auditorium you are in. When you use this feature, a call is placed to the participant and if and when they answer the call, they will join the conference directly (they will not go through the IVR screen and will not have to enter a PIN).

Automatic routing is used when a next-generation Infinity Connect client adds a new participant to a conference. This means that the dialed alias must match an appropriate Call Routing Rule that applies to Outgoing calls from a conference for the call to be placed (using the protocols and call control systems etc. as configured for that rule). However, the user can force a specific dial out protocol by prefixing the destination address with sip: or mssip: or h323: or rtmp: (which can be used to support dial out to streaming services) — in these cases a Call Routing Rule is not required.

To dial out to a participant using Infinity Connect, from within a VMR:

  1. From the toolbar at the bottom of the screen, select Add participant:

  2. At the prompt, enter the address of the person you want to dial:

  3. Select whether you want the participant to have Host or Guest privileges.
  4. Select Call in.

The call is placed from the VMR to the participant and they will appear in the participant list with a green line under their name while their endpoint is ringing. If and when the participant answers the call they will join the conference; if they do not answer, or do not accept the call, they will disappear from the participant list.

For legacy Infinity Connect clients, there is the option to select a protocol of SIP (the default), Automatic, H.323, Lync/Skype or RTMP. Automatic means that the protocol is selected according to Call Routing Rules; RTMP is typically used when connecting to a streaming or recording service.