To configure the SIP endpoints to work with the MCU, go to
. This makes it easier to add endpoints to conferences because you can choose names from a list rather than adding network addresses.Refer to the table below for tips on adding a SIP endpoint to the MCU. After entering the settings, click
.
Field | Field description | Usage tips |
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Name |
The name of the endpoint. |
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Address |
The IP address, host name, directory number, or SIP URI (in the format [email protected]). |
The address of the SIP endpoint can be a directory number if you are using a SIP registrar. Note that square brackets are mandatory for IPv6 addresses. |
Use SIP registrar |
Allows calls to this endpoint to use a directory number (in the Address field) and the SIP registrar. |
This setting is dependent on the MCU-wide SIP registrar usage setting. If the MCU-wide setting is disabled, then the endpoint will not use the SIP registrar (irrespective of whether you check this box). If SIP registrar usage is enabled on the page, then checking this box allows the endpoint to use the registrar. Only applicable if Outbound call configuration is set to Use registrar. If Use SIP registrar is checked, the dialed URI gets appended with Outbound domain if present, otherwise Outbound address gets appended. If Use SIP registrar is unchecked, then nothing is appended to the dialed URI. The Use SIP registrar check box has no effect if Outbound call configuration is set to Call direct or Use trunk. |
Outgoing transport |
Select the protocol to be used for call control messages for outgoing call connections to this endpoint. |
If you want this endpoint to use the MCU-wide outgoing transport setting, select <use box-wide setting>. If this endpoint uses TCP, select TCP as the outgoing transport. If this endpoint uses UDP, select UDP as the outgoing transport. If this endpoint uses TLS, select TLS. Note that if you want the MCU to use TLS for call setup, you must have the encryption feature key and the TLS service must be enabled on the page. Using TLS for call setup is not sufficient for the call to be considered encrypted such that it can participate in a conference which requires encryption. Where encryption is required in the conference configuration, a SIP call must use SRTP. For more information about SIP encryption, refer to Configuring encryption settings. This setting overrides the MCU-wide setting for Outgoing transport on the page. For more information about configuring SIP, refer to Configuring SIP settings. |
Redial behavior |
Defines whether and how the MCU will redial this endpoint if the connection fails:
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This setting defines redial behavior for an individual participant, which overrides the box-wide setting. |
Redial limit | Enables or disables the redial limit for this endpoint, and overrides the corresponding box-wide setting. |
The redial limit allows the MCU to stop trying to reconnect a failed call. When the limit is enabled, the MCU will attempt to reconnect up to ten times: once immediately after the connection failure; four times at one minute intervals thereafter, and once every five minutes for a further five attempts. When the redial limit is disabled, the MCU continues retrying - once every five minutes - after those first ten attempts. It will do this until the connection is made. If the connection is never made, the MCU continues retrying until either the conference or the participant is destroyed. The redial pattern has an initial delay when the redial behavior is set to Redial on any disconnection. With that setting, the MCU does not immediately redial after a deliberate disconnection - it waits 30 seconds. |
DTMF sequence |
The DTMF sequence to send to an endpoint after it answers the call. The sequence may be up to 127 characters long and may include digits 0-9 and the following characters: There is always a two second pause after the call connects, after which the MCU will send the DTMF tones at a rate of two per second. You can insert as many two second pauses as you need by inserting commas into the DTMF sequence. Leading and trailing commas are supported. |
This sequence enables the MCU to navigate through an audio menu. This is useful where a conference on the MCU dials out to an audio-only conference on an audio bridge. You can configure the audio bridge as a pre-configured endpoint (either H.323 or SIP) and specify the DTMF sequence which will than be used whenever the bridge is added to any conference. Alternatively, you can add the audio bridge as an ad hoc participant to an individual conference. For example, assume you want the MCU to dial out to a PIN-protected audio conference on an audio bridge. The conference ID is 555 and the PIN is 888. The audio bridge requires that you press # after entering the ID and after entering the PIN. In this example the DTMF sequence could be: 555#,,888#. The two commas represent a four second pause which allows the audio bridge's automated menu system time to process the ID and request the PIN. |
Suppress audio during DTMF |
Suppresses the audio stream while initial DTMF connection sequence is being sent, so that other conference participants do not hear the audio of this participant or interactive voice responders reacting to the tones. Outgoing only suppresses the audio to the endpoint while DTMF tones are sent to the endpoint. All also suppresses both incoming and outgoing audio for the participant while the initial DTMF sequence is being sent to the endpoint. |
This setting is independent of other audio muting mechanisms. Audio suppression is active for the duration of the DTMF tone sequence, including any deliberate pauses (commas in the DTMF sequence). |
Call-in match parameters |
These fields are used to identify incoming calls as being from the endpoint:
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The endpoint is recognized if all filled-in fields in this section are matched. Fields left blank are not considered in the match. Note that in some cases a SIP registrar can cause a call to appear to come from the IP address of the registrar rather than the IP address of the endpoint. In this case, to use call-in match parameters, leave the IP address field blank and enter the correct username. The call will be matched by username. When using LCS, the username that will be matched is the user's display name (e.g. Peter Rabbit) rather than the sign-in name ([email protected]). |
Display name override |
The name that is displayed in a conference as a label on the video from this endpoint. It is also the name of the endpoint as it appears on the MCU's web interface. |
The display name override is used in place of any identifier that appears on the endpoint's video or in the MCU's web interface. The endpoint could otherwise be identified by its SIP id, its E.164 number, or its IP address. If you use only non-printing characters, such as spaces, as a display name override, the MCU respects this 'blank' name as a label for video from this endpoint. In this case, the MCU does not show the non-printing override value when listing the endpoint in the web interface; it shows one of the endpoint's other identifiers instead. Note that once an endpoint has connected, you cannot change the display name via the web interface. |
Motion / sharpness trade off |
Choose whether to use the MCU-wide setting for motion/sharpness trade off, or configure an individual setting for this endpoint. Select from:
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The settings for motion (frames per second) and sharpness (frame size or resolution) are negotiated between the endpoint and the MCU. This setting controls how the MCU will negotiate the settings to be used with this endpoint. |
Transmitted video resolutions |
Choose the setting for transmitted video resolutions from the MCU to this endpoint. This setting overrides the MCU-wide setting on the page. |
Retain the default setting (use box-wide setting) unless you are experiencing problems with the display of certain resolutions by this endpoint. Endpoints advertise the resolutions that they are able to display. The MCU then chooses from those advertised resolutions, the resolution that it will use to transmit video. However, some endpoints do not display widescreen resolutions optimally. Therefore, you might want to use this setting to restrict the resolutions available to the MCU for transmissions to this endpoint. |
Content contribution |
Whether this endpoint is permitted to contribute the conference content channel. Select from: <use conference default>: this endpoint will use the Content contribution from endpoints setting from the per-conference configuration. Enabled: This endpoint is allowed to contribute the content channel, even if content contribution from endpoints is disabled in the per-conference configuration. Disabled: This endpoint is not allowed to contribute the conference channel, even if content contribution from endpoints is enabled in the per-conference configuration. |
This setting is provided to allow you to individually configure whether or not an endpoint is allowed to contribute content to a conference. To use the content channel, the Content status must be enabled at the MCU-wide level (on the page) and, for any given conference, Content mode must not be Disabled in the conference settings. |
Content receive |
Whether this endpoint is allowed to receive a separate content stream when in a conference. |
This setting is provided to allow you to individually configure whether or not an endpoint is allowed to receive content from a conference. To use the content channel, the Content status must be enabled at the MCU-wide level (on the Settings > Content page) and for any given conference Content channel video must also be enabled in the per-conference configuration. If set to Disabled the endpoint will not receive content or will receive the content in the normal video channel if that setting is enabled ( ). Note that Binary Floor Control Protocol (BFCP) content is supported. |
View border size |
Choose a border size for video transmitted to this endpoint. |
This sets a border thickness to display around the video image. This is useful where the image is displaying off the edges of the participant's screen; use a border to force the image to display properly. Applying a border size here means that this border size will always be used for this endpoint's transmitted video. Note that you can also apply a border to a participant in a conference by going to and clicking on the name of the conference and then altering this participant's settings. |
Default view family |
Sets the layout family to be used when calling out to this endpoint. |
If this is set to Use box-wide setting then the default view family that has been configured via the Conference settings page will be used. |
Preferred bandwidth from MCU |
Identifies the network capacity (measured in bits per second) used by the media channels established by the MCU to a single participant. |
These settings take priority over the Default bandwidth from MCU setting configured in the global conference settings. |
Preferred bandwidth to MCU |
The maximum combined media bandwidth advertised by the MCU to endpoints. |
These settings take priority over the Default bandwidth to MCU setting configured in the global Conference settings (see Conference settings). |
Layout control via FECC / DTMF | Whether this endpoint is able to change their view layout via far-end camera control (FECC) or
DTMF tones.
Choose from:
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This setting takes precedence over the per-conference layout control setting for conferences into which the endpoint is invited. This layout control setting does not govern layout control using the in call menu. The setting governs layout control using the endpoint's DTMF or FECC controls while the user is not using the menu. |
Send camera control to other participants |
Specifies whether FECC or DTMF from this participant may control a far end camera. This setting combines with layout control via FECC/DTMF to control the camera of the far end and the layouts of the conference. This setting overrides the global conference setting for this endpoint.
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There are two control mechanisms, FECC and DTMF, either (or both) of which can be used for camera control or layout control. If one mechanism is allowed for camera control but not for layout control, then that mechanism only controls the far end camera and does not affect the layout. Similarly, if one mechanism is allowed for layout control but not for camera control, then it is not possible to control the camera with that mechanism. In these cases, the endpoint can use FECC or DTMF controls directly to change the layout or adjust the far end camera. When one control mechanism can control either the layout or the far end camera, then that mechanism will always control the layout until "Zoom in" (FECC mechanism) or �1� (DTMF mechanism) is pressed. The control mechanism then switches over to control the camera. Far-end camera control always applies to the camera of the participant shown in the largest or top left pane (when panes are the same size). If you have no way to control the layout, then you cannot focus on a participant to allow you to adjust a particular camera. |
Mute in-band DTMF |
Use this option to mute in-band DTMF from this endpoint. Choose from:
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In some scenarios, where a conference is cascaded onto an audio bridge, it might be useful for one of the participants in that conference to be able to send in-band DTMF to the MCU. This is for the purposes of sending the conference ID or PIN to the audio conferencing bridge. In this case, the Mute in-band DTMF setting for the endpoint of that participant needs to be Never. However, you can instead send DTMF tones to the audio conferencing bridge directly from the MCU; for more information refer to Sending DTMF to an audio bridge. Unless you need to configure a particular setting for this endpoint, set this to use conference configuration and ensure you have the Mute in-band DTMF setting as required in the conference configuration (see Adding and updating conferences). This setting takes precedence over the per-conference Mute in-band DTMF setting for conferences into which the endpoint is invited. |
Appear as a recording device |
When this setting is enabled, the red recording indicator dot is visible to other conference participants. |
This participant is labeled as a recorder on the conference participants page. |
Video to use by default |
Allows you to replace this participant's video with that of another participant. Select <self> (default value) to display this participant's own video by default. If you select another preconfigured endpoint from the dropdown, the MCU will no longer use this participant's own video by default: instead, it will use the video stream from the selected endpoint in most circumstances, for example when this participant becomes the active speaker. |
If the selected Video to use by default is not available, the MCU will use this participant's own video if possible. If the selected Video to use by default is available, you can still show this participant's own video if you need to by explicitly choosing it in specific layout pane selections. |
Dial out as |
Determines whether this participant is a chair or a guest in automatic lecture mode. |
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Initial audio to MCU |
Select Active or Muted to define whether audio from this endpoint should be muted whenever it first joins a conference. |
If this is Muted, then the MCU will mute the stream whenever it successfully invites the endpoint or recognizes the endpoint when it calls in. Audio from preconfigured endpoints can also be muted by the conference's Mute on join configuration, but only when they call in. In that case, if either the conference's Mute on join checkbox for Audio is checked, or the endpoint's Initial audio to MCU is Muted (or both) then the audio from the endpoint is muted when it joins the conference. |
Initial video to MCU |
Select Active or Stopped to define whether video from this endpoint should be stopped whenever it first joins a conference. |
If this is Stopped, then the MCU will stop the stream whenever it successfully invites the endpoint or recognizes the endpoint when it calls in. Video from preconfigured endpoints can also be muted by the conference's Mute on join configuration, but only when they call in. In that case, if either the conference's Mute on join checkbox for Video is checked, or the endpoint's Initial video to MCU is Muted (or both) then the video from the endpoint is stopped when it joins the conference. |
Initial audio from MCU |
Select Active or Muted to define whether audio to this endpoint is muted whenever it first joins a conference. Note: The endpoint may not always detect DTMF tones from the MCU after you mute the audio from the MCU. |
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Initial video from MCU |
Select Active or Stopped to define whether video to this endpoint is stopped whenever it first joins a conference. |
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Initial Adaptive Gain Control | Defines whether or not the endpoint uses Adaptive Gain Control (AGC) when it first joins the conference. Select <use conference configuration> to inherit the setting from the conference. Otherwise, you can select Enabled or Disabled to override the conference-wide AGC setting. |
Any manual changes to the participant volume will turn AGC off for that participant. You can also manually enable or disable AGC for a participant that is already in the conference, on the page. |
Automatic disconnection |
When a participant disconnects from a conference and only endpoints set to Automatic disconnection are left, all those participants are disconnected. |
Set to enabled if you want this endpoint to be automatically disconnected from conferences when only endpoints set to Automatic disconnection remain in a conference when any other participant has disconnected. |
Custom codec selection |
Can be used to ensure only specific codecs are permitted on calls to (and received from) this endpoint. |
If Enabled, you can select which codecs are allowed to be used when communicating with this endpoint. This setting overrides the MCU-wide codec selection on the page. |
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