Wave Web users can change video resolution during video conference. When the user joins conference from Wave Web and shares video, by default the video resolution is 360p. To change video resolution, the user can click on the Options Gear icon on the upper right, open “Resolution” setting and select 1080p, 720p or 360p. Please note only one 1080p or 720p video feed is allowed in the video conference. If there is already a 1080p or 720p video feed in the conference, the user will not be allowed to change to 1080p or 720p.

For Wave mobile app, changing video resolution during conference is currently not supported.

Yes.

1. For endpoints:

- SIP server address: GDMS-generated address

- SIP server port: Pubic TLS Port value in the Value-added Features->UCM RemoteConnect->Plan page.

- SIP transport: TLS

- NAT traversal: STUN

- STUN server: nat-a.gdms.cloud

- Reboot the endpoint

2. For UCM VoIP trunks:

- Type: Peer SIP Trunk

- Host Name: GDMS-generated address + Public TLS Port value in the Value-added Features->UCM RemoteConnect->Plan page.

- From Domain: UCM Public Address found in the Value-added Features->UCM RemoteConnect->Plan page of the peer UCM.

- Transport: TLS

- Enable Heartbeat Detection: Enabled

- ICE Support: Enabled

This will be supported in a future update.

- For conference with 4 or less video feeds:

The required bandwidth for each participant (in Mbps) = (A*1+B*0.172)*1.2+0.1

A is the number of 1080p video feeds (A = 0 or 1);

B is the number of participants excluding A. (B= Total number of participants - A);

1.2 represents the consideration of extra video bandwidth cost such as packet headers;

0.1 represents the bandwidth for audio.

- For conference with more than 4 video feeds:

The required bandwidth for each participant (in Mbps) = X*0.172*1.2+0.1

X is the number of total participants;

1.2 represents the consideration of extra video bandwidth cost such as packet headers.

0.1 represents the bandwidth for audio.

- For conference with screen sharing on:

The required bandwidth for each participant (in Mbps) = (X*0.172+0.512)*1.2+0.1

X is the number of total participants,

1.2 represents the consideration of extra video bandwidth cost such as packet headers.

0.1 represents the bandwidth for audio.

Note:

The above calculation applies to bandwidth required for each participant. To estimate the bandwidth required for UCM6300 series, please consider the number of participants and add them together.

For 1080p video: 1Mbps/15fps.

For 720p video: 512kbps/15fps.

For 360p video, 172kbps/15fps.

For screen sharing, 512kbps/5fps.

The above configurations can be modified per user’s preferences and it may impact the overall max number of allowed conference attendees depending on the trade-offs between video quality and max number of attendees.

- When joining a meeting on Wave Web or mobile app, select the Moderator role and enter the room’s host code.

- If you are already in a meeting and there is no moderator in the meeting yet, you can click on the upper right Options Gear icon and select “Moderate meeting”. Then enter the host code to become moderator.

During meeting, the existing moderator can transfer moderator privileges to another meeting participant.

Click on the Options gear icon on the top right of the page and select "Audio Settings". You will then be able to select the audio input and output devices you want to use for Wave Web.

If the participant is using Firefox web browser to share video or screen from GS Wave web and the participant’s PC currently has high CPU usage, Firefox will automatically reduce the resolution of the shared video or shared screen from this participant. Therefore, the other participants may see video resolution reduced or presentation become blurring during that period.

There are 2 ways:

1. Ask the UCM system administrator to send the SIP account information email to you.

2. If you have logged in Wave Web already, click on your extension number on the upper right corner and select "My QR Code" option to display your QR code. Then you can use your Wave mobile app to scan to QR code to log in.

Since network traffic from Wave is encrypted, packet capturing tools will not be able to view the details of communication.

- For Wave web:

Logging is automatically enabled on Wave Web, and users can download logs at any time by click on the "Options" gear icon and selecting "Export Log".

Note: 

As long as the user hasn’t cleared cache on the browser, the log from Wave Web is available for export, even the tab/browser has been closed or the PC has been restarted.

- For Wave Mobile app:

On Wave mobile app login screen, users can enable log collection on Wave Mobile by tapping on the "Options" gear icon at the top right of the screen and toggling on the "Collect Logs" option. Tapping on "Export Logs" will download them to the device.

If the user is already logged in, users can navigate to Me->About->Export Logs and toggle on the "Collect Logs" option. Tapping on "Export Logs" will download them to the device.

This is caused by abnormal disconnections from the related party. For situations like these, if there is no RTP detected from the participant within 90 seconds, they will automatically be removed from the meeting.

In order to use the same SIP extension on deskphone, Wave web and Wave mobile app at the same time, the SIP extension must have 3 or more concurrent registrations allowed. By default, the number of concurrent registrations is set to 3. To confirm or change this setting, please log in UCM6300 series web UI as admin, navigate to Extension/Trunk->Extensions, edit the extension option “Concurrent Registrations” to be 3 or more.

Note:

To register SIP extension on deskphone or log in with the extension on Wave web / Wave mobile app, the same SIP registration password is used (which is the “SIP/IAX Password” on UCM6300 series web UI extension settings).

This could be due to either the UCM6300 series PBX firmware or the Wave mobile app not being updated to the latest version. Please make sure both the UCM6300 series PBX and Wave app are up-to-date.

UCM6300 series use rsync and port 873 for LDAP syncing. There is currently no way to modify this port number. Regardless of this, UCM6300 series can still retrieve LDAP phonebooks from the UCM6200 series/UCM6510 without needing to modify the LDAP sync port on the UCM6200 series/UCM6510. However, the UCM6200 series/UCM6510 are currently unable to retrieve phonebooks from UCM6300 series. This will be addressed in a later UCM6200 series/UCM6510 firmware.