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  • PBX Related FAQ

PBX Related FAQ

Can I restore a UCM6200 series/UCM6510 backup to a UCM6300 series device?

No, there is no native support and UCM6200 series/UCM6510 backup file cannot be directly restored on a UCM6300 series device. Please contact Grandstream support to assist you to convert UCM6200 series/UCM6510 backup files for compatible UCM6300 series models.

Can I use GVC/GXV/Wave Lite to join UCM6300 video conference?

Grandstream Device/App Display Video Display Presentation Share Presentation
GVC3200/3202 Moderator video only To view presentation shared from Wave Web client, GVC/GXV devices must register to UCM using the public address provided by GDMS (RemoteConnect) Select Presentation->PC to share presentation to Wave Web/Wave mobile app
GVC3210 Up to 9-way (conference control not fully supported and compatibility to be improved)
GXV3350/3370/3380 Moderator video only N/A
Wave Lite Moderator video only N/A N/A
Wave Moderator video only Yes N/A

 

Can I use UCM6300 series as MCU for Grandstream GVC3200 series video conferencing device to support more than 9 video feeds (max for UCM6300) during UCM video conference?

When GVC3200 video conferencing devices are used with IPVideoTalk cloud service, up to 49 video feeds can be supported. However, when used with UCM6300 series, the max video feeds supported from UCM is 9. This is because UCM6300 series video conference is different from IPVideoTalk cloud service for mixing/forwarding the video feeds. The video feeds displayed on GVC3200 series also rely on GVC3200 series model's capability of SFU/MCU support.

- UCM6300 series uses SFU (Selective Forwarding Unit) to receive and forward video streams for each video participant. The video participant device receives multiple video streams from UCM for each video feed, and needs to support SFU to decode all video streams before displaying them.

- IPVideoTalk uses MCU (Multipoint Control Unit) which mixes all video feeds into one video stream so the participants only decode one video to see all parties' video.

For GVC3200/3202, it currently only supports MCU and can only display one video feed from the UCM which is the video conference moderator. Other participants' video feed currently cannot be seen. For GVC3210, SFU is supported and up to 9-way video can be displayed (conference control not fully supported on GVC3210 and compatiblity to be improved).

For video conference, will the number of video feeds and voice codec affect the number of attendees allowed in the conference?

Yes. Please see examples to illustrate the relationship below.

Note:

By default, UCM6300 series has “Full Band HD Voice” enabled which allows Opus 48KHz sampling rate to be used. If the users prefer more than 4 video feeds or more attendees in the conference, please log in UCM6300 series web UI as admin, navigate to PBX Settings->General Settings and configure the "Conference Voice Quality" option to “Broad Band HD Voice” instead.

Model Video feeds/screen sharing Voice codec Max attendees
UCM6301  4 video feeds + 1 screen sharing  

Opus (48KHz sampling rate)

12

Opus (8KHz sampling rate)

12

G.722 (16KHz sampling rate)

12

G.711 (8KHz sampling rate)

12
9 video feeds + 1 screen sharing

Opus (8KHz sampling rate)

12

G.722 (16KHz sampling rate)

12

G.711 (8KHz sampling rate)

12
UCM6302 4 video feeds + 1 screen sharing

Opus (48KHz sampling rate)

13

Opus (8KHz sampling rate)

15

G.722 (16KHz sampling rate)

20

G.711 (8KHz sampling rate)

20
9 video feeds + 1 screen sharing

Opus (8KHz sampling rate)

13

G.722 (16KHz sampling rate)

20

G.711 (8KHz sampling rate)

20
UCM6304

4 video feeds + 1 screen sharing

Opus (48KHz sampling rate)

26

Opus (8KHz sampling rate)

33

G.722 (16KHz sampling rate)

40

G.711 (8KHz sampling rate)

40
9 video feeds + 1 screen sharing

Opus (8KHz sampling rate)

27

G.722 (16KHz sampling rate)

40

G.711 (8KHz sampling rate)

40
UCM6308 4 video feeds + 1 screen sharing

Opus (48KHz sampling rate)

26

Opus (8KHz sampling rate)

33

G.722 (16KHz sampling rate)

60

G.711 (8KHz sampling rate)

60

9 video feeds + 1 screen sharing

Opus (8KHz sampling rate)

27

G.722 (16KHz sampling rate)

42

G.711 (8KHz sampling rate)

42

 

For video conferencing on UCM6300 series, how can we calculate the bandwidth required for each participant?

- 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 video conferencing on UCM6300 series, is 1080p used for all the video feeds?

No, only one video feed can be 1080p or 720p. Also, if there are more than 4 video feeds during video conference, or if there is screen sharing, the video bitrate for the 1080p/720p video feed will be reduced.

For video conferencing on UCM6300 series, what’s the default video bitrate and frame rate?

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.

How can I use UCM6300 series behind NAT?

- If you are using UCM RemoteConnect service, the UCM6300 series behind NAT can already work with your Grandstream end devices, Wave Web and Wave mobile apps without additional NAT settings in your network routers. However, if the end user is experiencing audio issues, please navigate to UCM6300 series web UI->Value-Added Features->RemoteConnect->Plan Settings and ensure Media NAT Traversal Service is enabled. By default, this is already enabled.

- If you are not using UCM RemoteConnect service:

Method 1: Configure port forwarding for your UCM. This will be limited by the NAT type of the network that UCM and end devices are located.

Method 2: Use self-deployed or available STUN/TURN server. On UCM6300 series web UI, admin can navigate to PBX Settings->RTP Settings, manually enable ICE Support and configure STUN/TURN server settings properly.

How do I configure the audio quality of conferences?

Navigate to PBX Settings->General Settings and configure the "Conference Voice Quality" option. If set to “Full Band HD Voice”, 48KHz sampling rate will be used for Opus codec and the audio quality using Opus will be notably better.

Note:

When "Full Band HD Voice" is selected, "Max Number of Video Feeds" in the Call Features->Video Conference->Conference Settings page is currently limited to 4. Pending enhancements in future firmware releases may possibly lift this performance limit.

I cannot use my deskphone, Wave web and Wave mobile app with the same SIP extension at the same time. Why?

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).

The UCM6200 series/UCM6510 cannot retrieve phonebooks from UCM6300 series from LDAP sync. Why?

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.

What are the Audio FEC and Video FEC options?

These are new options to minimize the effects of audio/video packet loss and allow the UCM6300 series to handle up to 50% packet loss for audio/video. They use the proprietary GS-FEC algorithm.

What are the Packet Loss Retransmission options?

- NACK:

By enabling negative-acknowledgement (NACK), the UCM6300 series will retransmit packets that have been lost in the initial transmission to repair the media stream.

- NACK+RTX:

It’s a mechanism based on NACK, which means it relies on RTCP packets to find out which packets are lost first. For NACK, it will retransmit based on the original packet. With RTX, a special payload is used to retransmit the packets that a NACK request indicated as lost. Retransmitted packets are sent in a different stream from the original media stream. The payload of the retransmission packet contains the payload header of the retransmission followed by the payload of the original packet. RTX retransmits using an extra ssrc, which will be marked in SDP during negotiation.

NACK+RTX can have more accurate statistics and it’s recommended for RemoteConnect users if there is packet loss during usage.

What is the ICE Support option in the UCM6300 series PBX trunk settings for?

ICE is a NAT traversal method for the UCM6300 series’ Wave WebRTC functionality and UCM6300 series SIP trunk has supported it. When using UCM6300 series SIP trunk service for Wave, if ICE Support is not enabled on UCM6300 series, you may experience audio, video and screen sharing issues.

Note:

Besides enabling ICE support on the trunk of UCM6300 series, the remote side (UCM or other trunk provider service) of the trunk needs to support ICE as well.

What is the NetEQ jitter buffer option?

Enabling NetEQ on the UCM6300 series helps reduce the effects of packet loss on audio received by the UCM.

It is a dynamic jitter buffer and error concealment algorithm used for hiding the negative effects of network jitter and packet loss. It helps keep latency as low as possible while maintaining the highest voice quality.

Note:

Enabling NetEQ will help with minimizing the effects of packet loss on audio received by the UCM6300 series. If there is packet loss in the audio sent from UCM to an endpoint, then the endpoint will need to handle it with its own packet loss mitigation implementation. NetEQ is used in WebRTC for audio QoS purposes. As such, UCM6300 series’ Grandstream Wave, which uses WebRTC, comes with NetEQ support.

What version of Asterisk does UCM6300 series run on?

The basic telephony software is based on Asterisk 16. However, a number of Grandstream’s proprietary voice/video algorithms, advanced video collaborations, device provisioning and remote management features are also integrated into the UCM6300 series.

When accessing UCM6300 series web UI via UCM’s IP address, the web browser shows the connection is not private and prevents the user from accessing it. What should I do?

This is because the UCM6300 series use self-signed certificate for web server by default. To avoid this issue, you could use UCM RemoteConnect plan for the UCM6300 series, which will provide domain name for the UCM. When users access Wave Web or admin accesses UCM Web portal, the web server will trust it and allow access.

You could also purchase domain name and certificate from other parties. Then upload the certificate to UCM6300 series web UI->System Settings->HTTP Server->Certificate Settings.

When I am calling another UCM extension, I hear a prompt "Please wait while I connect your call" followed by ringback tone a few seconds later. Why do I hear this prompt instead of ringback tone right away?

If the UCM extension has concurrent registration on Mobile device using Wave app, the caller will hear this prompt. This is introduced in UCM6300 series 1.0.3.x firmware for Wave app incoming call notification feature. Once your UCM account has been properly logged out from Wave app on mobile device, the caller will not hear the prompt again.

When my end devices are provisioned by UCM ZeroConfig, the config server path is not using the intended WAN/LAN IP of UCM. How can I fix this?

The UCM6300/6200/6510 series follow the rules below when providing the end devices Config server URL based on UCM WAN/LAN address.

1. UCM network interface method: Dual

1)If the end device is in the same subnet as UCM LAN 1, Zeroconfig provisioning URL will be based on UCM LAN 1 IP.

2)If the end device is in the same subnet as UCM LAN 2, Zeroconfig provisioning URL will be based on UCM LAN 2 IP.

3)If the end device IP does not appear to be in the same subnet as UCM, for example, UCM LAN1 IP: 192.168.129.88, UCM LAN2 IP: 192.168.2.1, end device IP: 10.16.254.68, Zeroconfig provisioning URL will be based on UCM LAN 1 IP.

2. UCM network interface method: Route

1) If the end device is in the same subnet as UCM LAN, for example, UCM LAN IP is 192.168.2.1, end device IP is 192.168.2.100, UCM WAN IP is 10.16.254.68:

1a) If end device IP is in UCM Zeroconfig subnet whitelist, Zeroconfig provisioning IP will be based on UCM WAN IP.

1b) If end device IP is not in UCM Zeroconfig subnet whitelist, Zeroconfig provisioning IP will be based on UCM LAN IP.

2)If the end device is in the same subnet as UCM WAN, for example, end device IP: 192.168.129.88, UCM WAN IP: 192.168.129.66 (they have the same gateway), Zeroconfig provisioning IP will be based on UCM WAN IP.

3)If the end device IP does not appear to be in the same subnet as UCM, for example, UCM WAN IP: 192.168.129.88, UCM LAN IP: 192.168.2.1, end device IP: 10.16.254.68:

3a) If end device IP is in UCM Zeroconfig subnet whitelist, Zeroconfig provisioning URL will be based on UCM WAN IP.

3b) If end device IP is not in UCM Zeroconfig subnet whitelist, Zeroconfig provisioning URL will be based on UCM LAN IP.

3. UCM network interface method: Switch

The Zeroconfig provisioning URL will be based on the UCM network interface IP (only one).

 

Note:

If UCM network interface method is Dual and “Default Interface” is configured, Zeroconfig provisioning URL will not be based on the UCM default interface setting. It still follows the rule above.

Will UCM6300 series support High Availability (HA) feature for redundancy support?

UCM6300 series will support HA feature for redundancy in the future. The current firmware does not support it.

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