Configuration Tips for Polycom VVX phones and Trio800 SmartHub

In this blog post, I would like to cover some of the configuration tips on deploying and managing the Polycom VVX family of VoIP phones as well as the Polycom Trio8800 SmartHub registered to Skype for Business environments. Note that this blog post is based on an on-premise deployment of Skype for Business Server. This post does not cover Cloud Connector Edition (CCE) scenarios, which is targeted for a future series of blog posts. As always, IT professionals looking to deploy these devices in an enterprise production environment should leverage the documents available the Polycom Support site. The firmware versions I'm using for this blog post is UCS 5.4.1.14510 for the VVX and UCS 5.4.3.2342 for the Trio8800 SmartHub.

1. Preparing the Infrastructure for Polycom VVX/Trio deployments
A few basic infrastructure components are required to get a great out-of-box experience when deploying VVX/Trio phones for Skype for Business environments. First, the default behavior is that the phone obtains an IP address and time-zone from a DHCP server. NTP server address is provided via DHCP Option 42 and GMT Offset via DHCP Option 02. In cases where DHCP is not available, then the IP address can be manually set using the phone's UI. However, a NTP time server is required for the phone to obtain the correct date/time as this cannot be manually set on the phone. NTP requires UDP port 123 and the phone is automatically configured to connect to time.windows.com. Of course this can be changed to point to any internal NTP server as well. The manual IP address setting and NTP server plus time-zone setting on the phone UI are shown below:

Secondly, the necessary DNS records must be in place for the phone to do auto-discovery of the Skype for Business front end pool. Currently, the VVX's use the DNS SRV record _sipinternaltls._tcp.<sipdomain> and _sip._tls.<sipdomain> for internal and external networks respectively. In contrast, Skype4B clients use the lyncdiscover.<sipdomain> DNS A record by default, falling back to the SRV records. Thirdly, for on-premise Skype for Business environments, the VVX phone must be able to reach the Certificate Provisioning Service (STSURI)  in order to download the RootCA certificate and for TLS-DSK authentication. Without TLS-DSK then services that rely on web-tickets such as the Address Book web query will not work on the phones. Note that for Skype for Business on Office365, then there's no configuration needed as the necessary public DNS records will allow the phone to reach the Certificate Provisioning Service online automatically. For on-premise environments, DHCP Option 43 is typically used by the phone to discover the STSURI and this procedure is well documented in several blogs and on TechNet. Again, when DHCP is not available, then the VVX phones allow a manual entry of the STSURI using the web UI as shown below:

2. Dial Plans and other common settings
After successfully registering to Skype for Business, the phone will automatically obtain the dial plan via in-band provisioning. At this time, we recommend that the dial plan normalization rules be processed on the Skype4B server rather than on the phone itself. This is because the VVX phones do not yet support all of the RegEx available in Skype4B; a known limitation that will addressed in the near future with a new firmware update. Hence the following parameter on the phone is set by default to process rules on the Skype4B server:

For VVX phones registering directly to Office365 E5 CloudPBX, this value can be set to 1 instead in the event that normalization rules are not working as they should. There are also some addtional common settings that can be set either on the phone or in the case of CloudPBX registered devices, using the set-csIPPhonePolicy cmdlet as described in Adam Jacobs blog article. These common settings are self-explanatory and as shown below:

The following useful parameter controls whether a phone in keypad lock state can answer an incoming call. This parameter cannot be controlled using CloudPBX:

3. Implementing "Barge-in" functionality
Traditional PBX systems typically provide a featuer known as "Barge-in"  where a user can interrupt an ongoing active call without first being invited, hence the term "barge-in". This could be used in boss-admin scenarios where the boss has something urgent that requires him/her to join into the call that the admin is currently having. This feature is typically not available in modern Unified Communications platforms such as Skype for Business since there are better alternatives such as Instant Messaging and click-to-conference. That said, Polycom VVX phones do provide a similar "Barge-in" feature in the form of GroupPaging/PTT. This feature can be enabled using the phone's Web UI as shown below:

The GroupPaging/PTT feature is server independent and uses multicast. These settings can be configured as shown in the screenshot above along with other settings such as "Accept While Busy" which emulates a "Barge-In" by allowing the initiator to page any other user even if the latter is busy on a call. Up to 25 different Page Groups can be configured. In the example above, we have set the default page group 1 for the team and group 24 for the boss. When this feature is enabled, a new "Paging" softkey will appear on the phone's home screen and can then be used to initiate a page based on which groups have been configured for the phone:

Once a page is initiated, the all phones in the same page group will be able to hear each other even if they are currently on a call. The key difference between this and the traditional "Barge-in" feature is that the far side on existing active calls will not hear the audio from the page; ie page audio is only limited to those users within the page group.

4. Limiting Bandwidth usage for audio calls
The AudioBitRateKB parameter in set-CsConferencingPolicy was introduced in Lync2013 and is carried over to Skype for Business to allow administrators to set the network bandwidth utilization for audio calls. As per the TechNet article, this parameter controls the audio bitrate(Kb) used for audio transmissions and the default value is 200 which is the max. This setting applies for both conferences and peer-to-peer calls. While the default value poses no issues when deploying VVX phones for enterprise voice, some customers may want to set a lower value due to network constraints and in situations where CAC polices cannot be applied. In my tests, the out-of-box VVX default configuration will work fine when the AudioBitRateKB value is not less than 90. Below this, the phone will not be able to answer incoming calls. In this example, we create a new conferencing policy called "LowBandwidth" and set the AudioBitRateKB parameter to 64 before assigning it to a Skype4B user and a VVX Phone:

We then initiate a call from the Skype4B user to the VVX phone and from the SIP trace, the media codec negotiation shows that the G.722/8000 is chosen since that is the default codec used on the VVX:

However, since the AudioBitRateKB is set to a low value of 64 the call fails with a "Client Side Processing Error":

To resolve this issue, we can configure the VVX to use the "G722.1 (24 kbps)" audio codec using the phone's Web UI. Under the Codec Priorities section, select the G722 codec from the list on the right and use the arrow button to move it to the left as shown below:

Now we initiate the same call from the Skype4B user and the SIP trace shows that the codec is being used instead and the call proceeds without any issues: