Category Archives: VVX

Polycom Trio 5.5.2 firmware

Normally I’m not posting about specific firmware, but in the (5.5.2 Rev AC) release for Polycom Trio’s (yup, plural, there are two models now 8800 and 8500) the codec SILK has been made available.  Skype for Business SILK has only been added as an experimental feature so to make use of it, you need to modify the Codec Priorities on the device.  Rumors abound that we may see SILK in VVX500 and VVX600 series devices in the near future, but that lower end devices may not have the processing power needed.

SILK will only be capable of being used in peer-to-peer type calls as it’s not available in the Conferencing MCU, and a reminder that it is “Experimental” at this time.  There appears to be some typo’s currently if you want to modify a FTP Provisioning server cfg file for your Trio’s, the above exported config reports that SILK is in ksps instead of kbps.  Perhaps this is the change over to Kilo Samples Per Second, or a just a typo…  Either way, something to be mindful of when modifying config files.


This release for the RealPresence Trio 8800 system includes the following highlights:  (pdf release notes here)

• Screen Mirroring on RealPresence Trio Solution

• Software Update using Windows Server

• RealPresence Trio 8800 System Media Keepalive

• Toggle Content and People Video Streams

• Skype for Business User Experience Enhancements

• Viewing a Different Calendar in Skype for Business Mode

• Dynamic Port Ranges for Video and Content

• Adding a PSTN Participant to a Call

• Displaying Multiple Calendar Meetings on Connected Monitor

• Web Sign in for Skype for Business Online

• Secure Single Sign-On (SSO) with Third-Party Supporting Solutions

• Managing Skype for Business Conference Participant Level in the Call Roster Screen

• Device Lock

• Client Media Port Ranges for Quality of Experience (QoE)

• Microsoft Quality of Experience Monitoring Server Protocol (MS-QoE)

• Exchange Web Services Discovery

• Unified Contact Store

• Alert Tones for Mute Status

• Dial Plan Normalization

• Dial Plan for SIP URI Dialing

• Join a Meeting using SIP URI

• Hybrid Line Registration

• User Log Upload

• Audio, Video, and Content Port Ranges

• Media Transport Ports for audio, video, and content

• Experimental: Support for SILK Audio Codec

Firmware for the Trio’s can be found here:  Polycom Voice Software

VVX 5.5.x logon issue

I can’t take credit for this, but I document for my own future benefit as well as for the benefit of others.  A new client received a number of VVX phones which during the projects we standardized to 5.4.5 code but without a provisioning server, but that’s not important.  There were no issues with authenticating the phone with the Skype for business user account (not PIN, user account).  BUT when we went to test 5.5.1 firmware, the phones would not authenticate.  Even phones which were already authenticated prior to the update, were no longer able to sign in.

Setting the “dhcp.option43.override.stsUri” attribute of the phone, or manually setting “DHCP Option 43 Override STS-URI” setting under Settings | Provisioning Server, DHCP Menu, to the same value in DHCP Option 43, e.g.  allowed the phones to perform user authentications again.

Near as I can figure, this “might” be a result of using a VLAN for the VVX and perhaps 5.5.1 code can’t properly retrieve the option 43 from DHCP, which it doesn’t have this issue 5.4.x code.  This is merely speculation.  Very odd.


VVX Default Codec issues with Skype for Business

For a while now I’ve seen a randomly occurring call issues with clients using VVX phones.  There would sometimes be one way audio, or mostly no audio at all, but the call was connected.  Mainly with Response Group calls, but more recently I’ve encountered it on VVX to VVX Skype calls.

Fortunately it was so very similar to an issue I hit with Telus Cell phones making calls to Skype4b users who were behind and AudioCodes gateway and a Telus SIP trunks, which boiled down to a codec mismatch involving AMR.  Once the AudioCodes was locked down to only negotiate G.711Mu, problem solved. (I might have blogged about this already…)

Here is a screenshot from my VVX 600, with the default list of codec’s, though not in the default order:

Here is the RTP (Realtime Transport Protocol) mapping from a SIP trace, in bold are matching codec’s:

a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:112 G7221/16000
a=fmtp:112 bitrate=24000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000

And now one from a Skype for Business server for an inbound PSTN call to the same VVX phone:

a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:115 x-msrta/8000
a=fmtp:115 bitrate=11800
a=rtpmap:13 CN/8000
a=rtpmap:118 CN/16000
a=rtpmap:97 RED/8000

Oh, wait, 115 is a matching code, but the codec is all wrong.  This turns out to be G722.1C (48 kbps) from the VVX list.  According to Polycom forums which reference a page not found anymore, this codec is or is related to Siren14, and definitely not msrta/8000 aka Microsoft Realtime Audio Narrow band.

I did blog about this previously, Response Groups and Polycom VVX’s , but I hadn’t the time to dig in and confirm the offending codec, and I believe I now have.  I was also 3/4 the way through writing this when I realized I’ve already brought this subject to light.

Now from VVX600 to Skype for Business User, again bold are matching codecs:

a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:112 G7221/16000
a=fmtp:112 bitrate=24000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000

From Skype for Business User to VVX 600

a=rtpmap:104 SILK/16000
a=fmtp:104 useinbandfec=1; usedtx=0
a=rtpmap:114 x-msrta/16000
a=fmtp:114 bitrate=29000
a=rtpmap:9 G722/8000
a=rtpmap:112 G7221/16000
a=fmtp:112 bitrate=24000
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 SILK/8000
a=fmtp:103 useinbandfec=1; usedtx=0
a=rtpmap:116 AAL2-G726-32/8000
a=rtpmap:115 x-msrta/8000
a=fmtp:115 bitrate=11800
a=rtpmap:97 RED/8000
a=rtpmap:13 CN/8000
a=rtpmap:118 CN/16000
a=rtpmap:119 CN/24000

In this call, and because of my horrible ordering, I wanted to see if G7221/16000 was actually a viable codec.  Turns out it is, and according to a Jeff Schertz blog post, it’s a Siren 7 variant and nothing to do with G722.

I wasn’t able to test a VVX to VVX Skype call, but I suspect what may be happening in that situation is that the VVX’s are thinking 115 G7221/32000 but the Frontend translates and negotiates 115 x-msrta/8000, but that’s just a theory.

Resolution Time

Clean up time, and I have previously talked about this, but now I have a little bit more backing, and new case scenarios of when it’s impactful.

Siren22, G.722.1C, Siren14 and G.729AB can all be removed.  They’re not going to be used in a Lync/Skype environment, and because of potential cross matching on rtpmap=115 (I don’t know if it’s MS or Polycom issue), G.722.1c has to go.

Order Preference, G.722, G.711Mu (or A depending on your region) and optionally keep G.722.1 (24 kbps).  In the environments where I’ve cleared up the issues, I did remove G.722.1, but it wasn’t till today that I discovered it was actually a viable codec, doesn’t mean I trust it though.

If you happen to have a VVX 600, toast the Video Codecs as well, the camera that came with the phone last worked in a Lync 2010 environment.

If you have a Provisioning server, here is a code snippet to clean up your codec’s:

<WEB video.codecPref.H261=”0″ video.codecPref.H263=”0″ video.codecPref.H2631998=”0″ video.codecPref.H264=”0″ voice.codecPref.G711_A=”3″ voice.codecPref.G711_Mu=”2″ voice.codecPref.G722=”1″ voice.codecPref.G7221.24kbps=”4″ voice.codecPref.G7221_C.48kbps=”0″ voice.codecPref.Siren14.48kbps=”0″ voice.codecPref.Siren22.64kbps=”0″ voice.codecPref.G729_AB=”0″ />

If you still have troubles with VVX to VVX Skype calls, change the 4 to a 0 and get rid of it too.

VVX IP based Pairing

Hiding in the VVX 5.5.1 and BToE 3.4.x builds is some nifty code allowing for IP based phone pairing.  As it’s not in the released documentation we can assume this may not be meant for production.

Test/try this at your own peril, and certainly do not use with your Receptionist…  I have not encountered any issues, but I’m not a hard core handset user.  My configuration is with two separate LAN drops, one for my tower and one for my VVX 600.  I do not have a VDI environment to test this with and see if it works in that scenario, but I’ve had a few customers who have separate LAN ports for IP phones, with PoE gear, and are very anxious to have IP based BToE pairing become available.

First off, your VVX phone MUST be running or higher code. was released yesterday and so far so good with this new version.

Second, you need to have installed the Polycom BToE Connector, 3.4.x.  3.4.1 was released yesterday, so far so good.

Third, modify the Windows Registry.  If you are reading this post, I hope you are proficient enough with editing the registry with out blowing it up.

Close the BToE Connector, if it’s currently running, open up RegEdit and drill down to the following key:  HKEY_LOCAL_MACHINE\SOFTWARE\WOW6432Node\Polycom\Polycom BToE Connector  There should be a DWORD value called:  IP_PAIRING_EN, set it to 1.

Forthly, we need to add the following entries to the phone configuration as they are not exposed in the Web GUI.  I set up my own provisioning server, you can optionally Export  Configuration, add the entried, then Import the modified Configuration.

After you phone has rebooted, fire up the Polycom BToE Connector.  Right-click on the BToE icon in the SysTray and select “Pair with Phone”, and you’ll see the following screen.  Enter in the IP address of your phone and click Pair.

You may be prompted in your Skype client for credentials, but otherwise you should be good to go.

Last reminder, not supported at this time by Polycom, test/play at your own peril.


VVX 5.5.1 Log Setting Changes

This kept me and beta support spinning for a few days when I was trying to reproduce an issue in regards to logging, and suddenly I couldn’t set all of the logging levels I wanted through my provisioning server.

With 5.5.1 having tighter integration with Skype for Business, the devices now pull additional information from the server for items like QoE and also Log Level settings.   The following command impacts the 5.5.1 firmware according to the table below.  IF left at OFF, your provisioning server log settings will be overridden and not be applied.  Check the table, or the release notes guide for the settings you want, then add the additional ones either manually or through a Provisioning server process.

Set-CsUCPhoneConfiguration -LoggingLevel Off/Low/Medium/High


Not all of the various log setting are set via the Skype PS command, so you many still need to manually set via the web interface or use a provisioning server to set the other log items.

To see what things look like, or how it was eventually discovered, create a Phone Backup from the web interface and open up the .pbu file (I recommend Notepad++).  Waaaaaaay down, is a section labelled CALL_SERVER, these are items pulled from the Skype server.  Provisioning Server log level settings are shown in the CONFIG_FILE section.  Call server overrides Config File, search here if something you expect to be set isn’t working.

P.S. – Cautionary word of advice, prior to any updating of VVX firmware, set logging levels back to their defaults or remove the entries from the Provisioning server common.cfg file.  There is potentially an issue that I encountered which we’ve not been able to reproduce where we no longer can return the SIP logging level back to 4 (default) without breaking the BToE functionality.  Seems to apply to 310’s, possibly 410’s, but not 600’s.  Factory reset factory reset of the devices resolves, but not doing that for 150 devices, but leaving SIP logging level set to 2 fixed the BToE issue.  Yup, weird, reproduced with clients phones, reported to Polycom but haven’t been able to reproduce there.  Best guess is that something happened from going from 5.4.3 to 5.4.4, to 5.4.5 with elevated logging.  IF your BToE is suddenly weird and not connecting or working right, try changes SIP Log Level to 2.  IF that works, please also report it, I think the Polycom engineers think I’m crazy or something… :s

New Features of VVX 5.5.1

Polycom VVX 5.5.1 firmware was released today, along with BToE, Polycom VVX Firmware Download Site, I’ve been beta testing and beating up these VVX builds (lost count, probably 15+) since March, so I’ve been anticipating this day for awhile.  Soooo many features being added that I’m just giddy.

The biggest item in my opinion is the QoE reports generated by the phones and feed to the Skype Monitoring servers.  Yah, we can now see how the phones are performing and the quality of the calls.  Of course there is the Skype branded UI interface.

Additionally there is a IP Pairing feature, however I’m not seeing it in the release notes and may still be in beta.  It works for me but I am waiting to see if I need permission to share how to set that up.

Other new features, bells and whistles:  Polycom Release Notes

New Call Transfer User Interface Option – In this software version, users who transfer calls can more easily choose between Blind and Consultative transfers. On the Call Transfer screen for the user’s default transfer type, the user can press More to access a new soft key to change to the alternate transfer type. For example, if the user’s default transfer type is consultative, a Blind soft key is displayed.

Distribution List – Polycom phones registered with a Microsoft server enable you to perform multiple functions with a contact distribution list:
● Search for, add, and delete a distribution list
● View a distribution list, and expand a distribution list to view all      members
● View the contact card of a distribution list and of an individual member
● Conference with a distribution list
● Call an individual member of a distribution list
Distribution lists are available on the following VVX business media phones: VVX 201, VVX 300/310, VVX 301/311, VVX 400/410, VVX 401/411, VVX 500/501, VVX 600/601, and Polycom VVX Expansion Modules.

Microsoft Quality of Experience (QoE) Monitoring Server Protocol –  (MS-QoE) enables you to monitor the user’s audio quality and troubleshoot audio problems. QoE reports contain only audio metrics and do not contain video or content sharing metrics. This feature also enables you to query the QoE status of a phone from the Web Configuration Utility

Device Lock – You can configure phones to be protected with a lock code that enables users to access personal settings from different phones. You can configure Device Lock on the Skype for Business server or using Polycom parameters on a centralized provisioning server. If you enable Device Lock using both methods, centralized provisioning parameters take precedence. You cannot enable or disable Device Lock using the Web Configuration Utility or from the phone menu.

Polycom BToE PC Pairing – Administrators can use this feature to allow users to automatically or manually pair their VVX business media phone with their computer using the Polycom Better Together over Ethernet Connector application. Users can select the pairing mode in the Web Configuration Utility or in the Features menu on the phone. By default, BToE PC Pairing is enabled for phones registered with Skype for Business. When administrators disable BToE pairing, users cannot pair their VVX phone with their computer using BToE. In order to use this new functionality, you must install both BToE Connector App 3.4.0 and UC Software 5.5.1. For best results, Polycom recommends that you deploy BToE Connector App 3.4.0 before you deploy UC Software 5.5.1.

User Log Upload – To help troubleshoot user issues, administrators can enable or disable for users the ability to upload diagnostic logs from the phone or Web Configuration Utility and set log levels from the phone menu. This feature is available on all VVX business media phones registered with Skype for Business Server on-premises or online and with Microsoft Lync 2013 or 2010 Server.

Phone User Interface – The user interface for VVX 500 and 600 series business media phones was updated to match the theme used in the Skype for Business client. This feature is enabled by default on VVX 500/501 and 600/601 phones with the Lync/Skype Base Profile or SKU.

Unified Contact Store – Administrators can migrate users’ contacts to Microsoft Exchange Server to enable synchronization when users manage contacts or contact information from an application or device, for example, the VVX business media phone, Skype for Business client, Outlook, or Outlook Web Application from a mobile device.

Web Sign-In for Online Deployments – Web Sign-in enables users to securely log in to Skype for Business from the phone using a computer web browser or mobile device. Users can sign in concurrently to a maximum of eight devices by default. When users are signed in to multiple devices and sign out from one device, users remain signed in to all other devices. This sign in option is available only for Skype for Business Online deployments.

Expanded Support for USB Headsets – Support for the following Plantronics USB Headsets with VVX 500, VVX 600, VVX 501, VVX 601, and VVX 401 phones has been added to this release:
● Blackwire C310
● Blackwire C325
● Blackwire C725
● Blackwire C325.1
● Plantronics -CS520
● EncorePro HW540
● DA80 Headset Adapter